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LeafOS / LeafOS-Project / android_frameworks_av / 6197e41b609fbae78deb076f02056a717869b463 / . / services / audioflinger / Threads.cpp
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/*
**
** Copyright 2012, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
// #define LOG_NDEBUG 0
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include "Threads.h"
#include "Client.h"
#include "IAfEffect.h"
#include "MelReporter.h"
#include "ResamplerBufferProvider.h"
#include <afutils/DumpTryLock.h>
#include <afutils/Permission.h>
#include <afutils/TypedLogger.h>
#include <afutils/Vibrator.h>
#include <audio_utils/MelProcessor.h>
#include <audio_utils/Metadata.h>
#ifdef DEBUG_CPU_USAGE
#include <audio_utils/Statistics.h>
#include <cpustats/ThreadCpuUsage.h>
#endif
#include <audio_utils/channels.h>
#include <audio_utils/format.h>
#include <audio_utils/minifloat.h>
#include <audio_utils/mono_blend.h>
#include <audio_utils/primitives.h>
#include <audio_utils/safe_math.h>
#include <audiomanager/AudioManager.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <binder/PersistableBundle.h>
#include <com_android_media_audio.h>
#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <fastpath/AutoPark.h>
#include <media/AudioContainers.h>
#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
#ifdef ADD_BATTERY_DATA
#include <media/IMediaPlayerService.h>
#include <media/IMediaDeathNotifier.h>
#endif
#include <media/MmapStreamCallback.h>
#include <media/RecordBufferConverter.h>
#include <media/TypeConverter.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <media/audiohal/StreamHalInterface.h>
#include <media/nbaio/AudioStreamInSource.h>
#include <media/nbaio/AudioStreamOutSink.h>
#include <media/nbaio/MonoPipe.h>
#include <media/nbaio/MonoPipeReader.h>
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <media/nbaio/SourceAudioBufferProvider.h>
#include <mediautils/BatteryNotifier.h>
#include <mediautils/Process.h>
#include <mediautils/SchedulingPolicyService.h>
#include <mediautils/ServiceUtilities.h>
#include <powermanager/PowerManager.h>
#include <private/android_filesystem_config.h>
#include <private/media/AudioTrackShared.h>
#include <system/audio_effects/effect_aec.h>
#include <system/audio_effects/effect_downmix.h>
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_spatializer.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <fcntl.h>
#include <linux/futex.h>
#include <math.h>
#include <memory>
#include <pthread.h>
#include <sstream>
#include <string>
#include <sys/stat.h>
#include <sys/syscall.h>
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
// 0; but one side effect of this is to turn all LOGV's as well. Some messages
// are so verbose that we want to suppress them even when we have ALOG_ASSERT
// turned on. Do not uncomment the #def below unless you really know what you
// are doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
// TODO: Move these macro/inlines to a header file.
#define max(a, b) ((a) > (b) ? (a) : (b))
template <typename T>
static inline T min(const T& a, const T& b)
{
return a < b ? a : b;
}
namespace android {
using audioflinger::SyncEvent;
using media::IEffectClient;
using content::AttributionSourceState;
// Keep in sync with java definition in media/java/android/media/AudioRecord.java
static constexpr int32_t kMaxSharedAudioHistoryMs = 5000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
// allow less retry attempts on direct output thread.
// direct outputs can be a scarce resource in audio hardware and should
// be released as quickly as possible.
// Notes:
// 1) The retry duration kMaxTrackRetriesDirectMs may be increased
// in case the data write is bursty for the AudioTrack. The application
// should endeavor to write at least once every kMaxTrackRetriesDirectMs
// to prevent an underrun situation. If the data is bursty, then
// the application can also throttle the data sent to be even.
// 2) For compressed audio data, any data present in the AudioTrack buffer
// will be sent and reset the retry count. This delivers data as
// it arrives, with approximately kDirectMinSleepTimeUs = 10ms checking interval.
// 3) For linear PCM or proportional PCM, we wait one period for a period's worth
// of data to be available, then any remaining data is delivered.
// This is required to ensure the last bit of data is delivered before underrun.
//
// Sleep time per cycle is kDirectMinSleepTimeUs for compressed tracks
// or the size of the HAL period for proportional / linear PCM tracks.
static const int32_t kMaxTrackRetriesDirectMs = 200;
// don't warn about blocked writes or record buffer overflows more often than this
static const nsecs_t kWarningThrottleNs = seconds(5);
// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;
// maximum time to wait in sendConfigEvent_l() for a status to be received
static const nsecs_t kConfigEventTimeoutNs = seconds(2);
// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
// maximum divider applied to the active sleep time in the mixer thread loop
static const uint32_t kMaxThreadSleepTimeShift = 2;
// minimum normal sink buffer size, expressed in milliseconds rather than frames
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalSinkBufferSizeMs = 20;
// maximum normal sink buffer size
static const uint32_t kMaxNormalSinkBufferSizeMs = 24;
// minimum capture buffer size in milliseconds to _not_ need a fast capture thread
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalCaptureBufferSizeMs = 12;
// Offloaded output thread standby delay: allows track transition without going to standby
static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
// Direct output thread minimum sleep time in idle or active(underrun) state
static const nsecs_t kDirectMinSleepTimeUs = 10000;
// Minimum amount of time between checking to see if the timestamp is advancing
// for underrun detection. If we check too frequently, we may not detect a
// timestamp update and will falsely detect underrun.
static constexpr nsecs_t kMinimumTimeBetweenTimestampChecksNs = 150 /* ms */ * 1'000'000;
// The universal constant for ubiquitous 20ms value. The value of 20ms seems to provide a good
// balance between power consumption and latency, and allows threads to be scheduled reliably
// by the CFS scheduler.
// FIXME Express other hardcoded references to 20ms with references to this constant and move
// it appropriately.
#define FMS_20 20
// Whether to use fast mixer
static const enum {
FastMixer_Never, // never initialize or use: for debugging only
FastMixer_Always, // always initialize and use, even if not needed: for debugging only
// normal mixer multiplier is 1
FastMixer_Static, // initialize if needed, then use all the time if initialized,
// multiplier is calculated based on min & max normal mixer buffer size
FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
// multiplier is calculated based on min & max normal mixer buffer size
// FIXME for FastMixer_Dynamic:
// Supporting this option will require fixing HALs that can't handle large writes.
// For example, one HAL implementation returns an error from a large write,
// and another HAL implementation corrupts memory, possibly in the sample rate converter.
// We could either fix the HAL implementations, or provide a wrapper that breaks
// up large writes into smaller ones, and the wrapper would need to deal with scheduler.
} kUseFastMixer = FastMixer_Static;
// Whether to use fast capture
static const enum {
FastCapture_Never, // never initialize or use: for debugging only
FastCapture_Always, // always initialize and use, even if not needed: for debugging only
FastCapture_Static, // initialize if needed, then use all the time if initialized
} kUseFastCapture = FastCapture_Static;
// Priorities for requestPriority
static const int kPriorityAudioApp = 2;
static const int kPriorityFastMixer = 3;
static const int kPriorityFastCapture = 3;
// Request real-time priority for PlaybackThread in ARC
static const int kPriorityPlaybackThreadArc = 1;
// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the
// track buffer in shared memory. Zero on input means to use a default value. For fast tracks,
// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'.
// This is the default value, if not specified by property.
static const int kFastTrackMultiplier = 2;
// The minimum and maximum allowed values
static const int kFastTrackMultiplierMin = 1;
static const int kFastTrackMultiplierMax = 2;
// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
static int sFastTrackMultiplier = kFastTrackMultiplier;
// See Thread::readOnlyHeap().
// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
static const size_t kRecordThreadReadOnlyHeapSize = 0xD000;
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
static nsecs_t getStandbyTimeInNanos() {
static nsecs_t standbyTimeInNanos = []() {
const int ms = property_get_int32("ro.audio.flinger_standbytime_ms",
kDefaultStandbyTimeInNsecs / NANOS_PER_MILLISECOND);
ALOGI("%s: Using %d ms as standby time", __func__, ms);
return milliseconds(ms);
}();
return standbyTimeInNanos;
}
// Set kEnableExtendedChannels to true to enable greater than stereo output
// for the MixerThread and device sink. Number of channels allowed is
// FCC_2 <= channels <= FCC_LIMIT.
constexpr bool kEnableExtendedChannels = true;
// Returns true if channel mask is permitted for the PCM sink in the MixerThread
/* static */
bool IAfThreadBase::isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
switch (audio_channel_mask_get_representation(channelMask)) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
// Haptic channel mask is only applicable for channel position mask.
const uint32_t channelCount = audio_channel_count_from_out_mask(
static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL));
const uint32_t maxChannelCount = kEnableExtendedChannels
? FCC_LIMIT : FCC_2;
if (channelCount < FCC_2 // mono is not supported at this time
|| channelCount > maxChannelCount) {
return false;
}
// check that channelMask is the "canonical" one we expect for the channelCount.
return audio_channel_position_mask_is_out_canonical(channelMask);
}
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
if (kEnableExtendedChannels) {
const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
if (channelCount >= FCC_2 // mono is not supported at this time
&& channelCount <= FCC_LIMIT) {
return true;
}
}
return false;
default:
return false;
}
}
// Set kEnableExtendedPrecision to true to use extended precision in MixerThread
constexpr bool kEnableExtendedPrecision = true;
// Returns true if format is permitted for the PCM sink in the MixerThread
/* static */
bool IAfThreadBase::isValidPcmSinkFormat(audio_format_t format) {
switch (format) {
case AUDIO_FORMAT_PCM_16_BIT:
return true;
case AUDIO_FORMAT_PCM_FLOAT:
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
case AUDIO_FORMAT_PCM_32_BIT:
case AUDIO_FORMAT_PCM_8_24_BIT:
return kEnableExtendedPrecision;
default:
return false;
}
}
// ----------------------------------------------------------------------------
// formatToString() needs to be exact for MediaMetrics purposes.
// Do not use media/TypeConverter.h toString().
/* static */
std::string IAfThreadBase::formatToString(audio_format_t format) {
std::string result;
FormatConverter::toString(format, result);
return result;
}
// TODO: move all toString helpers to audio.h
// under #ifdef __cplusplus #endif
static std::string patchSinksToString(const struct audio_patch *patch)
{
std::stringstream ss;
for (size_t i = 0; i < patch->num_sinks; ++i) {
if (i > 0) {
ss << "|";
}
ss << "(" << toString(patch->sinks[i].ext.device.type)
<< ", " << patch->sinks[i].ext.device.address << ")";
}
return ss.str();
}
static std::string patchSourcesToString(const struct audio_patch *patch)
{
std::stringstream ss;
for (size_t i = 0; i < patch->num_sources; ++i) {
if (i > 0) {
ss << "|";
}
ss << "(" << toString(patch->sources[i].ext.device.type)
<< ", " << patch->sources[i].ext.device.address << ")";
}
return ss.str();
}
static std::string toString(audio_latency_mode_t mode) {
// We convert to the AIDL type to print (eventually the legacy type will be removed).
const auto result = legacy2aidl_audio_latency_mode_t_AudioLatencyMode(mode);
return result.has_value() ? media::audio::common::toString(*result) : "UNKNOWN";
}
// Could be made a template, but other toString overloads for std::vector are confused.
static std::string toString(const std::vector<audio_latency_mode_t>& elements) {
std::string s("{ ");
for (const auto& e : elements) {
s.append(toString(e));
s.append(" ");
}
s.append("}");
return s;
}
static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
static void sFastTrackMultiplierInit()
{
char value[PROPERTY_VALUE_MAX];
if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
char *endptr;
unsigned long ul = strtoul(value, &endptr, 0);
if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
sFastTrackMultiplier = (int) ul;
}
}
}
// ----------------------------------------------------------------------------
#ifdef ADD_BATTERY_DATA
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
if (service == NULL) {
// it already logged
return;
}
service->addBatteryData(params);
}
#endif
// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset
struct {
// call when you acquire a partial wakelock
void acquire(const sp<IBinder> &wakeLockToken) {
pthread_mutex_lock(&mLock);
if (wakeLockToken.get() == nullptr) {
adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
} else {
if (mCount == 0) {
adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
}
++mCount;
}
pthread_mutex_unlock(&mLock);
}
// call when you release a partial wakelock.
void release(const sp<IBinder> &wakeLockToken) {
if (wakeLockToken.get() == nullptr) {
return;
}
pthread_mutex_lock(&mLock);
if (--mCount < 0) {
ALOGE("negative wakelock count");
mCount = 0;
}
pthread_mutex_unlock(&mLock);
}
// retrieves the boottime timebase offset from monotonic.
int64_t getBoottimeOffset() {
pthread_mutex_lock(&mLock);
int64_t boottimeOffset = mBoottimeOffset;
pthread_mutex_unlock(&mLock);
return boottimeOffset;
}
// Adjusts the timebase offset between TIMEBASE_MONOTONIC
// and the selected timebase.
// Currently only TIMEBASE_BOOTTIME is allowed.
//
// This only needs to be called upon acquiring the first partial wakelock
// after all other partial wakelocks are released.
//
// We do an empirical measurement of the offset rather than parsing
// /proc/timer_list since the latter is not a formal kernel ABI.
static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) {
int clockbase;
switch (timebase) {
case ExtendedTimestamp::TIMEBASE_BOOTTIME:
clockbase = SYSTEM_TIME_BOOTTIME;
break;
default:
LOG_ALWAYS_FATAL("invalid timebase %d", timebase);
break;
}
// try three times to get the clock offset, choose the one
// with the minimum gap in measurements.
const int tries = 3;
nsecs_t bestGap = 0, measured = 0; // not required, initialized for clang-tidy
for (int i = 0; i < tries; ++i) {
const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
const nsecs_t tbase = systemTime(clockbase);
const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC);
const nsecs_t gap = tmono2 - tmono;
if (i == 0 || gap < bestGap) {
bestGap = gap;
measured = tbase - ((tmono + tmono2) >> 1);
}
}
// to avoid micro-adjusting, we don't change the timebase
// unless it is significantly different.
//
// Assumption: It probably takes more than toleranceNs to
// suspend and resume the device.
static int64_t toleranceNs = 10000; // 10 us
if (llabs(*offset - measured) > toleranceNs) {
ALOGV("Adjusting timebase offset old: %lld new: %lld",
(long long)*offset, (long long)measured);
*offset = measured;
}
}
pthread_mutex_t mLock;
int32_t mCount;
int64_t mBoottimeOffset;
} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization
// ----------------------------------------------------------------------------
// CPU Stats
// ----------------------------------------------------------------------------
class CpuStats {
public:
CpuStats();
void sample(const String8 &title);
#ifdef DEBUG_CPU_USAGE
private:
ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
audio_utils::Statistics<double> mWcStats; // statistics on thread CPU usage in wall clock ns
audio_utils::Statistics<double> mHzStats; // statistics on thread CPU usage in cycles
int mCpuNum; // thread's current CPU number
int mCpukHz; // frequency of thread's current CPU in kHz
#endif
};
CpuStats::CpuStats()
#ifdef DEBUG_CPU_USAGE
: mCpuNum(-1), mCpukHz(-1)
#endif
{
}
void CpuStats::sample(const String8 &title
#ifndef DEBUG_CPU_USAGE
__unused
#endif
) {
#ifdef DEBUG_CPU_USAGE
// get current thread's delta CPU time in wall clock ns
double wcNs;
bool valid = mCpuUsage.sampleAndEnable(wcNs);
// record sample for wall clock statistics
if (valid) {
mWcStats.add(wcNs);
}
// get the current CPU number
int cpuNum = sched_getcpu();
// get the current CPU frequency in kHz
int cpukHz = mCpuUsage.getCpukHz(cpuNum);
// check if either CPU number or frequency changed
if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
mCpuNum = cpuNum;
mCpukHz = cpukHz;
// ignore sample for purposes of cycles
valid = false;
}
// if no change in CPU number or frequency, then record sample for cycle statistics
if (valid && mCpukHz > 0) {
const double cycles = wcNs * cpukHz * 0.000001;
mHzStats.add(cycles);
}
const unsigned n = mWcStats.getN();
// mCpuUsage.elapsed() is expensive, so don't call it every loop
if ((n & 127) == 1) {
const long long elapsed = mCpuUsage.elapsed();
if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
const double perLoop = elapsed / (double) n;
const double perLoop100 = perLoop * 0.01;
const double perLoop1k = perLoop * 0.001;
const double mean = mWcStats.getMean();
const double stddev = mWcStats.getStdDev();
const double minimum = mWcStats.getMin();
const double maximum = mWcStats.getMax();
const double meanCycles = mHzStats.getMean();
const double stddevCycles = mHzStats.getStdDev();
const double minCycles = mHzStats.getMin();
const double maxCycles = mHzStats.getMax();
mCpuUsage.resetElapsed();
mWcStats.reset();
mHzStats.reset();
ALOGD("CPU usage for %s over past %.1f secs\n"
" (%u mixer loops at %.1f mean ms per loop):\n"
" us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
" %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
" MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
title.c_str(),
elapsed * .000000001, n, perLoop * .000001,
mean * .001,
stddev * .001,
minimum * .001,
maximum * .001,
mean / perLoop100,
stddev / perLoop100,
minimum / perLoop100,
maximum / perLoop100,
meanCycles / perLoop1k,
stddevCycles / perLoop1k,
minCycles / perLoop1k,
maxCycles / perLoop1k);
}
}
#endif
};
// ----------------------------------------------------------------------------
// ThreadBase
// ----------------------------------------------------------------------------
// static
const char* ThreadBase::threadTypeToString(ThreadBase::type_t type)
{
switch (type) {
case MIXER:
return "MIXER";
case DIRECT:
return "DIRECT";
case DUPLICATING:
return "DUPLICATING";
case RECORD:
return "RECORD";
case OFFLOAD:
return "OFFLOAD";
case MMAP_PLAYBACK:
return "MMAP_PLAYBACK";
case MMAP_CAPTURE:
return "MMAP_CAPTURE";
case SPATIALIZER:
return "SPATIALIZER";
case BIT_PERFECT:
return "BIT_PERFECT";
default:
return "unknown";
}
}
ThreadBase::ThreadBase(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
type_t type, bool systemReady, bool isOut)
: Thread(false /*canCallJava*/),
mType(type),
mAfThreadCallback(afThreadCallback),
mThreadMetrics(std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_THREAD) + std::to_string(id),
isOut),
mIsOut(isOut),
// mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
// are set by PlaybackThread::readOutputParameters_l() or
// RecordThread::readInputParameters_l()
//FIXME: mStandby should be true here. Is this some kind of hack?
mStandby(false),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
// mName will be set by concrete (non-virtual) subclass
mDeathRecipient(new PMDeathRecipient(this)),
mSystemReady(systemReady),
mSignalPending(false)
{
mThreadMetrics.logConstructor(getpid(), threadTypeToString(type), id);
memset(&mPatch, 0, sizeof(struct audio_patch));
}
ThreadBase::~ThreadBase()
{
// mConfigEvents should be empty, but just in case it isn't, free the memory it owns
mConfigEvents.clear();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
sp<IBinder> binder = IInterface::asBinder(mPowerManager);
binder->unlinkToDeath(mDeathRecipient);
}
sendStatistics(true /* force */);
}
status_t ThreadBase::readyToRun()
{
status_t status = initCheck();
if (status == NO_ERROR) {
ALOGI("AudioFlinger's thread %p tid=%d ready to run", this, getTid());
} else {
ALOGE("No working audio driver found.");
}
return status;
}
void ThreadBase::exit()
{
ALOGV("ThreadBase::exit");
// do any cleanup required for exit to succeed
preExit();
{
// This lock prevents the following race in thread (uniprocessor for illustration):
// if (!exitPending()) {
// // context switch from here to exit()
// // exit() calls requestExit(), what exitPending() observes
// // exit() calls signal(), which is dropped since no waiters
// // context switch back from exit() to here
// mWaitWorkCV.wait(...);
// // now thread is hung
// }
audio_utils::lock_guard lock(mutex());
requestExit();
mWaitWorkCV.notify_all();
}
// When Thread::requestExitAndWait is made virtual and this method is renamed to
// "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
requestExitAndWait();
}
status_t ThreadBase::setParameters(const String8& keyValuePairs)
{
ALOGV("ThreadBase::setParameters() %s", keyValuePairs.c_str());
audio_utils::lock_guard _l(mutex());
return sendSetParameterConfigEvent_l(keyValuePairs);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
status_t ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
NO_THREAD_SAFETY_ANALYSIS // condition variable
{
status_t status = NO_ERROR;
if (event->mRequiresSystemReady && !mSystemReady) {
event->mWaitStatus = false;
mPendingConfigEvents.add(event);
return status;
}
mConfigEvents.add(event);
ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
mWaitWorkCV.notify_one();
mutex().unlock();
{
audio_utils::unique_lock _l(event->mutex());
while (event->mWaitStatus) {
if (event->mCondition.wait_for(
_l, std::chrono::nanoseconds(kConfigEventTimeoutNs), getTid())
== std::cv_status::timeout) {
event->mStatus = TIMED_OUT;
event->mWaitStatus = false;
}
}
status = event->mStatus;
}
mutex().lock();
return status;
}
void ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId)
{
audio_utils::lock_guard _l(mutex());
sendIoConfigEvent_l(event, pid, portId);
}
// sendIoConfigEvent_l() must be called with ThreadBase::mutex() held
void ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId)
{
// The audio statistics history is exponentially weighted to forget events
// about five or more seconds in the past. In order to have
// crisper statistics for mediametrics, we reset the statistics on
// an IoConfigEvent, to reflect different properties for a new device.
mIoJitterMs.reset();
mLatencyMs.reset();
mProcessTimeMs.reset();
mMonopipePipeDepthStats.reset();
mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS);
sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid, portId);
sendConfigEvent_l(configEvent);
}
void ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
audio_utils::lock_guard _l(mutex());
sendPrioConfigEvent_l(pid, tid, prio, forApp);
}
// sendPrioConfigEvent_l() must be called with ThreadBase::mutex() held
void ThreadBase::sendPrioConfigEvent_l(
pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp);
sendConfigEvent_l(configEvent);
}
// sendSetParameterConfigEvent_l() must be called with ThreadBase::mutex() held
status_t ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
{
sp<ConfigEvent> configEvent;
AudioParameter param(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyMonoOutput), value) == NO_ERROR) {
setMasterMono_l(value != 0);
if (param.size() == 1) {
return NO_ERROR; // should be a solo parameter - we don't pass down
}
param.remove(String8(AudioParameter::keyMonoOutput));
configEvent = new SetParameterConfigEvent(param.toString());
} else {
configEvent = new SetParameterConfigEvent(keyValuePair);
}
return sendConfigEvent_l(configEvent);
}
status_t ThreadBase::sendCreateAudioPatchConfigEvent(
const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
audio_utils::lock_guard _l(mutex());
sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
status_t status = sendConfigEvent_l(configEvent);
if (status == NO_ERROR) {
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)configEvent->mData.get();
*handle = data->mHandle;
}
return status;
}
status_t ThreadBase::sendReleaseAudioPatchConfigEvent(
const audio_patch_handle_t handle)
{
audio_utils::lock_guard _l(mutex());
sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
return sendConfigEvent_l(configEvent);
}
status_t ThreadBase::sendUpdateOutDeviceConfigEvent(
const DeviceDescriptorBaseVector& outDevices)
{
if (type() != RECORD) {
// The update out device operation is only for record thread.
return INVALID_OPERATION;
}
audio_utils::lock_guard _l(mutex());
sp<ConfigEvent> configEvent = (ConfigEvent *)new UpdateOutDevicesConfigEvent(outDevices);
return sendConfigEvent_l(configEvent);
}
void ThreadBase::sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
{
ALOG_ASSERT(type() == RECORD, "sendResizeBufferConfigEvent_l() called on non record thread");
sp<ConfigEvent> configEvent =
(ConfigEvent *)new ResizeBufferConfigEvent(maxSharedAudioHistoryMs);
sendConfigEvent_l(configEvent);
}
void ThreadBase::sendCheckOutputStageEffectsEvent()
{
audio_utils::lock_guard _l(mutex());
sendCheckOutputStageEffectsEvent_l();
}
void ThreadBase::sendCheckOutputStageEffectsEvent_l()
{
sp<ConfigEvent> configEvent =
(ConfigEvent *)new CheckOutputStageEffectsEvent();
sendConfigEvent_l(configEvent);
}
void ThreadBase::sendHalLatencyModesChangedEvent_l()
{
sp<ConfigEvent> configEvent = sp<HalLatencyModesChangedEvent>::make();
sendConfigEvent_l(configEvent);
}
// post condition: mConfigEvents.isEmpty()
void ThreadBase::processConfigEvents_l()
{
bool configChanged = false;
while (!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
sp<ConfigEvent> event = mConfigEvents[0];
mConfigEvents.removeAt(0);
switch (event->mType) {
case CFG_EVENT_PRIO: {
PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
// FIXME Need to understand why this has to be done asynchronously
int err = requestPriority(data->mPid, data->mTid, data->mPrio, data->mForApp,
true /*asynchronous*/);
if (err != 0) {
ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
data->mPrio, data->mPid, data->mTid, err);
}
} break;
case CFG_EVENT_IO: {
IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
ioConfigChanged_l(data->mEvent, data->mPid, data->mPortId);
} break;
case CFG_EVENT_SET_PARAMETER: {
SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
configChanged = true;
mLocalLog.log("CFG_EVENT_SET_PARAMETER: (%s) configuration changed",
data->mKeyValuePairs.c_str());
}
} break;
case CFG_EVENT_CREATE_AUDIO_PATCH: {
const DeviceTypeSet oldDevices = getDeviceTypes_l();
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)event->mData.get();
event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
const DeviceTypeSet newDevices = getDeviceTypes_l();
configChanged = oldDevices != newDevices;
mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
dumpDeviceTypes(newDevices).c_str(), toString(newDevices).c_str());
} break;
case CFG_EVENT_RELEASE_AUDIO_PATCH: {
const DeviceTypeSet oldDevices = getDeviceTypes_l();
ReleaseAudioPatchConfigEventData *data =
(ReleaseAudioPatchConfigEventData *)event->mData.get();
event->mStatus = releaseAudioPatch_l(data->mHandle);
const DeviceTypeSet newDevices = getDeviceTypes_l();
configChanged = oldDevices != newDevices;
mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
dumpDeviceTypes(newDevices).c_str(), toString(newDevices).c_str());
} break;
case CFG_EVENT_UPDATE_OUT_DEVICE: {
UpdateOutDevicesConfigEventData *data =
(UpdateOutDevicesConfigEventData *)event->mData.get();
updateOutDevices(data->mOutDevices);
} break;
case CFG_EVENT_RESIZE_BUFFER: {
ResizeBufferConfigEventData *data =
(ResizeBufferConfigEventData *)event->mData.get();
resizeInputBuffer_l(data->mMaxSharedAudioHistoryMs);
} break;
case CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS: {
setCheckOutputStageEffects();
} break;
case CFG_EVENT_HAL_LATENCY_MODES_CHANGED: {
onHalLatencyModesChanged_l();
} break;
default:
ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
break;
}
{
audio_utils::lock_guard _l(event->mutex());
if (event->mWaitStatus) {
event->mWaitStatus = false;
event->mCondition.notify_one();
}
}
ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
}
if (configChanged) {
cacheParameters_l();
}
}
String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
String8 s;
const audio_channel_representation_t representation =
audio_channel_mask_get_representation(mask);
switch (representation) {
// Travel all single bit channel mask to convert channel mask to string.
case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
if (output) {
if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low-frequency, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT) s.append("top-side-left, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT) s.append("top-side-right, ");
if (mask & AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT) s.append("bottom-front-left, ");
if (mask & AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER) s.append("bottom-front-center, ");
if (mask & AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT) s.append("bottom-front-right, ");
if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) s.append("low-frequency-2, ");
if (mask & AUDIO_CHANNEL_OUT_HAPTIC_B) s.append("haptic-B, ");
if (mask & AUDIO_CHANNEL_OUT_HAPTIC_A) s.append("haptic-A, ");
if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
} else {
if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
if (mask & AUDIO_CHANNEL_IN_BACK_LEFT) s.append("back-left, ");
if (mask & AUDIO_CHANNEL_IN_BACK_RIGHT) s.append("back-right, ");
if (mask & AUDIO_CHANNEL_IN_CENTER) s.append("center, ");
if (mask & AUDIO_CHANNEL_IN_LOW_FREQUENCY) s.append("low-frequency, ");
if (mask & AUDIO_CHANNEL_IN_TOP_LEFT) s.append("top-left, ");
if (mask & AUDIO_CHANNEL_IN_TOP_RIGHT) s.append("top-right, ");
if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
}
const int len = s.length();
if (len > 2) {
(void) s.lockBuffer(len); // needed?
s.unlockBuffer(len - 2); // remove trailing ", "
}
return s;
}
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
return s;
default:
s.appendFormat("unknown mask, representation:%d bits:%#x",
representation, audio_channel_mask_get_bits(mask));
return s;
}
}
void ThreadBase::dump(int fd, const Vector<String16>& args)
NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
this, mThreadName, getTid(), type(), threadTypeToString(type()));
const bool locked = afutils::dumpTryLock(mutex());
if (!locked) {
dprintf(fd, " Thread may be deadlocked\n");
}
dumpBase_l(fd, args);
dumpInternals_l(fd, args);
dumpTracks_l(fd, args);
dumpEffectChains_l(fd, args);
if (locked) {
mutex().unlock();
}
dprintf(fd, " Local log:\n");
mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
// --all does the statistics
bool dumpAll = false;
for (const auto &arg : args) {
if (arg == String16("--all")) {
dumpAll = true;
}
}
if (dumpAll || type() == SPATIALIZER) {
const std::string sched = mThreadSnapshot.toString();
if (!sched.empty()) {
(void)write(fd, sched.c_str(), sched.size());
}
}
}
void ThreadBase::dumpBase_l(int fd, const Vector<String16>& /* args */)
{
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat,
IAfThreadBase::formatToString(mHALFormat).c_str());
dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize);
dprintf(fd, " Channel count: %u\n", mChannelCount);
dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).c_str());
dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat,
IAfThreadBase::formatToString(mFormat).c_str());
dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize);
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
const size_t SIZE = 256;
char buffer[SIZE];
for (size_t i = 0; i < numConfig; i++) {
mConfigEvents[i]->dump(buffer, SIZE);
dprintf(fd, "\n %s", buffer);
}
dprintf(fd, "\n");
} else {
dprintf(fd, " none\n");
}
// Note: output device may be used by capture threads for effects such as AEC.
dprintf(fd, " Output devices: %s (%s)\n",
dumpDeviceTypes(outDeviceTypes_l()).c_str(), toString(outDeviceTypes_l()).c_str());
dprintf(fd, " Input device: %#x (%s)\n",
inDeviceType_l(), toString(inDeviceType_l()).c_str());
dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, toString(mAudioSource).c_str());
// Dump timestamp statistics for the Thread types that support it.
if (mType == RECORD
|| mType == MIXER
|| mType == DUPLICATING
|| mType == DIRECT
|| mType == OFFLOAD
|| mType == SPATIALIZER) {
dprintf(fd, " Timestamp stats: %s\n", mTimestampVerifier.toString().c_str());
dprintf(fd, " Timestamp corrected: %s\n",
isTimestampCorrectionEnabled_l() ? "yes" : "no");
}
if (mLastIoBeginNs > 0) { // MMAP may not set this
dprintf(fd, " Last %s occurred (msecs): %lld\n",
isOutput() ? "write" : "read",
(long long) (systemTime() - mLastIoBeginNs) / NANOS_PER_MILLISECOND);
}
if (mProcessTimeMs.getN() > 0) {
dprintf(fd, " Process time ms stats: %s\n", mProcessTimeMs.toString().c_str());
}
if (mIoJitterMs.getN() > 0) {
dprintf(fd, " Hal %s jitter ms stats: %s\n",
isOutput() ? "write" : "read",
mIoJitterMs.toString().c_str());
}
if (mLatencyMs.getN() > 0) {
dprintf(fd, " Threadloop %s latency stats: %s\n",
isOutput() ? "write" : "read",
mLatencyMs.toString().c_str());
}
if (mMonopipePipeDepthStats.getN() > 0) {
dprintf(fd, " Monopipe %s pipe depth stats: %s\n",
isOutput() ? "write" : "read",
mMonopipePipeDepthStats.toString().c_str());
}
}
void ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
size_t numEffectChains = mEffectChains.size();
snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < numEffectChains; ++i) {
sp<IAfEffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
}
}
}
void ThreadBase::acquireWakeLock()
{
audio_utils::lock_guard _l(mutex());
acquireWakeLock_l();
}
String16 ThreadBase::getWakeLockTag()
{
switch (mType) {
case MIXER:
return String16("AudioMix");
case DIRECT:
return String16("AudioDirectOut");
case DUPLICATING:
return String16("AudioDup");
case RECORD:
return String16("AudioIn");
case OFFLOAD:
return String16("AudioOffload");
case MMAP_PLAYBACK:
return String16("MmapPlayback");
case MMAP_CAPTURE:
return String16("MmapCapture");
case SPATIALIZER:
return String16("AudioSpatial");
default:
ALOG_ASSERT(false);
return String16("AudioUnknown");
}
}
void ThreadBase::acquireWakeLock_l()
{
getPowerManager_l();
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
// Uses AID_AUDIOSERVER for wakelock. updateWakeLockUids_l() updates with client uids.
binder::Status status = mPowerManager->acquireWakeLockAsync(binder,
POWERMANAGER_PARTIAL_WAKE_LOCK,
getWakeLockTag(),
String16("audioserver"),
{} /* workSource */,
{} /* historyTag */);
if (status.isOk()) {
mWakeLockToken = binder;
}
ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status.exceptionCode());
}
gBoottime.acquire(mWakeLockToken);
mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] =
gBoottime.getBoottimeOffset();
}
void ThreadBase::releaseWakeLock()
{
audio_utils::lock_guard _l(mutex());
releaseWakeLock_l();
}
void ThreadBase::releaseWakeLock_l()
{
gBoottime.release(mWakeLockToken);
if (mWakeLockToken != 0) {
ALOGV("releaseWakeLock_l() %s", mThreadName);
if (mPowerManager != 0) {
mPowerManager->releaseWakeLockAsync(mWakeLockToken, 0);
}
mWakeLockToken.clear();
}
}
void ThreadBase::getPowerManager_l() {
if (mSystemReady && mPowerManager == 0) {
// use checkService() to avoid blocking if power service is not up yet
sp<IBinder> binder =
defaultServiceManager()->checkService(String16("power"));
if (binder == 0) {
ALOGW("Thread %s cannot connect to the power manager service", mThreadName);
} else {
mPowerManager = interface_cast<os::IPowerManager>(binder);
binder->linkToDeath(mDeathRecipient);
}
}
}
void ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t>& uids) {
getPowerManager_l();
#if !LOG_NDEBUG
std::stringstream s;
for (uid_t uid : uids) {
s << uid << " ";
}
ALOGD("updateWakeLockUids_l %s uids:%s", mThreadName, s.str().c_str());
#endif
if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
if (mSystemReady) {
ALOGE("no wake lock to update, but system ready!");
} else {
ALOGW("no wake lock to update, system not ready yet");
}
return;
}
if (mPowerManager != 0) {
std::vector<int> uidsAsInt(uids.begin(), uids.end()); // powermanager expects uids as ints
binder::Status status = mPowerManager->updateWakeLockUidsAsync(
mWakeLockToken, uidsAsInt);
ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status.exceptionCode());
}
}
void ThreadBase::clearPowerManager()
{
audio_utils::lock_guard _l(mutex());
releaseWakeLock_l();
mPowerManager.clear();
}
void ThreadBase::updateOutDevices(
const DeviceDescriptorBaseVector& outDevices __unused)
{
ALOGE("%s should only be called in RecordThread", __func__);
}
void ThreadBase::resizeInputBuffer_l(int32_t /* maxSharedAudioHistoryMs */)
{
ALOGE("%s should only be called in RecordThread", __func__);
}
void ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& /* who */)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
thread->clearPowerManager();
}
ALOGW("power manager service died !!!");
}
void ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
if (type != NULL) {
chain->setEffectSuspended_l(type, suspend);
} else {
chain->setEffectSuspendedAll_l(suspend);
}
}
updateSuspendedSessions_l(type, suspend, sessionId);
}
void ThreadBase::checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain)
{
ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
if (index < 0) {
return;
}
const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
mSuspendedSessions.valueAt(index);
for (size_t i = 0; i < sessionEffects.size(); i++) {
const sp<SuspendedSessionDesc>& desc = sessionEffects.valueAt(i);
for (int j = 0; j < desc->mRefCount; j++) {
if (sessionEffects.keyAt(i) == IAfEffectChain::kKeyForSuspendAll) {
chain->setEffectSuspendedAll_l(true);
} else {
ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
desc->mType.timeLow);
chain->setEffectSuspended_l(&desc->mType, true);
}
}
}
}
void ThreadBase::updateSuspendedSessions_l(const effect_uuid_t* type,
bool suspend,
audio_session_t sessionId)
{
ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
if (suspend) {
if (index >= 0) {
sessionEffects = mSuspendedSessions.valueAt(index);
} else {
mSuspendedSessions.add(sessionId, sessionEffects);
}
} else {
if (index < 0) {
return;
}
sessionEffects = mSuspendedSessions.valueAt(index);
}
int key = IAfEffectChain::kKeyForSuspendAll;
if (type != NULL) {
key = type->timeLow;
}
index = sessionEffects.indexOfKey(key);
sp<SuspendedSessionDesc> desc;
if (suspend) {
if (index >= 0) {
desc = sessionEffects.valueAt(index);
} else {
desc = new SuspendedSessionDesc();
if (type != NULL) {
desc->mType = *type;
}
sessionEffects.add(key, desc);
ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
}
desc->mRefCount++;
} else {
if (index < 0) {
return;
}
desc = sessionEffects.valueAt(index);
if (--desc->mRefCount == 0) {
ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
sessionEffects.removeItemsAt(index);
if (sessionEffects.isEmpty()) {
ALOGV("updateSuspendedSessions_l() restore removing session %d",
sessionId);
mSuspendedSessions.removeItem(sessionId);
}
}
}
if (!sessionEffects.isEmpty()) {
mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
}
}
void ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
audio_session_t sessionId,
bool threadLocked)
NO_THREAD_SAFETY_ANALYSIS // manual locking
{
if (!threadLocked) {
mutex().lock();
}
if (mType != RECORD) {
// suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
// another session. This gives the priority to well behaved effect control panels
// and applications not using global effects.
// Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
// global effects
if (!audio_is_global_session(sessionId)) {
setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
}
}
if (!threadLocked) {
mutex().unlock();
}
}
// checkEffectCompatibility_l() must be called with ThreadBase::mutex() held
status_t RecordThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global output effect sessions on record threads
if (sessionId == AUDIO_SESSION_OUTPUT_MIX
|| sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
// only pre processing effects on record thread
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) {
ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
// always allow effects without processing load or latency
if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
return NO_ERROR;
}
audio_input_flags_t flags = mInput->flags;
if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) {
if (flags & AUDIO_INPUT_FLAG_RAW) {
ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode",
desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode",
desc->name, mThreadName);
return BAD_VALUE;
}
}
if (IAfEffectModule::isHapticGenerator(&desc->type)) {
ALOGE("%s(): HapticGenerator is not supported in RecordThread", __func__);
return BAD_VALUE;
}
return NO_ERROR;
}
// checkEffectCompatibility_l() must be called with ThreadBase::mutex() held
status_t PlaybackThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// no preprocessing on playback threads
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) {
ALOGW("%s: pre processing effect %s created on playback"
" thread %s", __func__, desc->name, mThreadName);
return BAD_VALUE;
}
// always allow effects without processing load or latency
if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
return NO_ERROR;
}
if (IAfEffectModule::isHapticGenerator(&desc->type) && mHapticChannelCount == 0) {
ALOGW("%s: thread doesn't support haptic playback while the effect is HapticGenerator",
__func__);
return BAD_VALUE;
}
if (IAfEffectModule::isSpatializer(&desc->type)
&& mType != SPATIALIZER) {
ALOGW("%s: attempt to create a spatializer effect on a thread of type %d",
__func__, mType);
return BAD_VALUE;
}
switch (mType) {
case MIXER: {
audio_output_flags_t flags = mOutput->flags;
if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) {
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// global effects are applied only to non fast tracks if they are SW
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
break;
}
} else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// only post processing on output stage session
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("%s: non post processing effect %s not allowed on output stage session",
__func__, desc->name);
return BAD_VALUE;
}
} else if (sessionId == AUDIO_SESSION_DEVICE) {
// only post processing on output stage session
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("%s: non post processing effect %s not allowed on device session",
__func__, desc->name);
return BAD_VALUE;
}
} else {
// no restriction on effects applied on non fast tracks
if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) {
break;
}
}
if (flags & AUDIO_OUTPUT_FLAG_RAW) {
ALOGW("%s: effect %s on playback thread in raw mode", __func__, desc->name);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
ALOGW("%s: non HW effect %s on playback thread in fast mode",
__func__, desc->name);
return BAD_VALUE;
}
}
} break;
case OFFLOAD:
// nothing actionable on offload threads, if the effect:
// - is offloadable: the effect can be created
// - is NOT offloadable: the effect should still be created, but EffectHandle::enable()
// will take care of invalidating the tracks of the thread
break;
case DIRECT:
// Reject any effect on Direct output threads for now, since the format of
// mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
ALOGW("%s: effect %s on DIRECT output thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
case DUPLICATING:
if (audio_is_global_session(sessionId)) {
ALOGW("%s: global effect %s on DUPLICATING thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("%s: post processing effect %s on DUPLICATING thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
ALOGW("%s: HW tunneled effect %s on DUPLICATING thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
}
break;
case SPATIALIZER:
// Global effects (AUDIO_SESSION_OUTPUT_MIX) are not supported on spatializer mixer
// as there is no common accumulation buffer for sptialized and non sptialized tracks.
// Post processing effects (AUDIO_SESSION_OUTPUT_STAGE or AUDIO_SESSION_DEVICE)
// are supported and added after the spatializer.
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
ALOGW("%s: global effect %s not supported on spatializer thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
} else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// only post processing , downmixer or spatializer effects on output stage session
if (IAfEffectModule::isSpatializer(&desc->type)
|| memcmp(&desc->type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
break;
}
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("%s: non post processing effect %s not allowed on output stage session",
__func__, desc->name);
return BAD_VALUE;
}
} else if (sessionId == AUDIO_SESSION_DEVICE) {
// only post processing on output stage session
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("%s: non post processing effect %s not allowed on device session",
__func__, desc->name);
return BAD_VALUE;
}
}
break;
case BIT_PERFECT:
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
// Allow HW accelerated effects of tunnel type
break;
}
// As bit-perfect tracks will not be allowed to apply audio effect that will touch the audio
// data, effects will not be allowed on 1) global effects (AUDIO_SESSION_OUTPUT_MIX),
// 2) post-processing effects (AUDIO_SESSION_OUTPUT_STAGE or AUDIO_SESSION_DEVICE) and
// 3) there is any bit-perfect track with the given session id.
if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE ||
sessionId == AUDIO_SESSION_DEVICE) {
ALOGW("%s: effect %s not supported on bit-perfect thread %s",
__func__, desc->name, mThreadName);
return BAD_VALUE;
} else if ((hasAudioSession_l(sessionId) & ThreadBase::BIT_PERFECT_SESSION) != 0) {
ALOGW("%s: effect %s not supported as there is a bit-perfect track with session as %d",
__func__, desc->name, sessionId);
return BAD_VALUE;
}
break;
default:
LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
}
return NO_ERROR;
}
// ThreadBase::createEffect_l() must be called with AudioFlinger::mutex() held
sp<IAfEffectHandle> ThreadBase::createEffect_l(
const sp<Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_session_t sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status,
bool pinned,
bool probe,
bool notifyFramesProcessed)
{
sp<IAfEffectModule> effect;
sp<IAfEffectHandle> handle;
status_t lStatus;
sp<IAfEffectChain> chain;
bool chainCreated = false;
bool effectCreated = false;
audio_unique_id_t effectId = AUDIO_UNIQUE_ID_USE_UNSPECIFIED;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGW("createEffect_l() Audio driver not initialized.");
goto Exit;
}
ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
{ // scope for mutex()
audio_utils::lock_guard _l(mutex());
lStatus = checkEffectCompatibility_l(desc, sessionId);
if (probe || lStatus != NO_ERROR) {
goto Exit;
}
// check for existing effect chain with the requested audio session
chain = getEffectChain_l(sessionId);
if (chain == 0) {
// create a new chain for this session
ALOGV("createEffect_l() new effect chain for session %d", sessionId);
chain = IAfEffectChain::create(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
} else {
effect = chain->getEffectFromDesc_l(desc);
}
ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
if (effect == 0) {
effectId = mAfThreadCallback->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
// create a new effect module if none present in the chain
lStatus = chain->createEffect_l(effect, desc, effectId, sessionId, pinned);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectCreated = true;
// FIXME: use vector of device and address when effect interface is ready.
effect->setDevices(outDeviceTypeAddrs());
effect->setInputDevice(inDeviceTypeAddr());
effect->setMode(mAfThreadCallback->getMode());
effect->setAudioSource(mAudioSource);
}
if (effect->isHapticGenerator()) {
// TODO(b/184194057): Use the vibrator information from the vibrator that will be used
// for the HapticGenerator.
const std::optional<media::AudioVibratorInfo> defaultVibratorInfo =
std::move(mAfThreadCallback->getDefaultVibratorInfo_l());
if (defaultVibratorInfo) {
// Only set the vibrator info when it is a valid one.
effect->setVibratorInfo_l(*defaultVibratorInfo);
}
}
// create effect handle and connect it to effect module
handle = IAfEffectHandle::create(
effect, client, effectClient, priority, notifyFramesProcessed);
lStatus = handle->initCheck();
if (lStatus == OK) {
lStatus = effect->addHandle(handle.get());
sendCheckOutputStageEffectsEvent_l();
}
if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
}
}
Exit:
if (!probe && lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
audio_utils::lock_guard _l(mutex());
if (effectCreated) {
chain->removeEffect_l(effect);
}
if (chainCreated) {
removeEffectChain_l(chain);
}
// handle must be cleared by caller to avoid deadlock.
}
*status = lStatus;
return handle;
}
void ThreadBase::disconnectEffectHandle(IAfEffectHandle* handle,
bool unpinIfLast)
{
bool remove = false;
sp<IAfEffectModule> effect;
{
audio_utils::lock_guard _l(mutex());
sp<IAfEffectBase> effectBase = handle->effect().promote();
if (effectBase == nullptr) {
return;
}
effect = effectBase->asEffectModule();
if (effect == nullptr) {
return;
}
// restore suspended effects if the disconnected handle was enabled and the last one.
remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast);
if (remove) {
removeEffect_l(effect, true);
}
sendCheckOutputStageEffectsEvent_l();
}
if (remove) {
mAfThreadCallback->updateOrphanEffectChains(effect);
if (handle->enabled()) {
effect->checkSuspendOnEffectEnabled(false, false /*threadLocked*/);
}
}
}
void ThreadBase::onEffectEnable(const sp<IAfEffectModule>& effect) {
if (isOffloadOrMmap()) {
audio_utils::lock_guard _l(mutex());
broadcast_l();
}
if (!effect->isOffloadable()) {
if (mType == ThreadBase::OFFLOAD) {
PlaybackThread *t = (PlaybackThread *)this;
t->invalidateTracks(AUDIO_STREAM_MUSIC);
}
if (effect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
mAfThreadCallback->onNonOffloadableGlobalEffectEnable();
}
}
}
void ThreadBase::onEffectDisable() {
if (isOffloadOrMmap()) {
audio_utils::lock_guard _l(mutex());
broadcast_l();
}
}
sp<IAfEffectModule> ThreadBase::getEffect(audio_session_t sessionId,
int effectId) const
{
audio_utils::lock_guard _l(mutex());
return getEffect_l(sessionId, effectId);
}
sp<IAfEffectModule> ThreadBase::getEffect_l(audio_session_t sessionId,
int effectId) const
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
std::vector<int> ThreadBase::getEffectIds_l(audio_session_t sessionId) const
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
return chain != nullptr ? chain->getEffectIds_l() : std::vector<int>{};
}
// PlaybackThread::addEffect_ll() must be called with AudioFlinger::mutex() and
// ThreadBase::mutex() held
status_t ThreadBase::addEffect_ll(const sp<IAfEffectModule>& effect)
{
// check for existing effect chain with the requested audio session
audio_session_t sessionId = effect->sessionId();
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
bool chainCreated = false;
ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
"%s: on offloaded thread %p: effect %s does not support offload flags %#x",
__func__, this, effect->desc().name, effect->desc().flags);
if (chain == 0) {
// create a new chain for this session
ALOGV("%s: new effect chain for session %d", __func__, sessionId);
chain = IAfEffectChain::create(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
}
ALOGV("%s: %p chain %p effect %p", __func__, this, chain.get(), effect.get());
if (chain->getEffectFromId_l(effect->id()) != 0) {
ALOGW("%s: %p effect %s already present in chain %p",
__func__, this, effect->desc().name, chain.get());
return BAD_VALUE;
}
effect->setOffloaded_l(mType == OFFLOAD, mId);
status_t status = chain->addEffect_l(effect);
if (status != NO_ERROR) {
if (chainCreated) {
removeEffectChain_l(chain);
}
return status;
}
effect->setDevices(outDeviceTypeAddrs());
effect->setInputDevice(inDeviceTypeAddr());
effect->setMode(mAfThreadCallback->getMode());
effect->setAudioSource(mAudioSource);
return NO_ERROR;
}
void ThreadBase::removeEffect_l(const sp<IAfEffectModule>& effect, bool release) {
ALOGV("%s %p effect %p", __FUNCTION__, this, effect.get());
effect_descriptor_t desc = effect->desc();
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
detachAuxEffect_l(effect->id());
}
sp<IAfEffectChain> chain = effect->getCallback()->chain().promote();
if (chain != 0) {
// remove effect chain if removing last effect
if (chain->removeEffect_l(effect, release) == 0) {
removeEffectChain_l(chain);
}
} else {
ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
}
}
void ThreadBase::lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains)
NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::lock()
{
effectChains = mEffectChains;
for (const auto& effectChain : effectChains) {
effectChain->mutex().lock();
}
}
void ThreadBase::unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains)
NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::unlock()
{
for (const auto& effectChain : effectChains) {
effectChain->mutex().unlock();
}
}
sp<IAfEffectChain> ThreadBase::getEffectChain(audio_session_t sessionId) const
{
audio_utils::lock_guard _l(mutex());
return getEffectChain_l(sessionId);
}
sp<IAfEffectChain> ThreadBase::getEffectChain_l(audio_session_t sessionId)
const
{
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
return mEffectChains[i];
}
}
return 0;
}
void ThreadBase::setMode(audio_mode_t mode)
{
audio_utils::lock_guard _l(mutex());
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
mEffectChains[i]->setMode_l(mode);
}
}
void ThreadBase::toAudioPortConfig(struct audio_port_config* config)
{
config->type = AUDIO_PORT_TYPE_MIX;
config->ext.mix.handle = mId;
config->sample_rate = mSampleRate;
config->format = mHALFormat;
config->channel_mask = mChannelMask;
config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT;
}
void ThreadBase::systemReady()
{
audio_utils::lock_guard _l(mutex());
if (mSystemReady) {
return;
}
mSystemReady = true;
for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
}
mPendingConfigEvents.clear();
}
template <typename T>
ssize_t ThreadBase::ActiveTracks<T>::add(const sp<T>& track) {
ssize_t index = mActiveTracks.indexOf(track);
if (index >= 0) {
ALOGW("ActiveTracks<T>::add track %p already there", track.get());
return index;
}
logTrack("add", track);
mActiveTracksGeneration++;
mLatestActiveTrack = track;
track->beginBatteryAttribution();
mHasChanged = true;
return mActiveTracks.add(track);
}
template <typename T>
ssize_t ThreadBase::ActiveTracks<T>::remove(const sp<T>& track) {
ssize_t index = mActiveTracks.remove(track);
if (index < 0) {
ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get());
return index;
}
logTrack("remove", track);
mActiveTracksGeneration++;
track->endBatteryAttribution();
// mLatestActiveTrack is not cleared even if is the same as track.
mHasChanged = true;
#ifdef TEE_SINK
track->dumpTee(-1 /* fd */, "_REMOVE");
#endif
track->logEndInterval(); // log to MediaMetrics
return index;
}
template <typename T>
void ThreadBase::ActiveTracks<T>::clear() {
for (const sp<T> &track : mActiveTracks) {
track->endBatteryAttribution();
logTrack("clear", track);
}
mLastActiveTracksGeneration = mActiveTracksGeneration;
if (!mActiveTracks.empty()) { mHasChanged = true; }
mActiveTracks.clear();
mLatestActiveTrack.clear();
}
template <typename T>
void ThreadBase::ActiveTracks<T>::updatePowerState_l(
const sp<ThreadBase>& thread, bool force) {
// Updates ActiveTracks client uids to the thread wakelock.
if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
thread->updateWakeLockUids_l(getWakeLockUids());
mLastActiveTracksGeneration = mActiveTracksGeneration;
}
}
template <typename T>
bool ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
bool hasChanged = mHasChanged;
mHasChanged = false;
for (const sp<T> &track : mActiveTracks) {
// Do not short-circuit as all hasChanged states must be reset
// as all the metadata are going to be sent
hasChanged |= track->readAndClearHasChanged();
}
return hasChanged;
}
template <typename T>
void ThreadBase::ActiveTracks<T>::logTrack(
const char *funcName, const sp<T> &track) const {
if (mLocalLog != nullptr) {
String8 result;
track->appendDump(result, false /* active */);
mLocalLog->log("AT::%-10s(%p) %s", funcName, track.get(), result.c_str());
}
}
void ThreadBase::broadcast_l()
{
// Thread could be blocked waiting for async
// so signal it to handle state changes immediately
// If threadLoop is currently unlocked a signal of mWaitWorkCV will
// be lost so we also flag to prevent it blocking on mWaitWorkCV
mSignalPending = true;
mWaitWorkCV.notify_all();
}
// Call only from threadLoop() or when it is idle.
// Do not call from high performance code as this may do binder rpc to the MediaMetrics service.
void ThreadBase::sendStatistics(bool force)
NO_THREAD_SAFETY_ANALYSIS
{
// Do not log if we have no stats.
// We choose the timestamp verifier because it is the most likely item to be present.
const int64_t nstats = mTimestampVerifier.getN() - mLastRecordedTimestampVerifierN;
if (nstats == 0) {
return;
}
// Don't log more frequently than once per 12 hours.
// We use BOOTTIME to include suspend time.
const int64_t timeNs = systemTime(SYSTEM_TIME_BOOTTIME);
const int64_t sinceNs = timeNs - mLastRecordedTimeNs; // ok if mLastRecordedTimeNs = 0
if (!force && sinceNs <= 12 * NANOS_PER_HOUR) {
return;
}
mLastRecordedTimestampVerifierN = mTimestampVerifier.getN();
mLastRecordedTimeNs = timeNs;
std::unique_ptr<mediametrics::Item> item(mediametrics::Item::create("audiothread"));
#define MM_PREFIX "android.media.audiothread." // avoid cut-n-paste errors.
// thread configuration
item->setInt32(MM_PREFIX "id", (int32_t)mId); // IO handle
// item->setInt32(MM_PREFIX "portId", (int32_t)mPortId);
item->setCString(MM_PREFIX "type", threadTypeToString(mType));
item->setInt32(MM_PREFIX "sampleRate", (int32_t)mSampleRate);
item->setInt64(MM_PREFIX "channelMask", (int64_t)mChannelMask);
item->setCString(MM_PREFIX "encoding", toString(mFormat).c_str());
item->setInt32(MM_PREFIX "frameCount", (int32_t)mFrameCount);
item->setCString(MM_PREFIX "outDevice", toString(outDeviceTypes_l()).c_str());
item->setCString(MM_PREFIX "inDevice", toString(inDeviceType_l()).c_str());
// thread statistics
if (mIoJitterMs.getN() > 0) {
item->setDouble(MM_PREFIX "ioJitterMs.mean", mIoJitterMs.getMean());
item->setDouble(MM_PREFIX "ioJitterMs.std", mIoJitterMs.getStdDev());
}
if (mProcessTimeMs.getN() > 0) {
item->setDouble(MM_PREFIX "processTimeMs.mean", mProcessTimeMs.getMean());
item->setDouble(MM_PREFIX "processTimeMs.std", mProcessTimeMs.getStdDev());
}
const auto tsjitter = mTimestampVerifier.getJitterMs();
if (tsjitter.getN() > 0) {
item->setDouble(MM_PREFIX "timestampJitterMs.mean", tsjitter.getMean());
item->setDouble(MM_PREFIX "timestampJitterMs.std", tsjitter.getStdDev());
}
if (mLatencyMs.getN() > 0) {
item->setDouble(MM_PREFIX "latencyMs.mean", mLatencyMs.getMean());
item->setDouble(MM_PREFIX "latencyMs.std", mLatencyMs.getStdDev());
}
if (mMonopipePipeDepthStats.getN() > 0) {
item->setDouble(MM_PREFIX "monopipePipeDepthStats.mean",
mMonopipePipeDepthStats.getMean());
item->setDouble(MM_PREFIX "monopipePipeDepthStats.std",
mMonopipePipeDepthStats.getStdDev());
}
item->selfrecord();
}
product_strategy_t ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
{
if (!mAfThreadCallback->isAudioPolicyReady()) {
return PRODUCT_STRATEGY_NONE;
}
return AudioSystem::getStrategyForStream(stream);
}
// startMelComputation_l() must be called with AudioFlinger::mutex() held
void ThreadBase::startMelComputation_l(
const sp<audio_utils::MelProcessor>& /*processor*/)
{
// Do nothing
ALOGW("%s: ThreadBase does not support CSD", __func__);
}
// stopMelComputation_l() must be called with AudioFlinger::mutex() held
void ThreadBase::stopMelComputation_l()
{
// Do nothing
ALOGW("%s: ThreadBase does not support CSD", __func__);
}
// ----------------------------------------------------------------------------
// Playback
// ----------------------------------------------------------------------------
PlaybackThread::PlaybackThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output,
audio_io_handle_t id,
type_t type,
bool systemReady,
audio_config_base_t *mixerConfig)
: ThreadBase(afThreadCallback, id, type, systemReady, true /* isOut */),
mNormalFrameCount(0), mSinkBuffer(NULL),
mMixerBufferEnabled(kEnableExtendedPrecision || type == SPATIALIZER),
mMixerBuffer(NULL),
mMixerBufferSize(0),
mMixerBufferFormat(AUDIO_FORMAT_INVALID),
mMixerBufferValid(false),
mEffectBufferEnabled(kEnableExtendedPrecision || type == SPATIALIZER),
mEffectBuffer(NULL),
mEffectBufferSize(0),
mEffectBufferFormat(AUDIO_FORMAT_INVALID),
mEffectBufferValid(false),
mSuspended(0), mBytesWritten(0),
mFramesWritten(0),
mSuspendedFrames(0),
mActiveTracks(&this->mLocalLog),
// mStreamTypes[] initialized in constructor body
mTracks(type == MIXER),
mOutput(output),
mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
mMixerStatus(MIXER_IDLE),
mMixerStatusIgnoringFastTracks(MIXER_IDLE),
mStandbyDelayNs(getStandbyTimeInNanos()),
mBytesRemaining(0),
mCurrentWriteLength(0),
mUseAsyncWrite(false),
mWriteAckSequence(0),
mDrainSequence(0),
mScreenState(mAfThreadCallback->getScreenState()),
// index 0 is reserved for normal mixer's submix
mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
mHwSupportsPause(false), mHwPaused(false), mFlushPending(false),
mLeftVolFloat(-1.0), mRightVolFloat(-1.0),
mDownStreamPatch{},
mIsTimestampAdvancing(kMinimumTimeBetweenTimestampChecksNs)
{
snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
mNBLogWriter = afThreadCallback->newWriter_l(kLogSize, mThreadName);
// Assumes constructor is called by AudioFlinger with its mutex() held, but
// it would be safer to explicitly pass initial masterVolume/masterMute as
// parameter.
//
// If the HAL we are using has support for master volume or master mute,
// then do not attenuate or mute during mixing (just leave the volume at 1.0
// and the mute set to false).
mMasterVolume = afThreadCallback->masterVolume_l();
mMasterMute = afThreadCallback->masterMute_l();
if (mOutput->audioHwDev) {
if (mOutput->audioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
}
if (mOutput->audioHwDev->canSetMasterMute()) {
mMasterMute = false;
}
mIsMsdDevice = strcmp(
mOutput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
}
if (mixerConfig != nullptr && mixerConfig->channel_mask != AUDIO_CHANNEL_NONE) {
mMixerChannelMask = mixerConfig->channel_mask;
}
readOutputParameters_l();
if (mType != SPATIALIZER
&& mMixerChannelMask != mChannelMask) {
LOG_ALWAYS_FATAL("HAL channel mask %#x does not match mixer channel mask %#x",
mChannelMask, mMixerChannelMask);
}
// TODO: We may also match on address as well as device type for
// AUDIO_DEVICE_OUT_BUS, AUDIO_DEVICE_OUT_ALL_A2DP, AUDIO_DEVICE_OUT_REMOTE_SUBMIX
if (type == MIXER || type == DIRECT || type == OFFLOAD) {
// TODO: This property should be ensure that only contains one single device type.
mTimestampCorrectedDevice = (audio_devices_t)property_get_int64(
"audio.timestamp.corrected_output_device",
(int64_t)(mIsMsdDevice ? AUDIO_DEVICE_OUT_BUS // turn on by default for MSD
: AUDIO_DEVICE_NONE));
}
for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
mStreamTypes[stream].volume = 0.0f;
mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
}
// Audio patch and call assistant volume are always max
mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
}
PlaybackThread::~PlaybackThread()
{
mAfThreadCallback->unregisterWriter(mNBLogWriter);
free(mSinkBuffer);
free(mMixerBuffer);
free(mEffectBuffer);
free(mPostSpatializerBuffer);
}
// Thread virtuals
void PlaybackThread::onFirstRef()
{
if (!isStreamInitialized()) {
ALOGE("The stream is not open yet"); // This should not happen.
} else {
// Callbacks take strong or weak pointers as a parameter.
// Since PlaybackThread passes itself as a callback handler, it can only
// be done outside of the constructor. Creating weak and especially strong
// pointers to a refcounted object in its own constructor is strongly
// discouraged, see comments in system/core/libutils/include/utils/RefBase.h.
// Even if a function takes a weak pointer, it is possible that it will
// need to convert it to a strong pointer down the line.
if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING &&
mOutput->stream->setCallback(this) == OK) {
mUseAsyncWrite = true;
mCallbackThread = sp<AsyncCallbackThread>::make(this);
}
if (mOutput->stream->setEventCallback(this) != OK) {
ALOGD("Failed to add event callback");
}
}
run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
mThreadSnapshot.setTid(getTid());
}
// ThreadBase virtuals
void PlaybackThread::preExit()
{
ALOGV(" preExit()");
status_t result = mOutput->stream->exit();
ALOGE_IF(result != OK, "Error when calling exit(): %d", result);
}
void PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
result.appendFormat(" Stream volumes in dB: ");
for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
const stream_type_t *st = &mStreamTypes[i];
if (i > 0) {
result.appendFormat(", ");
}
result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
if (st->mute) {
result.append("M");
}
}
result.append("\n");
write(fd, result.c_str(), result.length());
result.clear();
// These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
FastTrackUnderruns underruns = getFastTrackUnderruns(0);
dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
size_t numtracks = mTracks.size();
size_t numactive = mActiveTracks.size();
dprintf(fd, " %zu Tracks", numtracks);
size_t numactiveseen = 0;
const char *prefix = " ";
if (numtracks) {
dprintf(fd, " of which %zu are active\n", numactive);
result.append(prefix);
mTracks[0]->appendDumpHeader(result);
for (size_t i = 0; i < numtracks; ++i) {
sp<IAfTrack> track = mTracks[i];
if (track != 0) {
bool active = mActiveTracks.indexOf(track) >= 0;
if (active) {
numactiveseen++;
}
result.append(prefix);
track->appendDump(result, active);
}
}
} else {
result.append("\n");
}
if (numactiveseen != numactive) {
// some tracks in the active list were not in the tracks list
result.append(" The following tracks are in the active list but"
" not in the track list\n");
result.append(prefix);
mActiveTracks[0]->appendDumpHeader(result);
for (size_t i = 0; i < numactive; ++i) {
sp<IAfTrack> track = mActiveTracks[i];
if (mTracks.indexOf(track) < 0) {
result.append(prefix);
track->appendDump(result, true /* active */);
}
}
}
write(fd, result.c_str(), result.size());
}
void PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
dprintf(fd, " Master volume: %f\n", mMasterVolume);
dprintf(fd, " Master mute: %s\n", mMasterMute ? "on" : "off");
dprintf(fd, " Mixer channel Mask: %#x (%s)\n",
mMixerChannelMask, channelMaskToString(mMixerChannelMask, true /* output */).c_str());
if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
dprintf(fd, " Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
channelMaskToString(mHapticChannelMask, true /* output */).c_str());
}
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Total writes: %d\n", mNumWrites);
dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
dprintf(fd, " Suspend count: %d\n", (int32_t)mSuspended);
dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs);
AudioStreamOut *output = mOutput;
audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n",
output, flags, toString(flags).c_str());
dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten);
dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames);
if (mPipeSink.get() != nullptr) {
dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten());
}
if (output != nullptr) {
dprintf(fd, " Hal stream dump:\n");
(void)output->stream->dump(fd, args);
}
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mutex() held
sp<IAfTrack> PlaybackThread::createTrack_l(
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
pid_t tid,
status_t *status,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback,
bool isSpatialized,
bool isBitPerfect,
audio_output_flags_t *afTrackFlags)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
sp<IAfTrack> track;
status_t lStatus;
audio_output_flags_t outputFlags = mOutput->flags;
audio_output_flags_t requestedFlags = *flags;
uint32_t sampleRate;
if (sharedBuffer != 0 && checkIMemory(sharedBuffer) != NO_ERROR) {
lStatus = BAD_VALUE;
goto Exit;
}
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
sampleRate = *pSampleRate;
// special case for FAST flag considered OK if fast mixer is present
if (hasFastMixer()) {
outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
}
// Check if requested flags are compatible with output stream flags
if ((*flags & outputFlags) != *flags) {
ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)",
*flags, outputFlags);
*flags = (audio_output_flags_t)(*flags & outputFlags);
}
if (isBitPerfect) {
audio_utils::lock_guard _l(mutex());
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
if (chain.get() != nullptr) {
// Bit-perfect is required according to the configuration and preferred mixer
// attributes, but it is not in the output flag from the client's request. Explicitly
// adding bit-perfect flag to check the compatibility
audio_output_flags_t flagsToCheck =
(audio_output_flags_t)(*flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT);
chain->checkOutputFlagCompatibility(&flagsToCheck);
if ((flagsToCheck & AUDIO_OUTPUT_FLAG_BIT_PERFECT) == AUDIO_OUTPUT_FLAG_NONE) {
ALOGE("%s cannot create track as there is data-processing effect attached to "
"given session id(%d)", __func__, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
*flags = flagsToCheck;
}
}
// client expresses a preference for FAST, but we get the final say
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (
// PCM data
audio_is_linear_pcm(format) &&
// TODO: extract as a data library function that checks that a computationally
// expensive downmixer is not required: isFastOutputChannelConversion()
(channelMask == (mChannelMask | mHapticChannelMask) ||
mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
(channelMask == AUDIO_CHANNEL_OUT_MONO
/* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// normal mixer has an associated fast mixer
hasFastMixer() &&
// there are sufficient fast track slots available
(mFastTrackAvailMask != 0)
// FIXME test that MixerThread for this fast track has a capable output HAL
// FIXME add a permission test also?
) {
// static tracks can have any nonzero framecount, streaming tracks check against minimum.
if (sharedBuffer == 0) {
// read the fast track multiplier property the first time it is needed
int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
if (ok != 0) {
ALOGE("%s pthread_once failed: %d", __func__, ok);
}
frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
}
// check compatibility with audio effects.
{ // scope for mutex()
audio_utils::lock_guard _l(mutex());
for (audio_session_t session : {
AUDIO_SESSION_DEVICE,
AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX,
sessionId,
}) {
sp<IAfEffectChain> chain = getEffectChain_l(session);
if (chain.get() != nullptr) {
audio_output_flags_t old = *flags;
chain->checkOutputFlagCompatibility(flags);
if (old != *flags) {
ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x",
(int)session, (int)old, (int)*flags);
}
}
}
}
ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0,
"AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
frameCount, mFrameCount);
} else {
ALOGD("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
"mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
"sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
audio_is_linear_pcm(format), channelMask, sampleRate,
mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (!audio_has_proportional_frames(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mNormalFrameCount;
}
if (notificationFrameCount != frameCount) {
notificationFrameCount = frameCount;
}
} else if (sharedBuffer != 0) {
// FIXME: Ensure client side memory buffers need
// not have additional alignment beyond sample
// (e.g. 16 bit stereo accessed as 32 bit frame).
size_t alignment = audio_bytes_per_sample(format);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
size_t frameSize = channelCount * audio_bytes_per_sample(format);
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)sharedBuffer->unsecurePointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
sharedBuffer->unsecurePointer(), channelCount);
lStatus = BAD_VALUE;
goto Exit;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
frameCount = sharedBuffer->size() / frameSize;
} else {
size_t minFrameCount = 0;
// For fast tracks we try to respect the application's request for notifications per buffer.
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (notificationsPerBuffer > 0) {
// Avoid possible arithmetic overflow during multiplication.
if (notificationsPerBuffer > SIZE_MAX / mFrameCount) {
ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
notificationsPerBuffer, mFrameCount);
} else {
minFrameCount = mFrameCount * notificationsPerBuffer;
}
}
} else {
// For normal PCM streaming tracks, update minimum frame count.
// Buffer depth is forced to be at least 2 x the normal mixer frame count and
// cover audio hardware latency.
// This is probably too conservative, but legacy application code may depend on it.
// If you change this calculation, also review the start threshold which is related.
uint32_t latencyMs = latency_l();
if (latencyMs == 0) {
ALOGE("Error when retrieving output stream latency");
lStatus = UNKNOWN_ERROR;
goto Exit;
}
minFrameCount = AudioSystem::calculateMinFrameCount(latencyMs, mNormalFrameCount,
mSampleRate, sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
}
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
// Make sure that application is notified with sufficient margin before underrun.
// The client can divide the AudioTrack buffer into sub-buffers,
// and expresses its desire to server as the notification frame count.
if (sharedBuffer == 0 && audio_is_linear_pcm(format)) {
size_t maxNotificationFrames;
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
// notify every HAL buffer, regardless of the size of the track buffer
maxNotificationFrames = mFrameCount;
} else {
// Triple buffer the notification period for a triple buffered mixer period;
// otherwise, double buffering for the notification period is fine.
//
// TODO: This should be moved to AudioTrack to modify the notification period
// on AudioTrack::setBufferSizeInFrames() changes.
const int nBuffering =
(uint64_t{frameCount} * mSampleRate)
/ (uint64_t{mNormalFrameCount} * sampleRate) == 3 ? 3 : 2;
maxNotificationFrames = frameCount / nBuffering;
// If client requested a fast track but this was denied, then use the smaller maximum.
if (requestedFlags & AUDIO_OUTPUT_FLAG_FAST) {
size_t maxNotificationFramesFastDenied = FMS_20 * sampleRate / 1000;
if (maxNotificationFrames > maxNotificationFramesFastDenied) {
maxNotificationFrames = maxNotificationFramesFastDenied;
}
}
}
if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
if (notificationFrameCount == 0) {
ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
maxNotificationFrames, frameCount);
} else {
ALOGW("Client adjusted notificationFrames from %zu to %zu for frameCount %zu",
notificationFrameCount, maxNotificationFrames, frameCount);
}
notificationFrameCount = maxNotificationFrames;
}
}
*pFrameCount = frameCount;
*pNotificationFrameCount = notificationFrameCount;
switch (mType) {
case BIT_PERFECT:
if (isBitPerfect) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("%s, bad parameter when request streaming bit-perfect, sampleRate=%u, "
"format=%#x, channelMask=%#x, mSampleRate=%u, mFormat=%#x, mChannelMask=%#x",
__func__, sampleRate, format, channelMask, mSampleRate, mFormat,
mChannelMask);
lStatus = BAD_VALUE;
goto Exit;
}
}
break;
case DIRECT:
if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
}
break;
case OFFLOAD:
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
break;
default:
if (!audio_is_linear_pcm(format)) {
ALOGE("createTrack_l() Bad parameter: format %#x \""
"for output %p with format %#x",
format, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
break;
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() audio driver not initialized");
goto Exit;
}
{ // scope for mutex()
audio_utils::lock_guard _l(mutex());
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
product_strategy_t strategy = getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<IAfTrack> t = mTracks[i];
if (t != 0 && t->isExternalTrack()) {
product_strategy_t actual = getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
}
}
// Set DIRECT flag if current thread is DirectOutputThread. This can
// happen when the playback is rerouted to direct output thread by
// dynamic audio policy.
// Do NOT report the flag changes back to client, since the client
// doesn't explicitly request a direct flag.
audio_output_flags_t trackFlags = *flags;
if (mType == DIRECT) {
trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
}
*afTrackFlags = trackFlags;
track = IAfTrack::create(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, creatorPid, attributionSource, trackFlags,
IAfTrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
speed, isSpatialized, isBitPerfect);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
{
audio_utils::lock_guard _atCbL(audioTrackCbMutex());
if (callback.get() != nullptr) {
mAudioTrackCallbacks.emplace(track, callback);
}
}
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) {
pid_t callingPid = IPCThreadState::self()->getCallingPid();
// we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
// so ask activity manager to do this on our behalf
sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/);
}
}
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}
template<typename T>
ssize_t PlaybackThread::Tracks<T>::remove(const sp<T>& track)
{
const int trackId = track->id();
const ssize_t index = mTracks.remove(track);
if (index >= 0) {
if (mSaveDeletedTrackIds) {
// We can't directly access mAudioMixer since the caller may be outside of threadLoop.
// Instead, we add to mDeletedTrackIds which is solely used for mAudioMixer update,
// to be handled when MixerThread::prepareTracks_l() next changes mAudioMixer.
mDeletedTrackIds.emplace(trackId);
}
}
return index;
}
uint32_t PlaybackThread::correctLatency_l(uint32_t latency) const
{
return latency;
}
uint32_t PlaybackThread::latency() const
{
audio_utils::lock_guard _l(mutex());
return latency_l();
}
uint32_t PlaybackThread::latency_l() const
NO_THREAD_SAFETY_ANALYSIS
// Fix later.
{
uint32_t latency;
if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) {
return correctLatency_l(latency);
}
return 0;
}
void PlaybackThread::setMasterVolume(float value)
{
audio_utils::lock_guard _l(mutex());
// Don't apply master volume in SW if our HAL can do it for us.
if (mOutput && mOutput->audioHwDev &&
mOutput->audioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
} else {
mMasterVolume = value;
}
}
void PlaybackThread::setMasterBalance(float balance)
{
mMasterBalance.store(balance);
}
void PlaybackThread::setMasterMute(bool muted)
{
if (isDuplicating()) {
return;
}
audio_utils::lock_guard _l(mutex());
// Don't apply master mute in SW if our HAL can do it for us.
if (mOutput && mOutput->audioHwDev &&
mOutput->audioHwDev->canSetMasterMute()) {
mMasterMute = false;
} else {
mMasterMute = muted;
}
}
void PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
audio_utils::lock_guard _l(mutex());
mStreamTypes[stream].volume = value;
broadcast_l();
}
void PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
audio_utils::lock_guard _l(mutex());
mStreamTypes[stream].mute = muted;
broadcast_l();
}
float PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
audio_utils::lock_guard _l(mutex());
return mStreamTypes[stream].volume;
}
void PlaybackThread::setVolumeForOutput_l(float left, float right) const
{
mOutput->stream->setVolume(left, right);
}
// addTrack_l() must be called with ThreadBase::mutex() held
status_t PlaybackThread::addTrack_l(const sp<IAfTrack>& track)
{
status_t status = ALREADY_EXISTS;
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
if (track->isExternalTrack()) {
IAfTrackBase::track_state state = track->state();
// Because the track is not on the ActiveTracks,
// at this point, only the TrackHandle will be adding the track.
mutex().unlock();
status = AudioSystem::startOutput(track->portId());
mutex().lock();
// abort track was stopped/paused while we released the lock
if (state != track->state()) {
if (status == NO_ERROR) {
mutex().unlock();
AudioSystem::stopOutput(track->portId());
mutex().lock();
}
return INVALID_OPERATION;
}
// abort if start is rejected by audio policy manager
if (status != NO_ERROR) {
// Do not replace the error if it is DEAD_OBJECT. When this happens, it indicates
// current playback thread is reopened, which may happen when clients set preferred
// mixer configuration. Returning DEAD_OBJECT will make the client restore track
// immediately.
return status == DEAD_OBJECT ? status : PERMISSION_DENIED;
}
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
#endif
sendIoConfigEvent_l(AUDIO_CLIENT_STARTED, track->creatorPid(), track->portId());
}
// set retry count for buffer fill
if (track->isOffloaded()) {
if (track->isStopping_1()) {
track->retryCount() = kMaxTrackStopRetriesOffload;
} else {
track->retryCount() = kMaxTrackStartupRetriesOffload;
}
track->fillingStatus() = mStandby ? IAfTrack::FS_FILLING : IAfTrack::FS_FILLED;
} else {
track->retryCount() = kMaxTrackStartupRetries;
track->fillingStatus() =
track->sharedBuffer() != 0 ? IAfTrack::FS_FILLED : IAfTrack::FS_FILLING;
}
sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
if (mHapticChannelMask != AUDIO_CHANNEL_NONE
&& ((track->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
|| (chain != nullptr && chain->containsHapticGeneratingEffect_l()))) {
// Unlock due to VibratorService will lock for this call and will
// call Tracks.mute/unmute which also require thread's lock.
mutex().unlock();
const os::HapticScale hapticScale = afutils::onExternalVibrationStart(
track->getExternalVibration());
std::optional<media::AudioVibratorInfo> vibratorInfo;
{
// TODO(b/184194780): Use the vibrator information from the vibrator that will be
// used to play this track.
audio_utils::lock_guard _l(mAfThreadCallback->mutex());
vibratorInfo = std::move(mAfThreadCallback->getDefaultVibratorInfo_l());
}
mutex().lock();
track->setHapticScale(hapticScale);
if (vibratorInfo) {
track->setHapticMaxAmplitude(vibratorInfo->maxAmplitude);
}
// Haptic playback should be enabled by vibrator service.
if (track->getHapticPlaybackEnabled()) {
// Disable haptic playback of all active track to ensure only
// one track playing haptic if current track should play haptic.
for (const auto &t : mActiveTracks) {
t->setHapticPlaybackEnabled(false);
}
}
// Set haptic intensity for effect
if (chain != nullptr) {
// TODO(b/324559333): Add adaptive haptics scaling support for the HapticGenerator.
chain->setHapticScale_l(track->id(), hapticScale);
}
}
track->setResetDone(false);
track->resetPresentationComplete();
// Do not release the ThreadBase mutex after the track is added to mActiveTracks unless
// all key changes are complete. It is possible that the threadLoop will begin
// processing the added track immediately after the ThreadBase mutex is released.
mActiveTracks.add(track);
if (chain != 0) {
ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
track->sessionId());
chain->incActiveTrackCnt();
}
track->logBeginInterval(patchSinksToString(&mPatch)); // log to MediaMetrics
status = NO_ERROR;
}
onAddNewTrack_l();
return status;
}
bool PlaybackThread::destroyTrack_l(const sp<IAfTrack>& track)
{
track->terminate();
// active tracks are removed by threadLoop()
bool trackActive = (mActiveTracks.indexOf(track) >= 0);
track->setState(IAfTrackBase::STOPPED);
if (!trackActive) {
removeTrack_l(track);
} else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
if (track->isPausePending()) {
track->pauseAck();
}
track->setState(IAfTrackBase::STOPPING_1);
}
return trackActive;
}
void PlaybackThread::removeTrack_l(const sp<IAfTrack>& track)
{
track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
String8 result;
track->appendDump(result, false /* active */);
mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.c_str());
mTracks.remove(track);
{
audio_utils::lock_guard _atCbL(audioTrackCbMutex());
mAudioTrackCallbacks.erase(track);
}
if (track->isFastTrack()) {
int index = track->fastIndex();
ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks);
ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
mFastTrackAvailMask |= 1 << index;
// redundant as track is about to be destroyed, for dumpsys only
track->fastIndex() = -1;
}
sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->decTrackCnt();
}
}
String8 PlaybackThread::getParameters(const String8& keys)
{
audio_utils::lock_guard _l(mutex());
String8 out_s8;
if (initCheck() == NO_ERROR && mOutput->stream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
return {};
}
status_t DirectOutputThread::selectPresentation(int presentationId, int programId) {
audio_utils::lock_guard _l(mutex());
if (!isStreamInitialized()) {
return NO_INIT;
}
return mOutput->stream->selectPresentation(presentationId, programId);
}
void PlaybackThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
sp<AudioIoDescriptor> desc;
const struct audio_patch patch = isMsdDevice() ? mDownStreamPatch : mPatch;
switch (event) {
case AUDIO_OUTPUT_OPENED:
case AUDIO_OUTPUT_REGISTERED:
case AUDIO_OUTPUT_CONFIG_CHANGED:
desc = sp<AudioIoDescriptor>::make(mId, patch, false /*isInput*/,
mSampleRate, mFormat, mChannelMask,
// FIXME AudioFlinger::frameCount(audio_io_handle_t) instead of mNormalFrameCount?
mNormalFrameCount, mFrameCount, latency_l());
break;
case AUDIO_CLIENT_STARTED:
desc = sp<AudioIoDescriptor>::make(mId, patch, portId);
break;
case AUDIO_OUTPUT_CLOSED:
default:
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
void PlaybackThread::onWriteReady()
{
mCallbackThread->resetWriteBlocked();
}
void PlaybackThread::onDrainReady()
{
mCallbackThread->resetDraining();
}
void PlaybackThread::onError()
{
mCallbackThread->setAsyncError();
}
void PlaybackThread::onCodecFormatChanged(
const std::vector<uint8_t>& metadataBs)
{
const auto weakPointerThis = wp<PlaybackThread>::fromExisting(this);
std::thread([this, metadataBs, weakPointerThis]() {
const sp<PlaybackThread> playbackThread = weakPointerThis.promote();
if (playbackThread == nullptr) {
ALOGW("PlaybackThread was destroyed, skip codec format change event");
return;
}
audio_utils::metadata::Data metadata =
audio_utils::metadata::dataFromByteString(metadataBs);
if (metadata.empty()) {
ALOGW("Can not transform the buffer to audio metadata, %s, %d",
reinterpret_cast<char*>(const_cast<uint8_t*>(metadataBs.data())),
(int)metadataBs.size());
return;
}
audio_utils::metadata::ByteString metaDataStr =
audio_utils::metadata::byteStringFromData(metadata);
std::vector metadataVec(metaDataStr.begin(), metaDataStr.end());
audio_utils::lock_guard _l(audioTrackCbMutex());
for (const auto& callbackPair : mAudioTrackCallbacks) {
callbackPair.second->onCodecFormatChanged(metadataVec);
}
}).detach();
}
void PlaybackThread::resetWriteBlocked(uint32_t sequence)
{
audio_utils::lock_guard _l(mutex());
// reject out of sequence requests
if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
mWriteAckSequence &= ~1;
mWaitWorkCV.notify_one();
}
}
void PlaybackThread::resetDraining(uint32_t sequence)
{
audio_utils::lock_guard _l(mutex());
// reject out of sequence requests
if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
// Register discontinuity when HW drain is completed because that can cause
// the timestamp frame position to reset to 0 for direct and offload threads.
// (Out of sequence requests are ignored, since the discontinuity would be handled
// elsewhere, e.g. in flush).
mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
mDrainSequence &= ~1;
mWaitWorkCV.notify_one();
}
}
void PlaybackThread::readOutputParameters_l()
NO_THREAD_SAFETY_ANALYSIS
// 'moveEffectChain_ll' requires holding mutex 'AudioFlinger_Mutex' exclusively
{
// unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
const audio_config_base_t audioConfig = mOutput->getAudioProperties();
mSampleRate = audioConfig.sample_rate;
mChannelMask = audioConfig.channel_mask;
if (!audio_is_output_channel(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
mChannelMask);
}
if (mMixerChannelMask == AUDIO_CHANNEL_NONE) {
mMixerChannelMask = mChannelMask;
}
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mBalance.setChannelMask(mChannelMask);
uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mMixerChannelMask);
// Get actual HAL format.
status_t result = mOutput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
// Get format from the shim, which will be different than the HAL format
// if playing compressed audio over HDMI passthrough.
mFormat = audioConfig.format;
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
mFrameSize = mOutput->getFrameSize();
result = mOutput->stream->getBufferSize(&mBufferSize);
LOG_ALWAYS_FATAL_IF(result != OK,
"Error when retrieving output stream buffer size: %d", result);
mFrameCount = mBufferSize / mFrameSize;
if (hasMixer() && (mFrameCount & 15)) {
ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
mFrameCount);
}
mHwSupportsPause = false;
if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) {
bool supportsPause = false, supportsResume = false;
if (mOutput->stream->supportsPauseAndResume(&supportsPause, &supportsResume) == OK) {
if (supportsPause && supportsResume) {
mHwSupportsPause = true;
} else if (supportsPause) {
ALOGW("direct output implements pause but not resume");
} else if (supportsResume) {
ALOGW("direct output implements resume but not pause");
}
}
}
if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume");
}
if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
// For best precision, we use float instead of the associated output
// device format (typically PCM 16 bit).
mFormat = AUDIO_FORMAT_PCM_FLOAT;
mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
mBufferSize = mFrameSize * mFrameCount;
// TODO: We currently use the associated output device channel mask and sample rate.
// (1) Perhaps use the ORed channel mask of all downstream MixerThreads
// (if a valid mask) to avoid premature downmix.
// (2) Perhaps use the maximum sample rate of all downstream MixerThreads
// instead of the output device sample rate to avoid loss of high frequency information.
// This may need to be updated as MixerThread/OutputTracks are added and not here.
}
// Calculate size of normal sink buffer relative to the HAL output buffer size
double multiplier = 1.0;
// Note: mType == SPATIALIZER does not support FastMixer.
if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
kUseFastMixer == FastMixer_Dynamic)) {
size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
// round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
maxNormalFrameCount = maxNormalFrameCount & ~15;
if (maxNormalFrameCount < minNormalFrameCount) {
maxNormalFrameCount = minNormalFrameCount;
}
multiplier = (double) minNormalFrameCount / (double) mFrameCount;
if (multiplier <= 1.0) {
multiplier = 1.0;
} else if (multiplier <= 2.0) {
if (2 * mFrameCount <= maxNormalFrameCount) {
multiplier = 2.0;
} else {
multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
}
} else {
multiplier = floor(multiplier);
}
}
mNormalFrameCount = multiplier * mFrameCount;
// round up to nearest 16 frames to satisfy AudioMixer
if (hasMixer()) {
mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
}
ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames",
(size_t)mFrameCount, mNormalFrameCount);
// Check if we want to throttle the processing to no more than 2x normal rate
mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
mThreadThrottleTimeMs = 0;
mThreadThrottleEndMs = 0;
mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate);
// mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
// Originally this was int16_t[] array, need to remove legacy implications.
free(mSinkBuffer);
mSinkBuffer = NULL;
// For sink buffer size, we use the frame size from the downstream sink to avoid problems
// with non PCM formats for compressed music, e.g. AAC, and Offload threads.
const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
(void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
// We resize the mMixerBuffer according to the requirements of the sink buffer which
// drives the output.
free(mMixerBuffer);
mMixerBuffer = NULL;
if (mMixerBufferEnabled) {
mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
mMixerBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mMixerBufferFormat);
(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
}
free(mEffectBuffer);
mEffectBuffer = NULL;
if (mEffectBufferEnabled) {
mEffectBufferFormat = AUDIO_FORMAT_PCM_FLOAT;
mEffectBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mEffectBufferFormat);
(void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
}
if (mType == SPATIALIZER) {
free(mPostSpatializerBuffer);
mPostSpatializerBuffer = nullptr;
mPostSpatializerBufferSize = mNormalFrameCount * mChannelCount
* audio_bytes_per_sample(mEffectBufferFormat);
(void)posix_memalign(&mPostSpatializerBuffer, 32, mPostSpatializerBufferSize);
}
mHapticChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
mChannelCount -= mHapticChannelCount;
mMixerChannelMask = static_cast<audio_channel_mask_t>(mMixerChannelMask & ~mHapticChannelMask);
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
// Note that mutex() is not held when readOutputParameters_l() is called from the constructor
// but in this case nothing is done below as no audio sessions have effect yet so it doesn't
// matter.
// create a copy of mEffectChains as calling moveEffectChain_ll()
// can reorder some effect chains
Vector<sp<IAfEffectChain>> effectChains = mEffectChains;
for (size_t i = 0; i < effectChains.size(); i ++) {
mAfThreadCallback->moveEffectChain_ll(effectChains[i]->sessionId(),
this/* srcThread */, this/* dstThread */);
}
audio_output_flags_t flags = mOutput->flags;
mediametrics::LogItem item(mThreadMetrics.getMetricsId()); // TODO: method in ThreadMetrics?
item.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_READPARAMETERS)
.set(AMEDIAMETRICS_PROP_ENCODING, IAfThreadBase::formatToString(mFormat).c_str())
.set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
.set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
.set(AMEDIAMETRICS_PROP_CHANNELCOUNT, (int32_t)mChannelCount)
.set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mNormalFrameCount)
.set(AMEDIAMETRICS_PROP_FLAGS, toString(flags).c_str())
.set(AMEDIAMETRICS_PROP_PREFIX_HAPTIC AMEDIAMETRICS_PROP_CHANNELMASK,
(int32_t)mHapticChannelMask)
.set(AMEDIAMETRICS_PROP_PREFIX_HAPTIC AMEDIAMETRICS_PROP_CHANNELCOUNT,
(int32_t)mHapticChannelCount)
.set(AMEDIAMETRICS_PROP_PREFIX_HAL AMEDIAMETRICS_PROP_ENCODING,
IAfThreadBase::formatToString(mHALFormat).c_str())
.set(AMEDIAMETRICS_PROP_PREFIX_HAL AMEDIAMETRICS_PROP_FRAMECOUNT,
(int32_t)mFrameCount) // sic - added HAL
;
uint32_t latencyMs;
if (mOutput->stream->getLatency(&latencyMs) == NO_ERROR) {
item.set(AMEDIAMETRICS_PROP_PREFIX_HAL AMEDIAMETRICS_PROP_LATENCYMS, (double)latencyMs);
}
item.record();
}
ThreadBase::MetadataUpdate PlaybackThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
}
StreamOutHalInterface::SourceMetadata metadata;
static const bool stereo_spatialization_property =
property_get_bool("ro.audio.stereo_spatialization_enabled", false);
const bool stereo_spatialization_enabled =
stereo_spatialization_property && com_android_media_audio_stereo_spatialization();
if (stereo_spatialization_enabled) {
std::map<audio_session_t, std::vector<playback_track_metadata_v7_t> >allSessionsMetadata;
for (const sp<IAfTrack>& track : mActiveTracks) {
std::vector<playback_track_metadata_v7_t>& sessionMetadata =
allSessionsMetadata[track->sessionId()];
auto backInserter = std::back_inserter(sessionMetadata);
// No track is invalid as this is called after prepareTrack_l in the same
// critical section
track->copyMetadataTo(backInserter);
}
std::vector<playback_track_metadata_v7_t> spatializedTracksMetaData;
for (const auto& [session, sessionTrackMetadata] : allSessionsMetadata) {
metadata.tracks.insert(metadata.tracks.end(),
sessionTrackMetadata.begin(), sessionTrackMetadata.end());
if (auto chain = getEffectChain_l(session) ; chain != nullptr) {
chain->sendMetadata_l(sessionTrackMetadata, {});
}
if ((hasAudioSession_l(session) & IAfThreadBase::SPATIALIZED_SESSION) != 0) {
spatializedTracksMetaData.insert(spatializedTracksMetaData.end(),
sessionTrackMetadata.begin(), sessionTrackMetadata.end());
}
}
if (auto chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); chain != nullptr) {
chain->sendMetadata_l(metadata.tracks, {});
}
if (auto chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE); chain != nullptr) {
chain->sendMetadata_l(metadata.tracks, spatializedTracksMetaData);
}
if (auto chain = getEffectChain_l(AUDIO_SESSION_DEVICE); chain != nullptr) {
chain->sendMetadata_l(metadata.tracks, {});
}
} else {
auto backInserter = std::back_inserter(metadata.tracks);
for (const sp<IAfTrack>& track : mActiveTracks) {
// No track is invalid as this is called after prepareTrack_l in the same
// critical section
track->copyMetadataTo(backInserter);
}
}
sendMetadataToBackend_l(metadata);
MetadataUpdate change;
change.playbackMetadataUpdate = metadata.tracks;
return change;
}
void PlaybackThread::sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata)
{
mOutput->stream->updateSourceMetadata(metadata);
};
status_t PlaybackThread::getRenderPosition(
uint32_t* halFrames, uint32_t* dspFrames) const
{
if (halFrames == NULL || dspFrames == NULL) {
return BAD_VALUE;
}
audio_utils::lock_guard _l(mutex());
if (initCheck() != NO_ERROR) {
return INVALID_OPERATION;
}
int64_t framesWritten = mBytesWritten / mFrameSize;
*halFrames = framesWritten;
if (isSuspended()) {
// return an estimation of rendered frames when the output is suspended
size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
*dspFrames = (uint32_t)
(framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0);
return NO_ERROR;
} else {
status_t status;
uint32_t frames;
status = mOutput->getRenderPosition(&frames);
*dspFrames = (size_t)frames;
return status;
}
}
product_strategy_t PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) const
{
// session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
return getStrategyForStream(AUDIO_STREAM_MUSIC);
}
for (size_t i = 0; i < mTracks.size(); i++) {
sp<IAfTrack> track = mTracks[i];
if (sessionId == track->sessionId() && !track->isInvalid()) {
return getStrategyForStream(track->streamType());
}
}
return getStrategyForStream(AUDIO_STREAM_MUSIC);
}
AudioStreamOut* PlaybackThread::getOutput() const
{
audio_utils::lock_guard _l(mutex());
return mOutput;
}
AudioStreamOut* PlaybackThread::clearOutput()
{
audio_utils::lock_guard _l(mutex());
AudioStreamOut *output = mOutput;
mOutput = NULL;
// FIXME FastMixer might also have a raw ptr to mOutputSink;
// must push a NULL and wait for ack
mOutputSink.clear();
mPipeSink.clear();
mNormalSink.clear();
return output;
}
// this method must always be called either with ThreadBase mutex() held or inside the thread loop
sp<StreamHalInterface> PlaybackThread::stream() const
{
if (mOutput == NULL) {
return NULL;
}
return mOutput->stream;
}
uint32_t PlaybackThread::activeSleepTimeUs() const
{
return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
}
status_t PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
{
if (!isValidSyncEvent(event)) {
return BAD_VALUE;
}
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<IAfTrack> track = mTracks[i];
if (event->triggerSession() == track->sessionId()) {
(void) track->setSyncEvent(event);
return NO_ERROR;
}
}
return NAME_NOT_FOUND;
}
bool PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
{
return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
}
void PlaybackThread::threadLoop_removeTracks(
[[maybe_unused]] const Vector<sp<IAfTrack>>& tracksToRemove)
{
// Miscellaneous track cleanup when removed from the active list,
// called without Thread lock but synchronized with threadLoop processing.
#ifdef ADD_BATTERY_DATA
for (const auto& track : tracksToRemove) {
if (track->isExternalTrack()) {
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
}
}
#endif
}
void PlaybackThread::checkSilentMode_l()
{
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
if (mOutDeviceTypeAddrs.empty()) {
ALOGD("ro.audio.silent is ignored since no output device is set");
return;
}
if (isSingleDeviceType(outDeviceTypes_l(), AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) {
ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
return;
}
if (property_get("ro.audio.silent", value, "0") > 0) {
char *endptr;
unsigned long ul = strtoul(value, &endptr, 0);
if (*endptr == '\0' && ul != 0) {
ALOGD("Silence is golden");
// The setprop command will not allow a property to be changed after
// the first time it is set, so we don't have to worry about un-muting.
setMasterMute_l(true);
}
}
}
}
// shared by MIXER and DIRECT, overridden by DUPLICATING
ssize_t PlaybackThread::threadLoop_write()
{
LOG_HIST_TS();
mInWrite = true;
ssize_t bytesWritten;
const size_t offset = mCurrentWriteLength - mBytesRemaining;
// If an NBAIO sink is present, use it to write the normal mixer's submix
if (mNormalSink != 0) {
const size_t count = mBytesRemaining / mFrameSize;
ATRACE_BEGIN("write");
// update the setpoint when AudioFlinger::mScreenState changes
const uint32_t screenState = mAfThreadCallback->getScreenState();
if (screenState != mScreenState) {
mScreenState = screenState;
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
if (pipe != NULL) {
pipe->setAvgFrames((mScreenState & 1) ?
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
ATRACE_END();
if (framesWritten > 0) {
bytesWritten = framesWritten * mFrameSize;
#ifdef TEE_SINK
mTee.write((char *)mSinkBuffer + offset, framesWritten);
#endif
} else {
bytesWritten = framesWritten;
}
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
mWriteAckSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
ATRACE_BEGIN("write");
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
ATRACE_END();
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
mWriteAckSequence &= ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
}
mNumWrites++;
mInWrite = false;
if (mStandby) {
mThreadMetrics.logBeginInterval();
mThreadSnapshot.onBegin();
mStandby = false;
}
return bytesWritten;
}
// startMelComputation_l() must be called with AudioFlinger::mutex() held
void PlaybackThread::startMelComputation_l(
const sp<audio_utils::MelProcessor>& processor)
{
auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
if (outputSink != nullptr) {
outputSink->startMelComputation(processor);
}
}
// stopMelComputation_l() must be called with AudioFlinger::mutex() held
void PlaybackThread::stopMelComputation_l()
{
auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
if (outputSink != nullptr) {
outputSink->stopMelComputation();
}
}
void PlaybackThread::threadLoop_drain()
{
bool supportsDrain = false;
if (mOutput->stream->supportsDrain(&supportsDrain) == OK && supportsDrain) {
ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
if (mUseAsyncWrite) {
ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
mDrainSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setDraining(mDrainSequence);
}
status_t result = mOutput->stream->drain(mMixerStatus == MIXER_DRAIN_TRACK);
ALOGE_IF(result != OK, "Error when draining stream: %d", result);
}
}
void PlaybackThread::threadLoop_exit()
{
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); i++) {
sp<IAfTrack> track = mTracks[i];
track->invalidate();
}
// Clear ActiveTracks to update BatteryNotifier in case active tracks remain.
// After we exit there are no more track changes sent to BatteryNotifier
// because that requires an active threadLoop.
// TODO: should we decActiveTrackCnt() of the cleared track effect chain?
mActiveTracks.clear();
}
}
/*
The derived values that are cached:
- mSinkBufferSize from frame count * frame size
- mActiveSleepTimeUs from activeSleepTimeUs()
- mIdleSleepTimeUs from idleSleepTimeUs()
- mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least
kDefaultStandbyTimeInNsecs when connected to an A2DP device.
- maxPeriod from frame count and sample rate (MIXER only)
The parameters that affect these derived values are:
- frame count
- frame size
- sample rate
- device type: A2DP or not
- device latency
- format: PCM or not
- active sleep time
- idle sleep time
*/
void PlaybackThread::cacheParameters_l()
{
mSinkBufferSize = mNormalFrameCount * mFrameSize;
mActiveSleepTimeUs = activeSleepTimeUs();
mIdleSleepTimeUs = idleSleepTimeUs();
mStandbyDelayNs = getStandbyTimeInNanos();
// make sure standby delay is not too short when connected to an A2DP sink to avoid
// truncating audio when going to standby.
if (!Intersection(outDeviceTypes_l(), getAudioDeviceOutAllA2dpSet()).empty()) {
if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
}
}
}
bool PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
{
ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
this, streamType, mTracks.size());
bool trackMatch = false;
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<IAfTrack> t = mTracks[i];
if (t->streamType() == streamType && t->isExternalTrack()) {
t->invalidate();
trackMatch = true;
}
}
return trackMatch;
}
void PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
audio_utils::lock_guard _l(mutex());
invalidateTracks_l(streamType);
}
void PlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
audio_utils::lock_guard _l(mutex());
invalidateTracks_l(portIds);
}
bool PlaybackThread::invalidateTracks_l(std::set<audio_port_handle_t>& portIds) {
bool trackMatch = false;
const size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<IAfTrack> t = mTracks[i];
if (t->isExternalTrack() && portIds.find(t->portId()) != portIds.end()) {
t->invalidate();
portIds.erase(t->portId());
trackMatch = true;
}
if (portIds.empty()) {
break;
}
}
return trackMatch;
}
// getTrackById_l must be called with holding thread lock
IAfTrack* PlaybackThread::getTrackById_l(
audio_port_handle_t trackPortId) {
for (size_t i = 0; i < mTracks.size(); i++) {
if (mTracks[i]->portId() == trackPortId) {
return mTracks[i].get();
}
}
return nullptr;
}
status_t PlaybackThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
sp<EffectBufferHalInterface> halInBuffer, halOutBuffer;
float *buffer = nullptr; // only used for non global sessions
if (mType == SPATIALIZER) {
if (!audio_is_global_session(session)) {
// player sessions on a spatializer output will use a dedicated input buffer and
// will either output multi channel to mEffectBuffer if the track is spatilaized
// or stereo to mPostSpatializerBuffer if not spatialized.
uint32_t channelMask;
bool isSessionSpatialized =
(hasAudioSession_l(session) & ThreadBase::SPATIALIZED_SESSION) != 0;
if (isSessionSpatialized) {
channelMask = mMixerChannelMask;
} else {
channelMask = mChannelMask;
}
size_t numSamples = mNormalFrameCount
* (audio_channel_count_from_out_mask(channelMask) + mHapticChannelCount);
status_t result = mAfThreadCallback->getEffectsFactoryHal()->allocateBuffer(
numSamples * sizeof(float),
&halInBuffer);
if (result != OK) return result;
result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
isSessionSpatialized ? mEffectBuffer : mPostSpatializerBuffer,
isSessionSpatialized ? mEffectBufferSize : mPostSpatializerBufferSize,
&halOutBuffer);
if (result != OK) return result;
buffer = halInBuffer ? halInBuffer->audioBuffer()->f32 : buffer;
ALOGV("addEffectChain_l() creating new input buffer %p session %d",
buffer, session);
} else {
// A global session on a SPATIALIZER thread is either OUTPUT_STAGE or DEVICE
// - OUTPUT_STAGE session uses the mEffectBuffer as input buffer and
// mPostSpatializerBuffer as output buffer
// - DEVICE session uses the mPostSpatializerBuffer as input and output buffer.
status_t result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
mEffectBuffer, mEffectBufferSize, &halInBuffer);
if (result != OK) return result;
result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
mPostSpatializerBuffer, mPostSpatializerBufferSize, &halOutBuffer);
if (result != OK) return result;
if (session == AUDIO_SESSION_DEVICE) {
halInBuffer = halOutBuffer;
}
}
} else {
status_t result = mAfThreadCallback->getEffectsFactoryHal()->mirrorBuffer(
mEffectBufferEnabled ? mEffectBuffer : mSinkBuffer,
mEffectBufferEnabled ? mEffectBufferSize : mSinkBufferSize,
&halInBuffer);
if (result != OK) return result;
halOutBuffer = halInBuffer;
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
if (!audio_is_global_session(session)) {
buffer = halInBuffer ? reinterpret_cast<float*>(halInBuffer->externalData())
: buffer;
// Only one effect chain can be present in direct output thread and it uses
// the sink buffer as input
if (mType != DIRECT) {
size_t numSamples = mNormalFrameCount
* (audio_channel_count_from_out_mask(mMixerChannelMask)
+ mHapticChannelCount);
const status_t allocateStatus =
mAfThreadCallback->getEffectsFactoryHal()->allocateBuffer(
numSamples * sizeof(float),
&halInBuffer);
if (allocateStatus != OK) return allocateStatus;
buffer = halInBuffer ? halInBuffer->audioBuffer()->f32 : buffer;
ALOGV("addEffectChain_l() creating new input buffer %p session %d",
buffer, session);
}
}
}
if (!audio_is_global_session(session)) {
// Attach all tracks with same session ID to this chain.
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<IAfTrack> track = mTracks[i];
if (session == track->sessionId()) {
ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p",
track.get(), buffer);
track->setMainBuffer(buffer);
chain->incTrackCnt();
}
}
// indicate all active tracks in the chain
for (const sp<IAfTrack>& track : mActiveTracks) {
if (session == track->sessionId()) {
ALOGV("addEffectChain_l() activating track %p on session %d",
track.get(), session);
chain->incActiveTrackCnt();
}
}
}
chain->setThread(this);
chain->setInBuffer(halInBuffer);
chain->setOutBuffer(halOutBuffer);
// Effect chain for session AUDIO_SESSION_DEVICE is inserted at end of effect
// chains list in order to be processed last as it contains output device effects.
// Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted just before to apply post
// processing effects specific to an output stream before effects applied to all streams
// routed to a given device.
// Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
// session AUDIO_SESSION_OUTPUT_STAGE to be processed
// after track specific effects and before output stage.
// It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
// that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX.
// Effect chain for other sessions are inserted at beginning of effect
// chains list to be processed before output mix effects. Relative order between other
// sessions is not important.
static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX &&
AUDIO_SESSION_DEVICE < AUDIO_SESSION_OUTPUT_STAGE,
"audio_session_t constants misdefined");
size_t size = mEffectChains.size();
size_t i = 0;
for (i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() < session) {
break;
}
}
mEffectChains.insertAt(chain, i);
checkSuspendOnAddEffectChain_l(chain);
return NO_ERROR;
}
size_t PlaybackThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (chain == mEffectChains[i]) {
mEffectChains.removeAt(i);
// detach all active tracks from the chain
for (const sp<IAfTrack>& track : mActiveTracks) {
if (session == track->sessionId()) {
ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
chain.get(), session);
chain->decActiveTrackCnt();
}
}
// detach all tracks with same session ID from this chain
for (size_t j = 0; j < mTracks.size(); ++j) {
sp<IAfTrack> track = mTracks[j];
if (session == track->sessionId()) {
track->setMainBuffer(reinterpret_cast<float*>(mSinkBuffer));
chain->decTrackCnt();
}
}
break;
}
}
return mEffectChains.size();
}
status_t PlaybackThread::attachAuxEffect(
const sp<IAfTrack>& track, int EffectId)
{
audio_utils::lock_guard _l(mutex());
return attachAuxEffect_l(track, EffectId);
}
status_t PlaybackThread::attachAuxEffect_l(
const sp<IAfTrack>& track, int EffectId)
{
status_t status = NO_ERROR;
if (EffectId == 0) {
track->setAuxBuffer(0, NULL);
} else {
// Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
sp<IAfEffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
if (effect != 0) {
if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
} else {
status = INVALID_OPERATION;
}
} else {
status = BAD_VALUE;
}
}
return status;
}
void PlaybackThread::detachAuxEffect_l(int effectId)
{
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<IAfTrack> track = mTracks[i];
if (track->auxEffectId() == effectId) {
attachAuxEffect_l(track, 0);
}
}
}
bool PlaybackThread::threadLoop()
NO_THREAD_SAFETY_ANALYSIS // manual locking of AudioFlinger
{
aflog::setThreadWriter(mNBLogWriter.get());
if (mType == SPATIALIZER) {
const pid_t tid = getTid();
if (tid == -1) { // odd: we are here, we must be a running thread.
ALOGW("%s: Cannot update Spatializer mixer thread priority, no tid", __func__);
} else {
const int priorityBoost = requestSpatializerPriority(getpid(), tid);
if (priorityBoost > 0) {
stream()->setHalThreadPriority(priorityBoost);
}
}
} else if (property_get_bool("ro.boot.container", false /* default_value */)) {
// In ARC experiments (b/73091832), the latency under using CFS scheduler with any priority
// is not enough for PlaybackThread to process audio data in time. We request the lowest
// real-time priority, SCHED_FIFO=1, for PlaybackThread in ARC. ro.boot.container is true
// only on ARC.
const pid_t tid = getTid();
if (tid == -1) {
ALOGW("%s: Cannot update PlaybackThread priority for ARC, no tid", __func__);
} else {
const status_t status = requestPriority(getpid(),
tid,
kPriorityPlaybackThreadArc,
false /* isForApp */,
true /* asynchronous */);
if (status != OK) {
ALOGW("%s: Cannot update PlaybackThread priority for ARC, status %d", __func__,
status);
} else {
stream()->setHalThreadPriority(kPriorityPlaybackThreadArc);
}
}
}
Vector<sp<IAfTrack>> tracksToRemove;
mStandbyTimeNs = systemTime();
int64_t lastLoopCountWritten = -2; // never matches "previous" loop, when loopCount = 0.
// MIXER
nsecs_t lastWarning = 0;
// DUPLICATING
// FIXME could this be made local to while loop?
writeFrames = 0;
cacheParameters_l();
mSleepTimeUs = mIdleSleepTimeUs;
if (mType == MIXER || mType == SPATIALIZER) {
sleepTimeShift = 0;
}
CpuStats cpuStats;
const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
acquireWakeLock();
// mNBLogWriter logging APIs can only be called by a single thread, typically the
// thread associated with this PlaybackThread.
// If you want to share the mNBLogWriter with other threads (for example, binder threads)
// then all such threads must agree to hold a common mutex before logging.
// So if you need to log when mutex is unlocked, set logString to a non-NULL string,
// and then that string will be logged at the next convenient opportunity.
// See reference to logString below.
const char *logString = NULL;
// Estimated time for next buffer to be written to hal. This is used only on
// suspended mode (for now) to help schedule the wait time until next iteration.
nsecs_t timeLoopNextNs = 0;
checkSilentMode_l();
audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
sendCheckOutputStageEffectsEvent();
// loopCount is used for statistics and diagnostics.
for (int64_t loopCount = 0; !exitPending(); ++loopCount)
{
// Log merge requests are performed during AudioFlinger binder transactions, but
// that does not cover audio playback. It's requested here for that reason.
mAfThreadCallback->requestLogMerge();
cpuStats.sample(myName);
Vector<sp<IAfEffectChain>> effectChains;
audio_session_t activeHapticSessionId = AUDIO_SESSION_NONE;
bool isHapticSessionSpatialized = false;
std::vector<sp<IAfTrack>> activeTracks;
// If the device is AUDIO_DEVICE_OUT_BUS, check for downstream latency.
//
// Note: we access outDeviceTypes() outside of mutex().
if (isMsdDevice() && outDeviceTypes_l().count(AUDIO_DEVICE_OUT_BUS) != 0) {
// Here, we try for the AF lock, but do not block on it as the latency
// is more informational.
if (mAfThreadCallback->mutex().try_lock()) {
std::vector<SoftwarePatch> swPatches;
double latencyMs = 0.; // not required; initialized for clang-tidy
status_t status = INVALID_OPERATION;
audio_patch_handle_t downstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (mAfThreadCallback->getPatchPanel()->getDownstreamSoftwarePatches(
id(), &swPatches) == OK
&& swPatches.size() > 0) {
status = swPatches[0].getLatencyMs_l(&latencyMs);
downstreamPatchHandle = swPatches[0].getPatchHandle();
}
if (downstreamPatchHandle != lastDownstreamPatchHandle) {
mDownstreamLatencyStatMs.reset();
lastDownstreamPatchHandle = downstreamPatchHandle;
}
if (status == OK) {
// verify downstream latency (we assume a max reasonable
// latency of 5 seconds).
const double minLatency = 0., maxLatency = 5000.;
if (latencyMs >= minLatency && latencyMs <= maxLatency) {
ALOGVV("new downstream latency %lf ms", latencyMs);
} else {
ALOGD("out of range downstream latency %lf ms", latencyMs);
latencyMs = std::clamp(latencyMs, minLatency, maxLatency);
}
mDownstreamLatencyStatMs.add(latencyMs);
}
mAfThreadCallback->mutex().unlock();
}
} else {
if (lastDownstreamPatchHandle != AUDIO_PATCH_HANDLE_NONE) {
// our device is no longer AUDIO_DEVICE_OUT_BUS, reset patch handle and stats.
mDownstreamLatencyStatMs.reset();
lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
}
}
if (mCheckOutputStageEffects.exchange(false)) {
checkOutputStageEffects();
}
MetadataUpdate metadataUpdate;
{ // scope for mutex()
audio_utils::unique_lock _l(mutex());
processConfigEvents_l();
if (mCheckOutputStageEffects.load()) {
continue;
}
// See comment at declaration of logString for why this is done under mutex()
if (logString != NULL) {
mNBLogWriter->logTimestamp();
mNBLogWriter->log(logString);
logString = NULL;
}
collectTimestamps_l();
saveOutputTracks();
if (mSignalPending) {
// A signal was raised while we were unlocked
mSignalPending = false;
} else if (waitingAsyncCallback_l()) {
if (exitPending()) {
break;
}
bool released = false;
if (!keepWakeLock()) {
releaseWakeLock_l();
released = true;
}
const int64_t waitNs = computeWaitTimeNs_l();
ALOGV("wait async completion (wait time: %lld)", (long long)waitNs);
std::cv_status cvstatus =
mWaitWorkCV.wait_for(_l, std::chrono::nanoseconds(waitNs));
if (cvstatus == std::cv_status::timeout) {
mSignalPending = true; // if timeout recheck everything
}
ALOGV("async completion/wake");
if (released) {
acquireWakeLock_l();
}
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mSleepTimeUs = 0;
continue;
}
if ((mActiveTracks.isEmpty() && systemTime() > mStandbyTimeNs) ||
isSuspended()) {
// put audio hardware into standby after short delay
if (shouldStandby_l()) {
threadLoop_standby();
// This is where we go into standby
if (!mStandby) {
LOG_AUDIO_STATE();
mThreadMetrics.logEndInterval();
mThreadSnapshot.onEnd();
setStandby_l();
}
sendStatistics(false /* force */);
}
if (mActiveTracks.isEmpty() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
clearOutputTracks();
if (exitPending()) {
break;
}
releaseWakeLock_l();
// wait until we have something to do...
ALOGV("%s going to sleep", myName.c_str());
mWaitWorkCV.wait(_l);
ALOGV("%s waking up", myName.c_str());
acquireWakeLock_l();
mMixerStatus = MIXER_IDLE;
mMixerStatusIgnoringFastTracks = MIXER_IDLE;
mBytesWritten = 0;
mBytesRemaining = 0;
checkSilentMode_l();
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mSleepTimeUs = mIdleSleepTimeUs;
if (mType == MIXER || mType == SPATIALIZER) {
sleepTimeShift = 0;
}
continue;
}
}
// mMixerStatusIgnoringFastTracks is also updated internally
mMixerStatus = prepareTracks_l(&tracksToRemove);
mActiveTracks.updatePowerState_l(this);
metadataUpdate = updateMetadata_l();
// Acquire a local copy of active tracks with lock (release w/o lock).
//
// Control methods on the track acquire the ThreadBase lock (e.g. start()
// stop(), pause(), etc.), but the threadLoop is entitled to call audio
// data / buffer methods on tracks from activeTracks without the ThreadBase lock.
activeTracks.insert(activeTracks.end(), mActiveTracks.begin(), mActiveTracks.end());
setHalLatencyMode_l();
// updateTeePatches_l will acquire the ThreadBase_Mutex of other threads,
// so this is done before we lock our effect chains.
for (const auto& track : mActiveTracks) {
track->updateTeePatches_l();
}
// signal actual start of output stream when the render position reported by
// the kernel starts moving.
if (!mHalStarted && ((isSuspended() && (mBytesWritten != 0)) || (!mStandby
&& (mKernelPositionOnStandby
!= mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL])))) {
mHalStarted = true;
mWaitHalStartCV.notify_all();
}
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
// Determine which session to pick up haptic data.
// This must be done under the same lock as prepareTracks_l().
// The haptic data from the effect is at a higher priority than the one from track.
// TODO: Write haptic data directly to sink buffer when mixing.
if (mHapticChannelCount > 0) {
for (const auto& track : mActiveTracks) {
sp<IAfEffectChain> effectChain = getEffectChain_l(track->sessionId());
if (effectChain != nullptr
&& effectChain->containsHapticGeneratingEffect_l()) {
activeHapticSessionId = track->sessionId();
isHapticSessionSpatialized =
mType == SPATIALIZER && track->isSpatialized();
break;
}
if (activeHapticSessionId == AUDIO_SESSION_NONE
&& track->getHapticPlaybackEnabled()) {
activeHapticSessionId = track->sessionId();
isHapticSessionSpatialized =
mType == SPATIALIZER && track->isSpatialized();
}
}
}
} // mutex() scope ends
if (mBytesRemaining == 0) {
mCurrentWriteLength = 0;
if (mMixerStatus == MIXER_TRACKS_READY) {
// threadLoop_mix() sets mCurrentWriteLength
threadLoop_mix();
} else if ((mMixerStatus != MIXER_DRAIN_TRACK)
&& (mMixerStatus != MIXER_DRAIN_ALL)) {
// threadLoop_sleepTime sets mSleepTimeUs to 0 if data
// must be written to HAL
threadLoop_sleepTime();
if (mSleepTimeUs == 0) {
mCurrentWriteLength = mSinkBufferSize;
// Tally underrun frames as we are inserting 0s here.
for (const auto& track : activeTracks) {
if (track->fillingStatus() == IAfTrack::FS_ACTIVE
&& !track->isStopped()
&& !track->isPaused()
&& !track->isTerminated()) {
ALOGV("%s: track(%d) %s underrun due to thread sleep of %zu frames",
__func__, track->id(), track->getTrackStateAsString(),
mNormalFrameCount);
track->audioTrackServerProxy()->tallyUnderrunFrames(
mNormalFrameCount);
}
}
}
}
// Either threadLoop_mix() or threadLoop_sleepTime() should have set
// mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
// Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
// or mSinkBuffer (if there are no effects and there is no data already copied to
// mSinkBuffer).
//
// This is done pre-effects computation; if effects change to
// support higher precision, this needs to move.
//
// mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
// TODO use mSleepTimeUs == 0 as an additional condition.
uint32_t mixerChannelCount = mEffectBufferValid ?
audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
if (mMixerBufferValid && (mEffectBufferValid || !mHasDataCopiedToSinkBuffer)) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
// Apply mono blending and balancing if the effect buffer is not valid. Otherwise,
// do these processes after effects are applied.
if (!mEffectBufferValid) {
// mono blend occurs for mixer threads only (not direct or offloaded)
// and is handled here if we're going directly to the sink.
if (requireMonoBlend()) {
mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount,
mNormalFrameCount, true /*limit*/);
}
if (!hasFastMixer()) {
// Balance must take effect after mono conversion.
// We do it here if there is no FastMixer.
// mBalance detects zero balance within the class for speed
// (not needed here).
mBalance.setBalance(mMasterBalance.load());
mBalance.process((float *)mMixerBuffer, mNormalFrameCount);
}
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
mNormalFrameCount * (mixerChannelCount + mHapticChannelCount));
// If we're going directly to the sink and there are haptic channels,
// we should adjust channels as the sample data is partially interleaved
// in this case.
if (!mEffectBufferValid && mHapticChannelCount > 0) {
adjust_channels_non_destructive(buffer, mChannelCount, buffer,
mChannelCount + mHapticChannelCount,
audio_bytes_per_sample(format),
audio_bytes_per_frame(mChannelCount, format) * mNormalFrameCount);
}
}
mBytesRemaining = mCurrentWriteLength;
if (isSuspended()) {
// Simulate write to HAL when suspended (e.g. BT SCO phone call).
mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer.
const size_t framesRemaining = mBytesRemaining / mFrameSize;
mBytesWritten += mBytesRemaining;
mFramesWritten += framesRemaining;
mSuspendedFrames += framesRemaining; // to adjust kernel HAL position
mBytesRemaining = 0;
}
// only process effects if we're going to write
if (mSleepTimeUs == 0 && mType != OFFLOAD) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
// TODO: Write haptic data directly to sink buffer when mixing.
if (activeHapticSessionId != AUDIO_SESSION_NONE
&& activeHapticSessionId == effectChains[i]->sessionId()) {
// Haptic data is active in this case, copy it directly from
// in buffer to out buffer.
uint32_t hapticSessionChannelCount = mEffectBufferValid ?
audio_channel_count_from_out_mask(mMixerChannelMask) :
mChannelCount;
if (mType == SPATIALIZER && !isHapticSessionSpatialized) {
hapticSessionChannelCount = mChannelCount;
}
const size_t audioBufferSize = mNormalFrameCount
* audio_bytes_per_frame(hapticSessionChannelCount,
AUDIO_FORMAT_PCM_FLOAT);
memcpy_by_audio_format(
(uint8_t*)effectChains[i]->outBuffer() + audioBufferSize,
AUDIO_FORMAT_PCM_FLOAT,
(const uint8_t*)effectChains[i]->inBuffer() + audioBufferSize,
AUDIO_FORMAT_PCM_FLOAT, mNormalFrameCount * mHapticChannelCount);
}
}
}
}
// Process effect chains for offloaded thread even if no audio
// was read from audio track: process only updates effect state
// and thus does have to be synchronized with audio writes but may have
// to be called while waiting for async write callback
if (mType == OFFLOAD) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
}
// Only if the Effects buffer is enabled and there is data in the
// Effects buffer (buffer valid), we need to
// copy into the sink buffer.
// TODO use mSleepTimeUs == 0 as an additional condition.
if (mEffectBufferValid && !mHasDataCopiedToSinkBuffer) {
//ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
void *effectBuffer = (mType == SPATIALIZER) ? mPostSpatializerBuffer : mEffectBuffer;
if (requireMonoBlend()) {
mono_blend(effectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
true /*limit*/);
}
if (!hasFastMixer()) {
// Balance must take effect after mono conversion.
// We do it here if there is no FastMixer.
// mBalance detects zero balance within the class for speed (not needed here).
mBalance.setBalance(mMasterBalance.load());
mBalance.process((float *)effectBuffer, mNormalFrameCount);
}
// for SPATIALIZER thread, Move haptics channels from mEffectBuffer to
// mPostSpatializerBuffer if the haptics track is spatialized.
// Otherwise, the haptics channels are already in mPostSpatializerBuffer.
// For other thread types, the haptics channels are already in mEffectBuffer.
if (mType == SPATIALIZER && isHapticSessionSpatialized) {
const size_t srcBufferSize = mNormalFrameCount *
audio_bytes_per_frame(audio_channel_count_from_out_mask(mMixerChannelMask),
mEffectBufferFormat);
const size_t dstBufferSize = mNormalFrameCount
* audio_bytes_per_frame(mChannelCount, mEffectBufferFormat);
memcpy_by_audio_format((uint8_t*)mPostSpatializerBuffer + dstBufferSize,
mEffectBufferFormat,
(uint8_t*)mEffectBuffer + srcBufferSize,
mEffectBufferFormat,
mNormalFrameCount * mHapticChannelCount);
}
const size_t framesToCopy = mNormalFrameCount * (mChannelCount + mHapticChannelCount);
if (mFormat == AUDIO_FORMAT_PCM_FLOAT &&
mEffectBufferFormat == AUDIO_FORMAT_PCM_FLOAT) {
// Clamp PCM float values more than this distance from 0 to insulate
// a HAL which doesn't handle NaN correctly.
static constexpr float HAL_FLOAT_SAMPLE_LIMIT = 2.0f;
memcpy_to_float_from_float_with_clamping(static_cast<float*>(mSinkBuffer),
static_cast<const float*>(effectBuffer),
framesToCopy, HAL_FLOAT_SAMPLE_LIMIT /* absMax */);
} else {
memcpy_by_audio_format(mSinkBuffer, mFormat,
effectBuffer, mEffectBufferFormat, framesToCopy);
}
// The sample data is partially interleaved when haptic channels exist,
// we need to adjust channels here.
if (mHapticChannelCount > 0) {
adjust_channels_non_destructive(mSinkBuffer, mChannelCount, mSinkBuffer,
mChannelCount + mHapticChannelCount,
audio_bytes_per_sample(mFormat),
audio_bytes_per_frame(mChannelCount, mFormat) * mNormalFrameCount);
}
}
// enable changes in effect chain
unlockEffectChains(effectChains);
if (!metadataUpdate.playbackMetadataUpdate.empty()) {
mAfThreadCallback->getMelReporter()->updateMetadataForCsd(id(),
metadataUpdate.playbackMetadataUpdate);
}
if (!waitingAsyncCallback()) {
// mSleepTimeUs == 0 means we must write to audio hardware
if (mSleepTimeUs == 0) {
ssize_t ret = 0;
// writePeriodNs is updated >= 0 when ret > 0.
int64_t writePeriodNs = -1;
if (mBytesRemaining) {
// FIXME rewrite to reduce number of system calls
const int64_t lastIoBeginNs = systemTime();
ret = threadLoop_write();
const int64_t lastIoEndNs = systemTime();
if (ret < 0) {
mBytesRemaining = 0;
} else if (ret > 0) {
mBytesWritten += ret;
mBytesRemaining -= ret;
const int64_t frames = ret / mFrameSize;
mFramesWritten += frames;
writePeriodNs = lastIoEndNs - mLastIoEndNs;
// process information relating to write time.
if (audio_has_proportional_frames(mFormat)) {
// we are in a continuous mixing cycle
if (mMixerStatus == MIXER_TRACKS_READY &&
loopCount == lastLoopCountWritten + 1) {
const double jitterMs =
TimestampVerifier<int64_t, int64_t>::computeJitterMs(
{frames, writePeriodNs},
{0, 0} /* lastTimestamp */, mSampleRate);
const double processMs =
(lastIoBeginNs - mLastIoEndNs) * 1e-6;
audio_utils::lock_guard _l(mutex());
mIoJitterMs.add(jitterMs);
mProcessTimeMs.add(processMs);
if (mPipeSink.get() != nullptr) {
// Using the Monopipe availableToWrite, we estimate the current
// buffer size.
MonoPipe* monoPipe = static_cast<MonoPipe*>(mPipeSink.get());
const ssize_t
availableToWrite = mPipeSink->availableToWrite();
const size_t pipeFrames = monoPipe->maxFrames();
const size_t
remainingFrames = pipeFrames - max(availableToWrite, 0);
mMonopipePipeDepthStats.add(remainingFrames);
}
}
// write blocked detection
const int64_t deltaWriteNs = lastIoEndNs - lastIoBeginNs;
if ((mType == MIXER || mType == SPATIALIZER)
&& deltaWriteNs > maxPeriod) {
mNumDelayedWrites++;
if ((lastIoEndNs - lastWarning) > kWarningThrottleNs) {
ATRACE_NAME("underrun");
ALOGW("write blocked for %lld msecs, "
"%d delayed writes, thread %d",
(long long)deltaWriteNs / NANOS_PER_MILLISECOND,
mNumDelayedWrites, mId);
lastWarning = lastIoEndNs;
}
}
}
// update timing info.
mLastIoBeginNs = lastIoBeginNs;
mLastIoEndNs = lastIoEndNs;
lastLoopCountWritten = loopCount;
}
} else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
(mMixerStatus == MIXER_DRAIN_ALL)) {
threadLoop_drain();
}
if ((mType == MIXER || mType == SPATIALIZER) && !mStandby) {
if (mThreadThrottle
&& mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
&& writePeriodNs > 0) { // we have write period info
// Limit MixerThread data processing to no more than twice the
// expected processing rate.
//
// This helps prevent underruns with NuPlayer and other applications
// which may set up buffers that are close to the minimum size, or use
// deep buffers, and rely on a double-buffering sleep strategy to fill.
//
// The throttle smooths out sudden large data drains from the device,
// e.g. when it comes out of standby, which often causes problems with
// (1) mixer threads without a fast mixer (which has its own warm-up)
// (2) minimum buffer sized tracks (even if the track is full,
// the app won't fill fast enough to handle the sudden draw).
//
// Total time spent in last processing cycle equals time spent in
// 1. threadLoop_write, as well as time spent in
// 2. threadLoop_mix (significant for heavy mixing, especially
// on low tier processors)
// it's OK if deltaMs is an overestimate.
const int32_t deltaMs = writePeriodNs / NANOS_PER_MILLISECOND;
const int32_t throttleMs = (int32_t)mHalfBufferMs - deltaMs;
if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
mThreadMetrics.logThrottleMs((double)throttleMs);
usleep(throttleMs * 1000);
// notify of throttle start on verbose log
ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
"mixer(%p) throttle begin:"
" ret(%zd) deltaMs(%d) requires sleep %d ms",
this, ret, deltaMs, throttleMs);
mThreadThrottleTimeMs += throttleMs;
// Throttle must be attributed to the previous mixer loop's write time
// to allow back-to-back throttling.
// This also ensures proper timing statistics.
mLastIoEndNs = systemTime(); // we fetch the write end time again.
} else {
uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
if (diff > 0) {
// notify of throttle end on debug log
// but prevent spamming for bluetooth
ALOGD_IF(!isSingleDeviceType(
outDeviceTypes_l(), audio_is_a2dp_out_device) &&
!isSingleDeviceType(
outDeviceTypes_l(),
audio_is_hearing_aid_out_device),
"mixer(%p) throttle end: throttle time(%u)", this, diff);
mThreadThrottleEndMs = mThreadThrottleTimeMs;
}
}
}
}
} else {
ATRACE_BEGIN("sleep");
audio_utils::unique_lock _l(mutex());
// suspended requires accurate metering of sleep time.
if (isSuspended()) {
// advance by expected sleepTime
timeLoopNextNs += microseconds((nsecs_t)mSleepTimeUs);
const nsecs_t nowNs = systemTime();
// compute expected next time vs current time.
// (negative deltas are treated as delays).
nsecs_t deltaNs = timeLoopNextNs - nowNs;
if (deltaNs < -kMaxNextBufferDelayNs) {
// Delays longer than the max allowed trigger a reset.
ALOGV("DelayNs: %lld, resetting timeLoopNextNs", (long long) deltaNs);
deltaNs = microseconds((nsecs_t)mSleepTimeUs);
timeLoopNextNs = nowNs + deltaNs;
} else if (deltaNs < 0) {
// Delays within the max delay allowed: zero the delta/sleepTime
// to help the system catch up in the next iteration(s)
ALOGV("DelayNs: %lld, catching-up", (long long) deltaNs);
deltaNs = 0;
}
// update sleep time (which is >= 0)
mSleepTimeUs = deltaNs / 1000;
}
if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
mWaitWorkCV.wait_for(_l, std::chrono::microseconds(mSleepTimeUs));
}
ATRACE_END();
}
}
// Finally let go of removed track(s), without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock. This will also mutate and push a new fast mixer state.
threadLoop_removeTracks(tracksToRemove);
tracksToRemove.clear();
// FIXME I don't understand the need for this here;
// it was in the original code but maybe the
// assignment in saveOutputTracks() makes this unnecessary?
clearOutputTracks();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
// FIXME Note that the above .clear() is no longer necessary since effectChains
// is now local to this block, but will keep it for now (at least until merge done).
}
threadLoop_exit();
if (!mStandby) {
threadLoop_standby();
setStandby();
}
releaseWakeLock();
ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
void PlaybackThread::collectTimestamps_l()
{
if (mStandby) {
mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
return;
} else if (mHwPaused) {
mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS);
return;
}
// Gather the framesReleased counters for all active tracks,
// and associate with the sink frames written out. We need
// this to convert the sink timestamp to the track timestamp.
bool kernelLocationUpdate = false;
ExtendedTimestamp timestamp; // use private copy to fetch
// Always query HAL timestamp and update timestamp verifier. In standby or pause,
// HAL may be draining some small duration buffered data for fade out.
if (threadloop_getHalTimestamp_l(&timestamp) == OK) {
mTimestampVerifier.add(timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
mSampleRate);
if (isTimestampCorrectionEnabled_l()) {
ALOGVV("TS_BEFORE: %d %lld %lld", id(),
(long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
(long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
auto correctedTimestamp = mTimestampVerifier.getLastCorrectedTimestamp();
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
= correctedTimestamp.mFrames;
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
= correctedTimestamp.mTimeNs;
ALOGVV("TS_AFTER: %d %lld %lld", id(),
(long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
(long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
// Note: Downstream latency only added if timestamp correction enabled.
if (mDownstreamLatencyStatMs.getN() > 0) { // we have latency info.
const int64_t newPosition =
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- int64_t(mDownstreamLatencyStatMs.getMean() * mSampleRate * 1e-3);
// prevent retrograde
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = max(
newPosition,
(mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- mSuspendedFrames));
}
}
// We always fetch the timestamp here because often the downstream
// sink will block while writing.
// We keep track of the last valid kernel position in case we are in underrun
// and the normal mixer period is the same as the fast mixer period, or there
// is some error from the HAL.
if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
}
if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
kernelLocationUpdate = true;
} else {
ALOGVV("getTimestamp error - no valid kernel position");
}
// copy over kernel info
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ mSuspendedFrames; // add frames discarded when suspended
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
} else {
mTimestampVerifier.error();
}
// mFramesWritten for non-offloaded tracks are contiguous
// even after standby() is called. This is useful for the track frame
// to sink frame mapping.
bool serverLocationUpdate = false;
if (mFramesWritten != mLastFramesWritten) {
serverLocationUpdate = true;
mLastFramesWritten = mFramesWritten;
}
// Only update timestamps if there is a meaningful change.
// Either the kernel timestamp must be valid or we have written something.
if (kernelLocationUpdate || serverLocationUpdate) {
if (serverLocationUpdate) {
// use the time before we called the HAL write - it is a bit more accurate
// to when the server last read data than the current time here.
//
// If we haven't written anything, mLastIoBeginNs will be -1
// and we use systemTime().
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastIoBeginNs == -1
? systemTime() : (int64_t)mLastIoBeginNs;
}
for (const sp<IAfTrack>& t : mActiveTracks) {
if (!t->isFastTrack()) {
t->updateTrackFrameInfo(
t->audioTrackServerProxy()->framesReleased(),
mFramesWritten,
mSampleRate,
mTimestamp);
}
}
}
if (audio_has_proportional_frames(mFormat)) {
const double latencyMs = mTimestamp.getOutputServerLatencyMs(mSampleRate);
if (latencyMs != 0.) { // note 0. means timestamp is empty.
mLatencyMs.add(latencyMs);
}
}
#if 0
// logFormat example
if (z % 100 == 0) {
timespec ts;
clock_gettime(CLOCK_MONOTONIC, &ts);
LOGT("This is an integer %d, this is a float %f, this is my "
"pid %p %% %s %t", 42, 3.14, "and this is a timestamp", ts);
LOGT("A deceptive null-terminated string %\0");
}
++z;
#endif
}
// removeTracks_l() must be called with ThreadBase::mutex() held
void PlaybackThread::removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove)
NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mutex()
{
if (tracksToRemove.empty()) return;
// Block all incoming TrackHandle requests until we are finished with the release.
setThreadBusy_l(true);
for (const auto& track : tracksToRemove) {
ALOGV("%s(%d): removing track on session %d", __func__, track->id(), track->sessionId());
sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
ALOGV("%s(%d): stopping track on chain %p for session Id: %d",
__func__, track->id(), chain.get(), track->sessionId());
chain->decActiveTrackCnt();
}
// If an external client track, inform APM we're no longer active, and remove if needed.
// Since the track is active, we do it here instead of TrackBase::destroy().
if (track->isExternalTrack()) {
mutex().unlock();
AudioSystem::stopOutput(track->portId());
if (track->isTerminated()) {
AudioSystem::releaseOutput(track->portId());
}
mutex().lock();
}
if (mHapticChannelCount > 0 &&
((track->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
|| (chain != nullptr && chain->containsHapticGeneratingEffect_l()))) {
mutex().unlock();
// Unlock due to VibratorService will lock for this call and will
// call Tracks.mute/unmute which also require thread's lock.
afutils::onExternalVibrationStop(track->getExternalVibration());
mutex().lock();
// When the track is stop, set the haptic intensity as MUTE
// for the HapticGenerator effect.
if (chain != nullptr) {
chain->setHapticScale_l(track->id(), os::HapticScale::mute());
}
}
// Under lock, the track is removed from the active tracks list.
//
// Once the track is no longer active, the TrackHandle may directly
// modify it as the threadLoop() is no longer responsible for its maintenance.
// Do not modify the track from threadLoop after the mutex is unlocked
// if it is not active.
mActiveTracks.remove(track);
if (track->isTerminated()) {
// remove from our tracks vector
removeTrack_l(track);
}
}
// Allow incoming TrackHandle requests. We still hold the mutex,
// so pending TrackHandle requests will occur after we unlock it.
setThreadBusy_l(false);
}
status_t PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
{
if (mNormalSink != 0) {
ExtendedTimestamp ets;
status_t status = mNormalSink->getTimestamp(ets);
if (status == NO_ERROR) {
status = ets.getBestTimestamp(&timestamp);
}
return status;
}
if ((mType == OFFLOAD || mType == DIRECT) && mOutput != NULL) {
collectTimestamps_l();
if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] <= 0) {
return INVALID_OPERATION;
}
timestamp.mPosition = mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
const int64_t timeNs = mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
timestamp.mTime.tv_sec = timeNs / NANOS_PER_SECOND;
timestamp.mTime.tv_nsec = timeNs - (timestamp.mTime.tv_sec * NANOS_PER_SECOND);
return NO_ERROR;
}
return INVALID_OPERATION;
}
// For dedicated VoIP outputs, let the HAL apply the stream volume. Track volume is
// still applied by the mixer.
// All tracks attached to a mixer with flag VOIP_RX are tied to the same
// stream type STREAM_VOICE_CALL so this will only change the HAL volume once even
// if more than one track are active
status_t PlaybackThread::handleVoipVolume_l(float* volume)
{
status_t result = NO_ERROR;
if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
if (*volume != mLeftVolFloat) {
result = mOutput->stream->setVolume(*volume, *volume);
// HAL can return INVALID_OPERATION if operation is not supported.
ALOGE_IF(result != OK && result != INVALID_OPERATION,
"Error when setting output stream volume: %d", result);
if (result == NO_ERROR) {
mLeftVolFloat = *volume;
}
}
// if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we
// remove stream volume contribution from software volume.
if (mLeftVolFloat == *volume) {
*volume = 1.0f;
}
}
return result;
}
status_t MixerThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
{
status_t status;
if (property_get_bool("af.patch_park", false /* default_value */)) {
// Park FastMixer to avoid potential DOS issues with writing to the HAL
// or if HAL does not properly lock against access.
AutoPark<FastMixer> park(mFastMixer);
status = PlaybackThread::createAudioPatch_l(patch, handle);
} else {
status = PlaybackThread::createAudioPatch_l(patch, handle);
}
updateHalSupportedLatencyModes_l();
return status;
}
status_t PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
status_t status = NO_ERROR;
// store new device and send to effects
audio_devices_t type = AUDIO_DEVICE_NONE;
AudioDeviceTypeAddrVector deviceTypeAddrs;
for (unsigned int i = 0; i < patch->num_sinks; i++) {
LOG_ALWAYS_FATAL_IF(popcount(patch->sinks[i].ext.device.type) > 1
&& !mOutput->audioHwDev->supportsAudioPatches(),
"Enumerated device type(%#x) must not be used "
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
deviceTypeAddrs.emplace_back(patch->sinks[i].ext.device.type,
patch->sinks[i].ext.device.address);
}
audio_port_handle_t sinkPortId = patch->sinks[0].id;
#ifdef ADD_BATTERY_DATA
// when changing the audio output device, call addBatteryData to notify
// the change
if (outDeviceTypes() != deviceTypes) {
uint32_t params = 0;
// check whether speaker is on
if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER) > 0) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
// check if any other device (except speaker) is on
if (!isSingleDeviceType(deviceTypes, AUDIO_DEVICE_OUT_SPEAKER)) {
params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
}
if (params != 0) {
addBatteryData(params);
}
}
#endif
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevices_l(deviceTypeAddrs);
}
// mPatch.num_sinks is not set when the thread is created so that
// the first patch creation triggers an ioConfigChanged callback
bool configChanged = (mPatch.num_sinks == 0) ||
(mPatch.sinks[0].id != sinkPortId);
mPatch = *patch;
mOutDeviceTypeAddrs = deviceTypeAddrs;
checkSilentMode_l();
if (mOutput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
status = hwDevice->createAudioPatch(patch->num_sources,
patch->sources,
patch->num_sinks,
patch->sinks,
handle);
} else {
status = mOutput->stream->legacyCreateAudioPatch(patch->sinks[0], std::nullopt, type);
*handle = AUDIO_PATCH_HANDLE_NONE;
}
const std::string patchSinksAsString = patchSinksToString(patch);
mThreadMetrics.logEndInterval();
mThreadMetrics.logCreatePatch(/* inDevices */ {}, patchSinksAsString);
mThreadMetrics.logBeginInterval();
// also dispatch to active AudioTracks for MediaMetrics
for (const auto &track : mActiveTracks) {
track->logEndInterval();
track->logBeginInterval(patchSinksAsString);
}
if (configChanged) {
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
}
// Force metadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
status_t MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status;
if (property_get_bool("af.patch_park", false /* default_value */)) {
// Park FastMixer to avoid potential DOS issues with writing to the HAL
// or if HAL does not properly lock against access.
AutoPark<FastMixer> park(mFastMixer);
status = PlaybackThread::releaseAudioPatch_l(handle);
} else {
status = PlaybackThread::releaseAudioPatch_l(handle);
}
return status;
}
status_t PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
mPatch = audio_patch{};
mOutDeviceTypeAddrs.clear();
if (mOutput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
status = hwDevice->releaseAudioPatch(handle);
} else {
status = mOutput->stream->legacyReleaseAudioPatch();
}
// Force meteadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
void PlaybackThread::addPatchTrack(const sp<IAfPatchTrack>& track)
{
audio_utils::lock_guard _l(mutex());
mTracks.add(track);
}
void PlaybackThread::deletePatchTrack(const sp<IAfPatchTrack>& track)
{
audio_utils::lock_guard _l(mutex());
destroyTrack_l(track);
}
void PlaybackThread::toAudioPortConfig(struct audio_port_config* config)
{
ThreadBase::toAudioPortConfig(config);
config->role = AUDIO_PORT_ROLE_SOURCE;
config->ext.mix.hw_module = mOutput->audioHwDev->handle();
config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
config->flags.output = mOutput->flags;
}
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfPlaybackThread> IAfPlaybackThread::createMixerThread(
const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t* mixerConfig) {
return sp<MixerThread>::make(afThreadCallback, output, id, systemReady, type, mixerConfig);
}
MixerThread::MixerThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
: PlaybackThread(afThreadCallback, output, id, type, systemReady, mixerConfig),
// mAudioMixer below
// mFastMixer below
mBluetoothLatencyModesEnabled(false),
mFastMixerFutex(0),
mMasterMono(false)
// mOutputSink below
// mPipeSink below
// mNormalSink below
{
setMasterBalance(afThreadCallback->getMasterBalance_l());
ALOGV("MixerThread() id=%d type=%d", id, type);
ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
"mFrameCount=%zu, mNormalFrameCount=%zu",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
mNormalFrameCount);
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
if (type == DUPLICATING) {
// The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
// (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
// Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
return;
}
// create an NBAIO sink for the HAL output stream, and negotiate
mOutputSink = new AudioStreamOutSink(output->stream);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {Format_from_SR_C(
mSampleRate, mChannelCount + mHapticChannelCount, mFormat)};
#if !LOG_NDEBUG
ssize_t index =
#else
(void)
#endif
mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
// initialize fast mixer depending on configuration
bool initFastMixer;
if (mType == SPATIALIZER || mType == BIT_PERFECT) {
initFastMixer = false;
} else {
switch (kUseFastMixer) {
case FastMixer_Never:
initFastMixer = false;
break;
case FastMixer_Always:
initFastMixer = true;
break;
case FastMixer_Static:
case FastMixer_Dynamic:
initFastMixer = mFrameCount < mNormalFrameCount;
break;
}
ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount,
"FastMixer is preferred for this sink as frameCount %zu is less than threshold %zu",
mFrameCount, mNormalFrameCount);
}
if (initFastMixer) {
audio_format_t fastMixerFormat;
if (mMixerBufferEnabled && mEffectBufferEnabled) {
fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
} else {
fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
}
if (mFormat != fastMixerFormat) {
// change our Sink format to accept our intermediate precision
mFormat = fastMixerFormat;
free(mSinkBuffer);
mFrameSize = audio_bytes_per_frame(mChannelCount + mHapticChannelCount, mFormat);
const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
(void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
}
// create a MonoPipe to connect our submix to FastMixer
NBAIO_Format format = mOutputSink->format();
// adjust format to match that of the Fast Mixer
ALOGV("format changed from %#x to %#x", format.mFormat, fastMixerFormat);
format.mFormat = fastMixerFormat;
format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
// This pipe depth compensates for scheduling latency of the normal mixer thread.
// When it wakes up after a maximum latency, it runs a few cycles quickly before
// finally blocking. Note the pipe implementation rounds up the request to a power of 2.
MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
const NBAIO_Format offersFast[1] = {format};
size_t numCounterOffersFast = 0;
#if !LOG_NDEBUG
index =
#else
(void)
#endif
monoPipe->negotiate(offersFast, std::size(offersFast),
nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index == 0);
monoPipe->setAvgFrames((mScreenState & 1) ?
(monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
mPipeSink = monoPipe;
// create fast mixer and configure it initially with just one fast track for our submix
mFastMixer = new FastMixer(mId);
FastMixerStateQueue *sq = mFastMixer->sq();
#ifdef STATE_QUEUE_DUMP
sq->setObserverDump(&mStateQueueObserverDump);
sq->setMutatorDump(&mStateQueueMutatorDump);
#endif
FastMixerState *state = sq->begin();
FastTrack *fastTrack = &state->mFastTracks[0];
// wrap the source side of the MonoPipe to make it an AudioBufferProvider
fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
fastTrack->mVolumeProvider = NULL;
fastTrack->mChannelMask = static_cast<audio_channel_mask_t>(
mChannelMask | mHapticChannelMask); // mPipeSink channel mask for
// audio to FastMixer
fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
fastTrack->mHapticPlaybackEnabled = mHapticChannelMask != AUDIO_CHANNEL_NONE;
fastTrack->mHapticScale = {/*level=*/os::HapticLevel::NONE };
fastTrack->mHapticMaxAmplitude = NAN;
fastTrack->mGeneration++;
state->mFastTracksGen++;
state->mTrackMask = 1;
// fast mixer will use the HAL output sink
state->mOutputSink = mOutputSink.get();
state->mOutputSinkGen++;
state->mFrameCount = mFrameCount;
// specify sink channel mask when haptic channel mask present as it can not
// be calculated directly from channel count
state->mSinkChannelMask = mHapticChannelMask == AUDIO_CHANNEL_NONE
? AUDIO_CHANNEL_NONE
: static_cast<audio_channel_mask_t>(mChannelMask | mHapticChannelMask);
state->mCommand = FastMixerState::COLD_IDLE;
// already done in constructor initialization list
//mFastMixerFutex = 0;
state->mColdFutexAddr = &mFastMixerFutex;
state->mColdGen++;
state->mDumpState = &mFastMixerDumpState;
mFastMixerNBLogWriter = afThreadCallback->newWriter_l(kFastMixerLogSize, "FastMixer");
state->mNBLogWriter = mFastMixerNBLogWriter.get();
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
NBLog::thread_info_t info;
info.id = mId;
info.type = NBLog::FASTMIXER;
mFastMixerNBLogWriter->log<NBLog::EVENT_THREAD_INFO>(info);
// start the fast mixer
mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
pid_t tid = mFastMixer->getTid();
sendPrioConfigEvent(getpid(), tid, kPriorityFastMixer, false /*forApp*/);
stream()->setHalThreadPriority(kPriorityFastMixer);
#ifdef AUDIO_WATCHDOG
// create and start the watchdog
mAudioWatchdog = new AudioWatchdog();
mAudioWatchdog->setDump(&mAudioWatchdogDump);
mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
tid = mAudioWatchdog->getTid();
sendPrioConfigEvent(getpid(), tid, kPriorityFastMixer, false /*forApp*/);
#endif
} else {
#ifdef TEE_SINK
// Only use the MixerThread tee if there is no FastMixer.
mTee.set(mOutputSink->format(), NBAIO_Tee::TEE_FLAG_OUTPUT_THREAD);
mTee.setId(std::string("_") + std::to_string(mId) + "_M");
#endif
}
switch (kUseFastMixer) {
case FastMixer_Never:
case FastMixer_Dynamic:
mNormalSink = mOutputSink;
break;
case FastMixer_Always:
mNormalSink = mPipeSink;
break;
case FastMixer_Static:
mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
break;
}
}
MixerThread::~MixerThread()
{
if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (state->mCommand == FastMixerState::COLD_IDLE) {
int32_t old = android_atomic_inc(&mFastMixerFutex);
if (old == -1) {
(void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
}
}
state->mCommand = FastMixerState::EXIT;
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
mFastMixer->join();
// Though the fast mixer thread has exited, it's state queue is still valid.
// We'll use that extract the final state which contains one remaining fast track
// corresponding to our sub-mix.
state = sq->begin();
ALOG_ASSERT(state->mTrackMask == 1);
FastTrack *fastTrack = &state->mFastTracks[0];
ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
delete fastTrack->mBufferProvider;
sq->end(false /*didModify*/);
mFastMixer.clear();
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
mAudioWatchdog->requestExit();
mAudioWatchdog->requestExitAndWait();
mAudioWatchdog.clear();
}
#endif
}
mAfThreadCallback->unregisterWriter(mFastMixerNBLogWriter);
delete mAudioMixer;
}
void MixerThread::onFirstRef() {
PlaybackThread::onFirstRef();
audio_utils::lock_guard _l(mutex());
if (mOutput != nullptr && mOutput->stream != nullptr) {
status_t status = mOutput->stream->setLatencyModeCallback(this);
if (status != INVALID_OPERATION) {
updateHalSupportedLatencyModes_l();
}
// Default to enabled if the HAL supports it. This can be changed by Audioflinger after
// the thread construction according to AudioFlinger::mBluetoothLatencyModesEnabled
mBluetoothLatencyModesEnabled.store(
mOutput->audioHwDev->supportsBluetoothVariableLatency());
}
}
uint32_t MixerThread::correctLatency_l(uint32_t latency) const
{
if (mFastMixer != 0) {
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
}
return latency;
}
ssize_t MixerThread::threadLoop_write()
{
// FIXME we should only do one push per cycle; confirm this is true
// Start the fast mixer if it's not already running
if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (state->mCommand != FastMixerState::MIX_WRITE &&
(kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
if (state->mCommand == FastMixerState::COLD_IDLE) {
// FIXME workaround for first HAL write being CPU bound on some devices
ATRACE_BEGIN("write");
mOutput->write((char *)mSinkBuffer, 0);
ATRACE_END();
int32_t old = android_atomic_inc(&mFastMixerFutex);
if (old == -1) {
(void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
}
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
mAudioWatchdog->resume();
}
#endif
}
state->mCommand = FastMixerState::MIX_WRITE;
#ifdef FAST_THREAD_STATISTICS
mFastMixerDumpState.increaseSamplingN(mAfThreadCallback->isLowRamDevice() ?
FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
#endif
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mPipeSink;
}
} else {
sq->end(false /*didModify*/);
}
}
return PlaybackThread::threadLoop_write();
}
void MixerThread::threadLoop_standby()
{
// Idle the fast mixer if it's currently running
if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (!(state->mCommand & FastMixerState::IDLE)) {
// Report any frames trapped in the Monopipe
MonoPipe *monoPipe = (MonoPipe *)mPipeSink.get();
const long long pipeFrames = monoPipe->maxFrames() - monoPipe->availableToWrite();
mLocalLog.log("threadLoop_standby: framesWritten:%lld suspendedFrames:%lld "
"monoPipeWritten:%lld monoPipeLeft:%lld",
(long long)mFramesWritten, (long long)mSuspendedFrames,
(long long)mPipeSink->framesWritten(), pipeFrames);
mLocalLog.log("threadLoop_standby: %s", mTimestamp.toString().c_str());
state->mCommand = FastMixerState::COLD_IDLE;
state->mColdFutexAddr = &mFastMixerFutex;
state->mColdGen++;
mFastMixerFutex = 0;
sq->end();
// BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mOutputSink;
}
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
mAudioWatchdog->pause();
}
#endif
} else {
sq->end(false /*didModify*/);
}
}
PlaybackThread::threadLoop_standby();
}
bool PlaybackThread::waitingAsyncCallback_l()
{
return false;
}
bool PlaybackThread::shouldStandby_l()
{
return !mStandby;
}
bool PlaybackThread::waitingAsyncCallback()
{
audio_utils::lock_guard _l(mutex());
return waitingAsyncCallback_l();
}
// shared by MIXER and DIRECT, overridden by DUPLICATING
void PlaybackThread::threadLoop_standby()
{
ALOGV("%s: audio hardware entering standby, mixer %p, suspend count %d",
__func__, this, (int32_t)mSuspended);
mOutput->standby();
if (mUseAsyncWrite != 0) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
mDrainSequence = (mDrainSequence + 2) & ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
mCallbackThread->setDraining(mDrainSequence);
}
mHwPaused = false;
setHalLatencyMode_l();
}
void PlaybackThread::onAddNewTrack_l()
{
ALOGV("signal playback thread");
broadcast_l();
}
void PlaybackThread::onAsyncError()
{
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
invalidateTracks((audio_stream_type_t)i);
}
}
void MixerThread::threadLoop_mix()
{
// mix buffers...
mAudioMixer->process();
mCurrentWriteLength = mSinkBufferSize;
// increase sleep time progressively when application underrun condition clears.
// Only increase sleep time if the mixer is ready for two consecutive times to avoid
// that a steady state of alternating ready/not ready conditions keeps the sleep time
// such that we would underrun the audio HAL.
if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
sleepTimeShift--;
}
mSleepTimeUs = 0;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
//TODO: delay standby when effects have a tail
}
void MixerThread::threadLoop_sleepTime()
{
// If no tracks are ready, sleep once for the duration of an output
// buffer size, then write 0s to the output
if (mSleepTimeUs == 0) {
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
if (mPipeSink.get() != nullptr && mPipeSink == mNormalSink) {
// Using the Monopipe availableToWrite, we estimate the
// sleep time to retry for more data (before we underrun).
MonoPipe *monoPipe = static_cast<MonoPipe *>(mPipeSink.get());
const ssize_t availableToWrite = mPipeSink->availableToWrite();
const size_t pipeFrames = monoPipe->maxFrames();
const size_t framesLeft = pipeFrames - max(availableToWrite, 0);
// HAL_framecount <= framesDelay ~ framesLeft / 2 <= Normal_Mixer_framecount
const size_t framesDelay = std::min(
mNormalFrameCount, max(framesLeft / 2, mFrameCount));
ALOGV("pipeFrames:%zu framesLeft:%zu framesDelay:%zu",
pipeFrames, framesLeft, framesDelay);
mSleepTimeUs = framesDelay * MICROS_PER_SECOND / mSampleRate;
} else {
mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift;
if (mSleepTimeUs < kMinThreadSleepTimeUs) {
mSleepTimeUs = kMinThreadSleepTimeUs;
}
// reduce sleep time in case of consecutive application underruns to avoid
// starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
// duration we would end up writing less data than needed by the audio HAL if
// the condition persists.
if (sleepTimeShift < kMaxThreadSleepTimeShift) {
sleepTimeShift++;
}
}
} else {
mSleepTimeUs = mIdleSleepTimeUs;
}
} else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
// clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
// before effects processing or output.
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
if (mType == SPATIALIZER) {
memset(mSinkBuffer, 0, mSinkBufferSize);
}
} else {
memset(mSinkBuffer, 0, mSinkBufferSize);
}
mSleepTimeUs = 0;
ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
"anticipated start");
}
// TODO add standby time extension fct of effect tail
}
// prepareTracks_l() must be called with ThreadBase::mutex() held
PlaybackThread::mixer_state MixerThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove)
{
// clean up deleted track ids in AudioMixer before allocating new tracks
(void)mTracks.processDeletedTrackIds([this](int trackId) {
// for each trackId, destroy it in the AudioMixer
if (mAudioMixer->exists(trackId)) {
mAudioMixer->destroy(trackId);
}
});
mTracks.clearDeletedTrackIds();
mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = mActiveTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
// counts only _active_ fast tracks
size_t fastTracks = 0;
uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
}
// Delegate master volume control to effect in output mix effect chain if needed
sp<IAfEffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume(&v, &v);
masterVolume = (float)((v + (1 << 23)) >> 24);
chain.clear();
}
// prepare a new state to push
FastMixerStateQueue *sq = NULL;
FastMixerState *state = NULL;
bool didModify = false;
FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
bool coldIdle = false;
if (mFastMixer != 0) {
sq = mFastMixer->sq();
state = sq->begin();
coldIdle = state->mCommand == FastMixerState::COLD_IDLE;
}
mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
// DeferredOperations handles statistics after setting mixerStatus.
class DeferredOperations {
public:
DeferredOperations(mixer_state *mixerStatus, ThreadMetrics *threadMetrics)
: mMixerStatus(mixerStatus)
, mThreadMetrics(threadMetrics) {}
// when leaving scope, tally frames properly.
~DeferredOperations() {
// Tally underrun frames only if we are actually mixing (MIXER_TRACKS_READY)
// because that is when the underrun occurs.
// We do not distinguish between FastTracks and NormalTracks here.
size_t maxUnderrunFrames = 0;
if (*mMixerStatus == MIXER_TRACKS_READY && mUnderrunFrames.size() > 0) {
for (const auto &underrun : mUnderrunFrames) {
underrun.first->tallyUnderrunFrames(underrun.second);
maxUnderrunFrames = max(underrun.second, maxUnderrunFrames);
}
}
// send the max underrun frames for this mixer period
mThreadMetrics->logUnderrunFrames(maxUnderrunFrames);
}
// tallyUnderrunFrames() is called to update the track counters
// with the number of underrun frames for a particular mixer period.
// We defer tallying until we know the final mixer status.
void tallyUnderrunFrames(const sp<IAfTrack>& track, size_t underrunFrames) {
mUnderrunFrames.emplace_back(track, underrunFrames);
}
private:
const mixer_state * const mMixerStatus;
ThreadMetrics * const mThreadMetrics;
std::vector<std::pair<sp<IAfTrack>, size_t>> mUnderrunFrames;
} deferredOperations(&mixerStatus, &mThreadMetrics);
// implicit nested scope for variable capture
bool noFastHapticTrack = true;
for (size_t i=0 ; i<count ; i++) {
const sp<IAfTrack> t = mActiveTracks[i];
// this const just means the local variable doesn't change
IAfTrack* const track = t.get();
// process fast tracks
if (track->isFastTrack()) {
LOG_ALWAYS_FATAL_IF(mFastMixer.get() == nullptr,
"%s(%d): FastTrack(%d) present without FastMixer",
__func__, id(), track->id());
if (track->getHapticPlaybackEnabled()) {
noFastHapticTrack = false;
}
// It's theoretically possible (though unlikely) for a fast track to be created
// and then removed within the same normal mix cycle. This is not a problem, as
// the track never becomes active so it's fast mixer slot is never touched.
// The converse, of removing an (active) track and then creating a new track
// at the identical fast mixer slot within the same normal mix cycle,
// is impossible because the slot isn't marked available until the end of each cycle.
int j = track->fastIndex();
ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks);
ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
FastTrack *fastTrack = &state->mFastTracks[j];
// Determine whether the track is currently in underrun condition,
// and whether it had a recent underrun.
FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
FastTrackUnderruns underruns = ftDump->mUnderruns;
uint32_t recentFull = (underruns.mBitFields.mFull -
track->fastTrackUnderruns().mBitFields.mFull) & UNDERRUN_MASK;
uint32_t recentPartial = (underruns.mBitFields.mPartial -
track->fastTrackUnderruns().mBitFields.mPartial) & UNDERRUN_MASK;
uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
track->fastTrackUnderruns().mBitFields.mEmpty) & UNDERRUN_MASK;
uint32_t recentUnderruns = recentPartial + recentEmpty;
track->fastTrackUnderruns() = underruns;
// don't count underruns that occur while stopping or pausing
// or stopped which can occur when flush() is called while active
size_t underrunFrames = 0;
if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
recentUnderruns > 0) {
// FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
underrunFrames = recentUnderruns * mFrameCount;
}
// Immediately account for FastTrack underruns.
track->audioTrackServerProxy()->tallyUnderrunFrames(underrunFrames);
// This is similar to the state machine for normal tracks,
// with a few modifications for fast tracks.
bool isActive = true;
switch (track->state()) {
case IAfTrackBase::STOPPING_1:
// track stays active in STOPPING_1 state until first underrun
if (recentUnderruns > 0 || track->isTerminated()) {
track->setState(IAfTrackBase::STOPPING_2);
}
break;
case IAfTrackBase::PAUSING:
// ramp down is not yet implemented
track->setPaused();
break;
case IAfTrackBase::RESUMING:
// ramp up is not yet implemented
track->setState(IAfTrackBase::ACTIVE);
break;
case IAfTrackBase::ACTIVE:
if (recentFull > 0 || recentPartial > 0) {
// track has provided at least some frames recently: reset retry count
track->retryCount() = kMaxTrackRetries;
}
if (recentUnderruns == 0) {
// no recent underruns: stay active
break;
}
// there has recently been an underrun of some kind
if (track->sharedBuffer() == 0) {
// were any of the recent underruns "empty" (no frames available)?
if (recentEmpty == 0) {
// no, then ignore the partial underruns as they are allowed indefinitely
break;
}
// there has recently been an "empty" underrun: decrement the retry counter
if (--(track->retryCount()) > 0) {
break;
}
// indicate to client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
track->disable();
// remove from active list, but state remains ACTIVE [confusing but true]
isActive = false;
break;
}
FALLTHROUGH_INTENDED;
case IAfTrackBase::STOPPING_2:
case IAfTrackBase::PAUSED:
case IAfTrackBase::STOPPED:
case IAfTrackBase::FLUSHED: // flush() while active
// Check for presentation complete if track is inactive
// We have consumed all the buffers of this track.
// This would be incomplete if we auto-paused on underrun
{
uint32_t latency = 0;
status_t result = mOutput->stream->getLatency(&latency);
ALOGE_IF(result != OK,
"Error when retrieving output stream latency: %d", result);
size_t audioHALFrames = (latency * mSampleRate) / 1000;
int64_t framesWritten = mBytesWritten / mFrameSize;
if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
// track stays in active list until presentation is complete
break;
}
}
if (track->isStopping_2()) {
track->setState(IAfTrackBase::STOPPED);
}
if (track->isStopped()) {
// Can't reset directly, as fast mixer is still polling this track
// track->reset();
// So instead mark this track as needing to be reset after push with ack
resetMask |= 1 << i;
}
isActive = false;
break;
case IAfTrackBase::IDLE:
default:
LOG_ALWAYS_FATAL("unexpected track state %d", (int)track->state());
}
if (isActive) {
// was it previously inactive?
if (!(state->mTrackMask & (1 << j))) {
ExtendedAudioBufferProvider *eabp = track->asExtendedAudioBufferProvider();
VolumeProvider *vp = track->asVolumeProvider();
fastTrack->mBufferProvider = eabp;
fastTrack->mVolumeProvider = vp;
fastTrack->mChannelMask = track->channelMask();
fastTrack->mFormat = track->format();
fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
fastTrack->mHapticScale = track->getHapticScale();
fastTrack->mHapticMaxAmplitude = track->getHapticMaxAmplitude();
fastTrack->mGeneration++;
state->mTrackMask |= 1 << j;
didModify = true;
// no acknowledgement required for newly active tracks
}
sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
float volume;
if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
volume = 0.f;
} else {
volume = masterVolume * mStreamTypes[track->streamType()].volume;
}
handleVoipVolume_l(&volume);
// cache the combined master volume and stream type volume for fast mixer; this
// lacks any synchronization or barrier so VolumeProvider may read a stale value
const float vh = track->getVolumeHandler()->getVolume(
proxy->framesReleased()).first;
volume *= vh;
track->setCachedVolume(volume);
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
/*muteState=*/{masterVolume == 0.f,
mStreamTypes[track->streamType()].volume == 0.f,
mStreamTypes[track->streamType()].mute,
track->isPlaybackRestricted(),
vlf == 0.f && vrf == 0.f,
vh == 0.f});
vlf *= volume;
vrf *= volume;
track->setFinalVolume(vlf, vrf);
++fastTracks;
} else {
// was it previously active?
if (state->mTrackMask & (1 << j)) {
fastTrack->mBufferProvider = NULL;
fastTrack->mGeneration++;
state->mTrackMask &= ~(1 << j);
didModify = true;
// If any fast tracks were removed, we must wait for acknowledgement
// because we're about to decrement the last sp<> on those tracks.
block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
} else {
// ALOGW rather than LOG_ALWAYS_FATAL because it seems there are cases where an
// AudioTrack may start (which may not be with a start() but with a write()
// after underrun) and immediately paused or released. In that case the
// FastTrack state hasn't had time to update.
// TODO Remove the ALOGW when this theory is confirmed.
ALOGW("fast track %d should have been active; "
"mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
j, (int)track->state(), state->mTrackMask, recentUnderruns,
track->sharedBuffer() != 0);
// Since the FastMixer state already has the track inactive, do nothing here.
}
tracksToRemove->add(track);
// Avoids a misleading display in dumpsys
track->fastTrackUnderruns().mBitFields.mMostRecent = UNDERRUN_FULL;
}
if (fastTrack->mHapticPlaybackEnabled != track->getHapticPlaybackEnabled()) {
fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
didModify = true;
}
continue;
}
{ // local variable scope to avoid goto warning
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
const int trackId = track->id();
// if an active track doesn't exist in the AudioMixer, create it.
// use the trackId as the AudioMixer name.
if (!mAudioMixer->exists(trackId)) {
status_t status = mAudioMixer->create(
trackId,
track->channelMask(),
track->format(),
track->sessionId());
if (status != OK) {
ALOGW("%s(): AudioMixer cannot create track(%d)"
" mask %#x, format %#x, sessionId %d",
__func__, trackId,
track->channelMask(), track->format(), track->sessionId());
tracksToRemove->add(track);
track->invalidate(); // consider it dead.
continue;
}
}
// make sure that we have enough frames to mix one full buffer.
// enforce this condition only once to enable draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
// hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
// during last round
size_t desiredFrames;
const uint32_t sampleRate = track->audioTrackServerProxy()->getSampleRate();
const AudioPlaybackRate playbackRate = track->audioTrackServerProxy()->getPlaybackRate();
desiredFrames = sourceFramesNeededWithTimestretch(
sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
// TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(trackId);
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
minFrames = desiredFrames;
}
size_t framesReady = track->framesReady();
if (ATRACE_ENABLED()) {
// I wish we had formatted trace names
std::string traceName("nRdy");
traceName += std::to_string(trackId);
ATRACE_INT(traceName.c_str(), framesReady);
}
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
ALOGVV("track(%d) s=%08x [OK] on thread %p", trackId, cblk->mServer, this);
mixedTracks++;
// track->mainBuffer() != mSinkBuffer and mMixerBuffer means
// there is an effect chain connected to the track
chain.clear();
if (track->mainBuffer() != mSinkBuffer &&
track->mainBuffer() != mMixerBuffer) {
if (mEffectBufferEnabled) {
mEffectBufferValid = true; // Later can set directly.
}
chain = getEffectChain_l(track->sessionId());
// Delegate volume control to effect in track effect chain if needed
if (chain != 0) {
tracksWithEffect++;
} else {
ALOGW("prepareTracks_l(): track(%d) attached to effect but no chain found on "
"session %d",
trackId, track->sessionId());
}
}
int param = AudioMixer::VOLUME;
if (track->fillingStatus() == IAfTrack::FS_FILLED) {
// no ramp for the first volume setting
track->fillingStatus() = IAfTrack::FS_ACTIVE;
if (track->state() == IAfTrackBase::RESUMING) {
track->setState(IAfTrackBase::ACTIVE);
// If a new track is paused immediately after start, do not ramp on resume.
if (cblk->mServer != 0) {
param = AudioMixer::RAMP_VOLUME;
}
}
mAudioMixer->setParameter(trackId, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
mLeftVolFloat = -1.0;
// FIXME should not make a decision based on mServer
} else if (cblk->mServer != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
param = AudioMixer::RAMP_VOLUME;
}
// compute volume for this track
uint32_t vl, vr; // in U8.24 integer format
float vlf, vrf, vaf; // in [0.0, 1.0] float format
// read original volumes with volume control
float v = masterVolume * mStreamTypes[track->streamType()].volume;
// Always fetch volumeshaper volume to ensure state is updated.
const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
const float vh = track->getVolumeHandler()->getVolume(
track->audioTrackServerProxy()->framesReleased()).first;
if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
v = 0;
}
handleVoipVolume_l(&v);
if (track->isPausing()) {
vl = vr = 0;
vlf = vrf = vaf = 0.;
track->setPaused();
} else {
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vlf > GAIN_FLOAT_UNITY) {
ALOGV("Track left volume out of range: %.3g", vlf);
vlf = GAIN_FLOAT_UNITY;
}
if (vrf > GAIN_FLOAT_UNITY) {
ALOGV("Track right volume out of range: %.3g", vrf);
vrf = GAIN_FLOAT_UNITY;
}
track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
/*muteState=*/{masterVolume == 0.f,
mStreamTypes[track->streamType()].volume == 0.f,
mStreamTypes[track->streamType()].mute,
track->isPlaybackRestricted(),
vlf == 0.f && vrf == 0.f,
vh == 0.f});
// now apply the master volume and stream type volume and shaper volume
vlf *= v * vh;
vrf *= v * vh;
// assuming master volume and stream type volume each go up to 1.0,
// then derive vl and vr as U8.24 versions for the effect chain
const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
vl = (uint32_t) (scaleto8_24 * vlf);
vr = (uint32_t) (scaleto8_24 * vrf);
// vl and vr are now in U8.24 format
uint16_t sendLevel = proxy->getSendLevel_U4_12();
// send level comes from shared memory and so may be corrupt
if (sendLevel > MAX_GAIN_INT) {
ALOGV("Track send level out of range: %04X", sendLevel);
sendLevel = MAX_GAIN_INT;
}
// vaf is represented as [0.0, 1.0] float by rescaling sendLevel
vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
track->setFinalVolume(vrf, vlf);
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume(&vl, &vr)) {
// Do not ramp volume if volume is controlled by effect
param = AudioMixer::VOLUME;
// Update remaining floating point volume levels
vlf = (float)vl / (1 << 24);
vrf = (float)vr / (1 << 24);
track->setHasVolumeController(true);
} else {
// force no volume ramp when volume controller was just disabled or removed
// from effect chain to avoid volume spike
if (track->hasVolumeController()) {
param = AudioMixer::VOLUME;
}
track->setHasVolumeController(false);
}
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(trackId, track->asExtendedAudioBufferProvider());
mAudioMixer->enable(trackId);
mAudioMixer->setParameter(trackId, param, AudioMixer::VOLUME0, &vlf);
mAudioMixer->setParameter(trackId, param, AudioMixer::VOLUME1, &vrf);
mAudioMixer->setParameter(trackId, param, AudioMixer::AUXLEVEL, &vaf);
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
if (mType == SPATIALIZER && !track->isSpatialized()) {
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK,
(void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
} else {
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK,
(void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
}
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
uint32_t reqSampleRate = proxy->getSampleRate();
if (reqSampleRate == 0) {
reqSampleRate = mSampleRate;
} else if (reqSampleRate > maxSampleRate) {
reqSampleRate = maxSampleRate;
}
mAudioMixer->setParameter(
trackId,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(uintptr_t)reqSampleRate);
mAudioMixer->setParameter(
trackId,
AudioMixer::TIMESTRETCH,
AudioMixer::PLAYBACK_RATE,
// cast away constness for this generic API.
const_cast<void *>(reinterpret_cast<const void *>(&playbackRate)));
/*
* Select the appropriate output buffer for the track.
*
* Tracks with effects go into their own effects chain buffer
* and from there into either mEffectBuffer or mSinkBuffer.
*
* Other tracks can use mMixerBuffer for higher precision
* channel accumulation. If this buffer is enabled
* (mMixerBufferEnabled true), then selected tracks will accumulate
* into it.
*
*/
if (mMixerBufferEnabled
&& (track->mainBuffer() == mSinkBuffer
|| track->mainBuffer() == mMixerBuffer)) {
if (mType == SPATIALIZER && !track->isSpatialized()) {
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mEffectBufferFormat);
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)mPostSpatializerBuffer);
} else {
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
// TODO: override track->mainBuffer()?
mMixerBufferValid = true;
}
} else {
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_FLOAT);
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
}
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::HAPTIC_ENABLED, (void *)(uintptr_t)track->getHapticPlaybackEnabled());
const os::HapticScale hapticScale = track->getHapticScale();
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::HAPTIC_SCALE, (void *)&hapticScale);
const float hapticMaxAmplitude = track->getHapticMaxAmplitude();
mAudioMixer->setParameter(
trackId,
AudioMixer::TRACK,
AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)&hapticMaxAmplitude);
// reset retry count
track->retryCount() = kMaxTrackRetries;
// If one track is ready, set the mixer ready if:
// - the mixer was not ready during previous round OR
// - no other track is not ready
if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_ENABLED) {
mixerStatus = MIXER_TRACKS_READY;
}
// Enable the next few lines to instrument a test for underrun log handling.
// TODO: Remove when we have a better way of testing the underrun log.
#if 0
static int i;
if ((++i & 0xf) == 0) {
deferredOperations.tallyUnderrunFrames(track, 10 /* underrunFrames */);
}
#endif
} else {
size_t underrunFrames = 0;
if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
ALOGV("track(%d) underrun, track state %s framesReady(%zu) < framesDesired(%zd)",
trackId, track->getTrackStateAsString(), framesReady, desiredFrames);
underrunFrames = desiredFrames;
}
deferredOperations.tallyUnderrunFrames(track, underrunFrames);
// clear effect chain input buffer if an active track underruns to avoid sending
// previous audio buffer again to effects
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->clearInputBuffer();
}
ALOGVV("track(%d) s=%08x [NOT READY] on thread %p", trackId, cblk->mServer, this);
if ((track->sharedBuffer() != 0) || track->isTerminated() ||
track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
// TODO: use actual buffer filling status instead of latency when available from
// audio HAL
size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
int64_t framesWritten = mBytesWritten / mFrameSize;
if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
if (track->isStopped()) {
track->reset();
}
tracksToRemove->add(track);
}
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->retryCount()) <= 0) {
ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p",
trackId, this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
track->disable();
// If one track is not ready, mark the mixer also not ready if:
// - the mixer was ready during previous round OR
// - no other track is ready
} else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(trackId);
}
} // local variable scope to avoid goto warning
}
if (mHapticChannelMask != AUDIO_CHANNEL_NONE && sq != NULL) {
// When there is no fast track playing haptic and FastMixer exists,
// enabling the first FastTrack, which provides mixed data from normal
// tracks, to play haptic data.
FastTrack *fastTrack = &state->mFastTracks[0];
if (fastTrack->mHapticPlaybackEnabled != noFastHapticTrack) {
fastTrack->mHapticPlaybackEnabled = noFastHapticTrack;
didModify = true;
}
}
// Push the new FastMixer state if necessary
[[maybe_unused]] bool pauseAudioWatchdog = false;
if (didModify) {
state->mFastTracksGen++;
// if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
if (kUseFastMixer == FastMixer_Dynamic &&
state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
state->mCommand = FastMixerState::COLD_IDLE;
state->mColdFutexAddr = &mFastMixerFutex;
state->mColdGen++;
mFastMixerFutex = 0;
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mOutputSink;
}
// If we go into cold idle, need to wait for acknowledgement
// so that fast mixer stops doing I/O.
block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
pauseAudioWatchdog = true;
}
}
if (sq != NULL) {
sq->end(didModify);
// No need to block if the FastMixer is in COLD_IDLE as the FastThread
// is not active. (We BLOCK_UNTIL_ACKED when entering COLD_IDLE
// when bringing the output sink into standby.)
//
// We will get the latest FastMixer state when we come out of COLD_IDLE.
//
// This occurs with BT suspend when we idle the FastMixer with
// active tracks, which may be added or removed.
sq->push(coldIdle ? FastMixerStateQueue::BLOCK_NEVER : block);
}
#ifdef AUDIO_WATCHDOG
if (pauseAudioWatchdog && mAudioWatchdog != 0) {
mAudioWatchdog->pause();
}
#endif
// Now perform the deferred reset on fast tracks that have stopped
while (resetMask != 0) {
size_t i = __builtin_ctz(resetMask);
ALOG_ASSERT(i < count);
resetMask &= ~(1 << i);
sp<IAfTrack> track = mActiveTracks[i];
ALOG_ASSERT(track->isFastTrack() && track->isStopped());
track->reset();
}
// Track destruction may occur outside of threadLoop once it is removed from active tracks.
// Ensure the AudioMixer doesn't have a raw "buffer provider" pointer to the track if
// it ceases to be active, to allow safe removal from the AudioMixer at the start
// of prepareTracks_l(); this releases any outstanding buffer back to the track.
// See also the implementation of destroyTrack_l().
for (const auto &track : *tracksToRemove) {
const int trackId = track->id();
if (mAudioMixer->exists(trackId)) { // Normal tracks here, fast tracks in FastMixer.
mAudioMixer->setBufferProvider(trackId, nullptr /* bufferProvider */);
}
}
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0 ||
getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE) != 0) {
mEffectBufferValid = true;
}
if (mEffectBufferValid) {
// as long as there are effects we should clear the effects buffer, to avoid
// passing a non-clean buffer to the effect chain
memset(mEffectBuffer, 0, mEffectBufferSize);
if (mType == SPATIALIZER) {
memset(mPostSpatializerBuffer, 0, mPostSpatializerBufferSize);
}
}
// sink or mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to the sink or mix buffer
// and track effects will accumulate into it
// always clear sink buffer for spatializer output as the output of the spatializer
// effect will be accumulated into it
if ((mBytesRemaining == 0) && (((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
(mixedTracks == 0 && fastTracks > 0)) || (mType == SPATIALIZER))) {
// FIXME as a performance optimization, should remember previous zero status
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
// TODO: In testing, mSinkBuffer below need not be cleared because
// the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
// after mixing.
//
// To enforce this guarantee:
// ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
// (mixedTracks == 0 && fastTracks > 0))
// must imply MIXER_TRACKS_READY.
// Later, we may clear buffers regardless, and skip much of this logic.
}
// FIXME as a performance optimization, should remember previous zero status
memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
}
// if any fast tracks, then status is ready
mMixerStatusIgnoringFastTracks = mixerStatus;
if (fastTracks > 0) {
mixerStatus = MIXER_TRACKS_READY;
}
return mixerStatus;
}
// trackCountForUid_l() must be called with ThreadBase::mutex() held
uint32_t PlaybackThread::trackCountForUid_l(uid_t uid) const
{
uint32_t trackCount = 0;
for (size_t i = 0; i < mTracks.size() ; i++) {
if (mTracks[i]->uid() == uid) {
trackCount++;
}
}
return trackCount;
}
bool PlaybackThread::IsTimestampAdvancing::check(AudioStreamOut* output)
{
// Check the timestamp to see if it's advancing once every 150ms. If we check too frequently, we
// could falsely detect that the frame position has stalled due to underrun because we haven't
// given the Audio HAL enough time to update.
const nsecs_t nowNs = systemTime();
if (nowNs - mPreviousNs < mMinimumTimeBetweenChecksNs) {
return mLatchedValue;
}
mPreviousNs = nowNs;
mLatchedValue = false;
// Determine if the presentation position is still advancing.
uint64_t position = 0;
struct timespec unused;
const status_t ret = output->getPresentationPosition(&position, &unused);
if (ret == NO_ERROR) {
if (position != mPreviousPosition) {
mPreviousPosition = position;
mLatchedValue = true;
}
}
return mLatchedValue;
}
void PlaybackThread::IsTimestampAdvancing::clear()
{
mLatchedValue = true;
mPreviousPosition = 0;
mPreviousNs = 0;
}
// isTrackAllowed_l() must be called with ThreadBase::mutex() held
bool MixerThread::isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
audio_session_t sessionId, uid_t uid) const
{
if (!PlaybackThread::isTrackAllowed_l(channelMask, format, sessionId, uid)) {
return false;
}
// Check validity as we don't call AudioMixer::create() here.
if (!mAudioMixer->isValidFormat(format)) {
ALOGW("%s: invalid format: %#x", __func__, format);
return false;
}
if (!mAudioMixer->isValidChannelMask(channelMask)) {
ALOGW("%s: invalid channelMask: %#x", __func__, channelMask);
return false;
}
return true;
}
// checkForNewParameter_l() must be called with ThreadBase::mutex() held
bool MixerThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
status = NO_ERROR;
AutoPark<FastMixer> park(mFastMixer);
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
if (!isValidPcmSinkFormat(static_cast<audio_format_t>(value))) {
status = BAD_VALUE;
} else {
// no need to save value, since it's constant
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
if (!isValidPcmSinkChannelMask(static_cast<audio_channel_mask_t>(value))) {
status = BAD_VALUE;
} else {
// no need to save value, since it's constant
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
LOG_FATAL("Should not set routing device in MixerThread");
}
if (status == NO_ERROR) {
status = mOutput->stream->setParameters(keyValuePair);
if (!mStandby && status == INVALID_OPERATION) {
ALOGW("%s: setParameters failed with keyValuePair %s, entering standby",
__func__, keyValuePair.c_str());
mOutput->standby();
mThreadMetrics.logEndInterval();
mThreadSnapshot.onEnd();
setStandby_l();
mBytesWritten = 0;
status = mOutput->stream->setParameters(keyValuePair);
}
if (status == NO_ERROR && reconfig) {
readOutputParameters_l();
delete mAudioMixer;
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
for (const auto &track : mTracks) {
const int trackId = track->id();
const status_t createStatus = mAudioMixer->create(
trackId,
track->channelMask(),
track->format(),
track->sessionId());
ALOGW_IF(createStatus != NO_ERROR,
"%s(): AudioMixer cannot create track(%d)"
" mask %#x, format %#x, sessionId %d",
__func__,
trackId, track->channelMask(), track->format(), track->sessionId());
}
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
}
}
return reconfig;
}
void MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Thread throttle time (msecs): %u\n", (uint32_t)mThreadThrottleTimeMs);
dprintf(fd, " AudioMixer tracks: %s\n", mAudioMixer->trackNames().c_str());
dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
dprintf(fd, " Master balance: %f (%s)\n", mMasterBalance.load(),
(hasFastMixer() ? std::to_string(mFastMixer->getMasterBalance())
: mBalance.toString()).c_str());
if (hasFastMixer()) {
dprintf(fd, " FastMixer thread %p tid=%d", mFastMixer.get(), mFastMixer->getTid());
// Make a non-atomic copy of fast mixer dump state so it won't change underneath us
// while we are dumping it. It may be inconsistent, but it won't mutate!
// This is a large object so we place it on the heap.
// FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
const std::unique_ptr<FastMixerDumpState> copy =
std::make_unique<FastMixerDumpState>(mFastMixerDumpState);
copy->dump(fd);
#ifdef STATE_QUEUE_DUMP
// Similar for state queue
StateQueueObserverDump observerCopy = mStateQueueObserverDump;
observerCopy.dump(fd);
StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
mutatorCopy.dump(fd);
#endif
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
// Make a non-atomic copy of audio watchdog dump so it won't change underneath us
AudioWatchdogDump wdCopy = mAudioWatchdogDump;
wdCopy.dump(fd);
}
#endif
} else {
dprintf(fd, " No FastMixer\n");
}
dprintf(fd, "Bluetooth latency modes are %senabled\n",
mBluetoothLatencyModesEnabled ? "" : "not ");
dprintf(fd, "HAL does %ssupport Bluetooth latency modes\n", mOutput != nullptr &&
mOutput->audioHwDev->supportsBluetoothVariableLatency() ? "" : "not ");
dprintf(fd, "Supported latency modes: %s\n", toString(mSupportedLatencyModes).c_str());
}
uint32_t MixerThread::idleSleepTimeUs() const
{
return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
}
uint32_t MixerThread::suspendSleepTimeUs() const
{
return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
}
void MixerThread::cacheParameters_l()
{
PlaybackThread::cacheParameters_l();
// FIXME: Relaxed timing because of a certain device that can't meet latency
// Should be reduced to 2x after the vendor fixes the driver issue
// increase threshold again due to low power audio mode. The way this warning
// threshold is calculated and its usefulness should be reconsidered anyway.
maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
}
void MixerThread::onHalLatencyModesChanged_l() {
mAfThreadCallback->onSupportedLatencyModesChanged(mId, mSupportedLatencyModes);
}
void MixerThread::setHalLatencyMode_l() {
// Only handle latency mode if:
// - mBluetoothLatencyModesEnabled is true
// - the HAL supports latency modes
// - the selected device is Bluetooth LE or A2DP
if (!mBluetoothLatencyModesEnabled.load() || mSupportedLatencyModes.empty()) {
return;
}
if (mOutDeviceTypeAddrs.size() != 1
|| !(audio_is_a2dp_out_device(mOutDeviceTypeAddrs[0].mType)
|| audio_is_ble_out_device(mOutDeviceTypeAddrs[0].mType))) {
return;
}
audio_latency_mode_t latencyMode = AUDIO_LATENCY_MODE_FREE;
if (mSupportedLatencyModes.size() == 1) {
// If the HAL only support one latency mode currently, confirm the choice
latencyMode = mSupportedLatencyModes[0];
} else if (mSupportedLatencyModes.size() > 1) {
// Request low latency if:
// - At least one active track is either:
// - a fast track with gaming usage or
// - a track with acessibility usage
for (const auto& track : mActiveTracks) {
if ((track->isFastTrack() && track->attributes().usage == AUDIO_USAGE_GAME)
|| track->attributes().usage == AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY) {
latencyMode = AUDIO_LATENCY_MODE_LOW;
break;
}
}
}
if (latencyMode != mSetLatencyMode) {
status_t status = mOutput->stream->setLatencyMode(latencyMode);
ALOGD("%s: thread(%d) setLatencyMode(%s) returned %d",
__func__, mId, toString(latencyMode).c_str(), status);
if (status == NO_ERROR) {
mSetLatencyMode = latencyMode;
}
}
}
void MixerThread::updateHalSupportedLatencyModes_l() {
if (mOutput == nullptr || mOutput->stream == nullptr) {
return;
}
std::vector<audio_latency_mode_t> latencyModes;
const status_t status = mOutput->stream->getRecommendedLatencyModes(&latencyModes);
if (status != NO_ERROR) {
latencyModes.clear();
}
if (latencyModes != mSupportedLatencyModes) {
ALOGD("%s: thread(%d) status %d supported latency modes: %s",
__func__, mId, status, toString(latencyModes).c_str());
mSupportedLatencyModes.swap(latencyModes);
sendHalLatencyModesChangedEvent_l();
}
}
status_t MixerThread::getSupportedLatencyModes(
std::vector<audio_latency_mode_t>* modes) {
if (modes == nullptr) {
return BAD_VALUE;
}
audio_utils::lock_guard _l(mutex());
*modes = mSupportedLatencyModes;
return NO_ERROR;
}
void MixerThread::onRecommendedLatencyModeChanged(
std::vector<audio_latency_mode_t> modes) {
audio_utils::lock_guard _l(mutex());
if (modes != mSupportedLatencyModes) {
ALOGD("%s: thread(%d) supported latency modes: %s",
__func__, mId, toString(modes).c_str());
mSupportedLatencyModes.swap(modes);
sendHalLatencyModesChangedEvent_l();
}
}
status_t MixerThread::setBluetoothVariableLatencyEnabled(bool enabled) {
if (mOutput == nullptr || mOutput->audioHwDev == nullptr
|| !mOutput->audioHwDev->supportsBluetoothVariableLatency()) {
return INVALID_OPERATION;
}
mBluetoothLatencyModesEnabled.store(enabled);
return NO_ERROR;
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfPlaybackThread> IAfPlaybackThread::createDirectOutputThread(
const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo) {
return sp<DirectOutputThread>::make(
afThreadCallback, output, id, systemReady, offloadInfo);
}
DirectOutputThread::DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output, audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
const audio_offload_info_t& offloadInfo)
: PlaybackThread(afThreadCallback, output, id, type, systemReady)
, mOffloadInfo(offloadInfo)
{
setMasterBalance(afThreadCallback->getMasterBalance_l());
}
DirectOutputThread::~DirectOutputThread()
{
}
void DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Master balance: %f Left: %f Right: %f\n",
mMasterBalance.load(), mMasterBalanceLeft, mMasterBalanceRight);
}
void DirectOutputThread::setMasterBalance(float balance)
{
audio_utils::lock_guard _l(mutex());
if (mMasterBalance != balance) {
mMasterBalance.store(balance);
mBalance.computeStereoBalance(balance, &mMasterBalanceLeft, &mMasterBalanceRight);
broadcast_l();
}
}
void DirectOutputThread::processVolume_l(IAfTrack* track, bool lastTrack)
{
float left, right;
// Ensure volumeshaper state always advances even when muted.
const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
const int64_t frames = mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
const int64_t time = mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
ALOGVV("%s: Direct/Offload bufferConsumed:%zu timestamp frames:%lld time:%lld",
__func__, proxy->framesReleased(), (long long)frames, (long long)time);
const int64_t volumeShaperFrames =
mMonotonicFrameCounter.updateAndGetMonotonicFrameCount(frames, time);
const auto [shaperVolume, shaperActive] =
track->getVolumeHandler()->getVolume(volumeShaperFrames);
mVolumeShaperActive = shaperActive;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
left = float_from_gain(gain_minifloat_unpack_left(vlr));
right = float_from_gain(gain_minifloat_unpack_right(vlr));
const bool clientVolumeMute = (left == 0.f && right == 0.f);
if (mMasterMute || mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
left = right = 0;
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
const float v = mMasterVolume * typeVolume * shaperVolume;
if (left > GAIN_FLOAT_UNITY) {
left = GAIN_FLOAT_UNITY;
}
if (right > GAIN_FLOAT_UNITY) {
right = GAIN_FLOAT_UNITY;
}
left *= v;
right *= v;
if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
|| audio_channel_count_from_out_mask(mChannelMask) > 1) {
left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
right *= mMasterBalanceRight;
}
}
track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
/*muteState=*/{mMasterMute,
mStreamTypes[track->streamType()].volume == 0.f,
mStreamTypes[track->streamType()].mute,
track->isPlaybackRestricted(),
clientVolumeMute,
shaperVolume == 0.f});
if (lastTrack) {
track->setFinalVolume(left, right);
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!mEffectChains.isEmpty()) {
// if effect chain exists, volume is handled by it.
// Convert volumes from float to 8.24
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
// Direct/Offload effect chains set output volume in setVolume().
(void)mEffectChains[0]->setVolume(&vl, &vr);
} else {
// otherwise we directly set the volume.
setVolumeForOutput_l(left, right);
}
}
}
}
void DirectOutputThread::onAddNewTrack_l()
{
sp<IAfTrack> previousTrack = mPreviousTrack.promote();
sp<IAfTrack> latestTrack = mActiveTracks.getLatest();
if (previousTrack != 0 && latestTrack != 0) {
if (mType == DIRECT) {
if (previousTrack.get() != latestTrack.get()) {
mFlushPending = true;
}
} else /* mType == OFFLOAD */ {
if (previousTrack->sessionId() != latestTrack->sessionId() ||
previousTrack->isFlushPending()) {
mFlushPending = true;
}
}
} else if (previousTrack == 0) {
// there could be an old track added back during track transition for direct
// output, so always issues flush to flush data of the previous track if it
// was already destroyed with HAL paused, then flush can resume the playback
mFlushPending = true;
}
PlaybackThread::onAddNewTrack_l();
}
PlaybackThread::mixer_state DirectOutputThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove
)
{
size_t count = mActiveTracks.size();
mixer_state mixerStatus = MIXER_IDLE;
bool doHwPause = false;
bool doHwResume = false;
// find out which tracks need to be processed
for (const sp<IAfTrack>& t : mActiveTracks) {
if (t->isInvalid()) {
ALOGW("An invalidated track shouldn't be in active list");
tracksToRemove->add(t);
continue;
}
IAfTrack* const track = t.get();
#ifdef VERY_VERY_VERBOSE_LOGGING
audio_track_cblk_t* cblk = track->cblk();
#endif
// Only consider last track started for volume and mixer state control.
// In theory an older track could underrun and restart after the new one starts
// but as we only care about the transition phase between two tracks on a
// direct output, it is not a problem to ignore the underrun case.
sp<IAfTrack> l = mActiveTracks.getLatest();
bool last = l.get() == track;
if (track->isPausePending()) {
track->pauseAck();
// It is possible a track might have been flushed or stopped.
// Other operations such as flush pending might occur on the next prepare.
if (track->isPausing()) {
track->setPaused();
}
// Always perform pause, as an immediate flush will change
// the pause state to be no longer isPausing().
if (mHwSupportsPause && last && !mHwPaused) {
doHwPause = true;
mHwPaused = true;
}
} else if (track->isFlushPending()) {
track->flushAck();
if (last) {
mFlushPending = true;
}
} else if (track->isResumePending()) {
track->resumeAck();
if (last) {
mLeftVolFloat = mRightVolFloat = -1.0;
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
}
// The first time a track is added we wait
// for all its buffers to be filled before processing it.
// Allow draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
// hence the test on (track->retryCount() > 1).
// If track->retryCount() <= 1 then track is about to be disabled, paused, removed,
// so we accept any nonzero amount of data delivered by the AudioTrack (which will
// reset the retry counter).
// Do not use a high threshold for compressed audio.
// target retry count that we will use is based on the time we wait for retries.
const int32_t targetRetryCount = kMaxTrackRetriesDirectMs * 1000 / mActiveSleepTimeUs;
// the retry threshold is when we accept any size for PCM data. This is slightly
// smaller than the retry count so we can push small bits of data without a glitch.
const int32_t retryThreshold = targetRetryCount > 2 ? targetRetryCount - 1 : 1;
uint32_t minFrames;
if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
&& (track->retryCount() > retryThreshold) && audio_has_proportional_frames(mFormat)) {
minFrames = mNormalFrameCount;
} else {
minFrames = 1;
}
const size_t framesReady = track->framesReady();
const int trackId = track->id();
if (ATRACE_ENABLED()) {
std::string traceName("nRdy");
traceName += std::to_string(trackId);
ATRACE_INT(traceName.c_str(), framesReady);
}
if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() &&
!track->isStopping_2() && !track->isStopped())
{
ALOGVV("track(%d) s=%08x [OK]", trackId, cblk->mServer);
if (track->fillingStatus() == IAfTrack::FS_FILLED) {
track->fillingStatus() = IAfTrack::FS_ACTIVE;
if (last) {
// make sure processVolume_l() will apply new volume even if 0
mLeftVolFloat = mRightVolFloat = -1.0;
}
if (!mHwSupportsPause) {
track->resumeAck();
}
}
// compute volume for this track
processVolume_l(track, last);
if (last) {
sp<IAfTrack> previousTrack = mPreviousTrack.promote();
if (previousTrack != 0) {
if (track != previousTrack.get()) {
// Flush any data still being written from last track
mBytesRemaining = 0;
// Invalidate previous track to force a seek when resuming.
previousTrack->invalidate();
}
}
mPreviousTrack = track;
// reset retry count
track->retryCount() = targetRetryCount;
mActiveTrack = t;
mixerStatus = MIXER_TRACKS_READY;
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
} else {
// clear effect chain input buffer if the last active track started underruns
// to avoid sending previous audio buffer again to effects
if (!mEffectChains.isEmpty() && last) {
mEffectChains[0]->clearInputBuffer();
}
if (track->isStopping_1()) {
track->setState(IAfTrackBase::STOPPING_2);
if (last && mHwPaused) {
doHwResume = true;
mHwPaused = false;
}
}
if ((track->sharedBuffer() != 0) || track->isStopped() ||
track->isStopping_2() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
bool presComplete = false;
if (mStandby || !last ||
(presComplete = track->presentationComplete(latency_l())) ||
track->isPaused() || mHwPaused) {
if (presComplete) {
mOutput->presentationComplete();
}
if (track->isStopping_2()) {
track->setState(IAfTrackBase::STOPPED);
}
if (track->isStopped()) {
track->reset();
}
tracksToRemove->add(track);
}
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
// Only consider last track started for mixer state control
bool isTimestampAdvancing = mIsTimestampAdvancing.check(mOutput);
if (!isTunerStream() // tuner streams remain active in underrun
&& --(track->retryCount()) <= 0) {
if (isTimestampAdvancing) { // HAL is still playing audio, give us more time.
track->retryCount() = kMaxTrackRetriesOffload;
} else {
ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", trackId);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of
// underrun; it will then automatically call start() when data is available
track->disable();
// only do hw pause when track is going to be removed due to BUFFER TIMEOUT.
// unlike mixerthread, HAL can be paused for direct output
ALOGW("pause because of UNDERRUN, framesReady = %zu,"
"minFrames = %u, mFormat = %#x",
framesReady, minFrames, mFormat);
if (last && mHwSupportsPause && !mHwPaused && !mStandby) {
doHwPause = true;
mHwPaused = true;
}
}
} else if (last) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
}
// if an active track did not command a flush, check for pending flush on stopped tracks
if (!mFlushPending) {
for (size_t i = 0; i < mTracks.size(); i++) {
if (mTracks[i]->isFlushPending()) {
mTracks[i]->flushAck();
mFlushPending = true;
}
}
}
// make sure the pause/flush/resume sequence is executed in the right order.
// If a flush is pending and a track is active but the HW is not paused, force a HW pause
// before flush and then resume HW. This can happen in case of pause/flush/resume
// if resume is received before pause is executed.
if (mHwSupportsPause && !mStandby &&
(doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
status_t result = mOutput->stream->pause();
ALOGE_IF(result != OK, "Error when pausing output stream: %d", result);
doHwResume = !doHwPause; // resume if pause is due to flush.
}
if (mFlushPending) {
flushHw_l();
}
if (mHwSupportsPause && !mStandby && doHwResume) {
status_t result = mOutput->stream->resume();
ALOGE_IF(result != OK, "Error when resuming output stream: %d", result);
}
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
return mixerStatus;
}
void DirectOutputThread::threadLoop_mix()
{
size_t frameCount = mFrameCount;
int8_t *curBuf = (int8_t *)mSinkBuffer;
// output audio to hardware
while (frameCount) {
AudioBufferProvider::Buffer buffer;
buffer.frameCount = frameCount;
status_t status = mActiveTrack->getNextBuffer(&buffer);
if (status != NO_ERROR || buffer.raw == NULL) {
// no need to pad with 0 for compressed audio
if (audio_has_proportional_frames(mFormat)) {
memset(curBuf, 0, frameCount * mFrameSize);
}
break;
}
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
frameCount -= buffer.frameCount;
curBuf += buffer.frameCount * mFrameSize;
mActiveTrack->releaseBuffer(&buffer);
}
mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
mSleepTimeUs = 0;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mActiveTrack.clear();
}
void DirectOutputThread::threadLoop_sleepTime()
{
// do not write to HAL when paused
if (mHwPaused || (usesHwAvSync() && mStandby)) {
mSleepTimeUs = mIdleSleepTimeUs;
return;
}
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
mSleepTimeUs = mActiveSleepTimeUs;
} else {
mSleepTimeUs = mIdleSleepTimeUs;
}
// Note: In S or later, we do not write zeroes for
// linear or proportional PCM direct tracks in underrun.
}
void DirectOutputThread::threadLoop_exit()
{
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); i++) {
if (mTracks[i]->isFlushPending()) {
mTracks[i]->flushAck();
mFlushPending = true;
}
}
if (mFlushPending) {
flushHw_l();
}
}
PlaybackThread::threadLoop_exit();
}
// must be called with thread mutex locked
bool DirectOutputThread::shouldStandby_l()
{
bool trackPaused = false;
bool trackStopped = false;
// do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
// after a timeout and we will enter standby then.
if (mTracks.size() > 0) {
trackPaused = mTracks[mTracks.size() - 1]->isPaused();
trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
mTracks[mTracks.size() - 1]->state() == IAfTrackBase::IDLE;
}
return !mStandby && !(trackPaused || (mHwPaused && !trackStopped));
}
// checkForNewParameter_l() must be called with ThreadBase::mutex() held
bool DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
status = NO_ERROR;
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
LOG_FATAL("Should not set routing device in DirectOutputThread");
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (status == NO_ERROR) {
status = mOutput->stream->setParameters(keyValuePair);
if (!mStandby && status == INVALID_OPERATION) {
mOutput->standby();
if (!mStandby) {
mThreadMetrics.logEndInterval();
mThreadSnapshot.onEnd();
setStandby_l();
}
mBytesWritten = 0;
status = mOutput->stream->setParameters(keyValuePair);
}
if (status == NO_ERROR && reconfig) {
readOutputParameters_l();
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
}
}
return reconfig;
}
uint32_t DirectOutputThread::activeSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
time = PlaybackThread::activeSleepTimeUs();
} else {
time = kDirectMinSleepTimeUs;
}
return time;
}
uint32_t DirectOutputThread::idleSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
} else {
time = kDirectMinSleepTimeUs;
}
return time;
}
uint32_t DirectOutputThread::suspendSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
time = kDirectMinSleepTimeUs;
}
return time;
}
void DirectOutputThread::cacheParameters_l()
{
PlaybackThread::cacheParameters_l();
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
// no delay on outputs with HW A/V sync
if (usesHwAvSync()) {
mStandbyDelayNs = 0;
} else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) {
mStandbyDelayNs = kOffloadStandbyDelayNs;
} else {
mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2);
}
}
void DirectOutputThread::flushHw_l()
{
PlaybackThread::flushHw_l();
mOutput->flush();
mHwPaused = false;
mFlushPending = false;
mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
mTimestamp.clear();
mMonotonicFrameCounter.onFlush();
}
int64_t DirectOutputThread::computeWaitTimeNs_l() const {
// If a VolumeShaper is active, we must wake up periodically to update volume.
const int64_t NS_PER_MS = 1000000;
return mVolumeShaperActive ?
kMinNormalSinkBufferSizeMs * NS_PER_MS : PlaybackThread::computeWaitTimeNs_l();
}
// ----------------------------------------------------------------------------
AsyncCallbackThread::AsyncCallbackThread(
const wp<PlaybackThread>& playbackThread)
: Thread(false /*canCallJava*/),
mPlaybackThread(playbackThread),
mWriteAckSequence(0),
mDrainSequence(0),
mAsyncError(false)
{
}
void AsyncCallbackThread::onFirstRef()
{
run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
}
bool AsyncCallbackThread::threadLoop()
{
while (!exitPending()) {
uint32_t writeAckSequence;
uint32_t drainSequence;
bool asyncError;
{
audio_utils::unique_lock _l(mutex());
while (!((mWriteAckSequence & 1) ||
(mDrainSequence & 1) ||
mAsyncError ||
exitPending())) {
mWaitWorkCV.wait(_l);
}
if (exitPending()) {
break;
}
ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
mWriteAckSequence, mDrainSequence);
writeAckSequence = mWriteAckSequence;
mWriteAckSequence &= ~1;
drainSequence = mDrainSequence;
mDrainSequence &= ~1;
asyncError = mAsyncError;
mAsyncError = false;
}
{
const sp<PlaybackThread> playbackThread = mPlaybackThread.promote();
if (playbackThread != 0) {
if (writeAckSequence & 1) {
playbackThread->resetWriteBlocked(writeAckSequence >> 1);
}
if (drainSequence & 1) {
playbackThread->resetDraining(drainSequence >> 1);
}
if (asyncError) {
playbackThread->onAsyncError();
}
}
}
}
return false;
}
void AsyncCallbackThread::exit()
{
ALOGV("AsyncCallbackThread::exit");
audio_utils::lock_guard _l(mutex());
requestExit();
mWaitWorkCV.notify_all();
}
void AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
{
audio_utils::lock_guard _l(mutex());
// bit 0 is cleared
mWriteAckSequence = sequence << 1;
}
void AsyncCallbackThread::resetWriteBlocked()
{
audio_utils::lock_guard _l(mutex());
// ignore unexpected callbacks
if (mWriteAckSequence & 2) {
mWriteAckSequence |= 1;
mWaitWorkCV.notify_one();
}
}
void AsyncCallbackThread::setDraining(uint32_t sequence)
{
audio_utils::lock_guard _l(mutex());
// bit 0 is cleared
mDrainSequence = sequence << 1;
}
void AsyncCallbackThread::resetDraining()
{
audio_utils::lock_guard _l(mutex());
// ignore unexpected callbacks
if (mDrainSequence & 2) {
mDrainSequence |= 1;
mWaitWorkCV.notify_one();
}
}
void AsyncCallbackThread::setAsyncError()
{
audio_utils::lock_guard _l(mutex());
mAsyncError = true;
mWaitWorkCV.notify_one();
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfPlaybackThread> IAfPlaybackThread::createOffloadThread(
const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo) {
return sp<OffloadThread>::make(afThreadCallback, output, id, systemReady, offloadInfo);
}
OffloadThread::OffloadThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo)
: DirectOutputThread(afThreadCallback, output, id, OFFLOAD, systemReady, offloadInfo),
mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true)
{
//FIXME: mStandby should be set to true by ThreadBase constructo
mStandby = true;
mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
}
void OffloadThread::threadLoop_exit()
{
if (mFlushPending || mHwPaused) {
// If a flush is pending or track was paused, just discard buffered data
audio_utils::lock_guard l(mutex());
flushHw_l();
} else {
mMixerStatus = MIXER_DRAIN_ALL;
threadLoop_drain();
}
if (mUseAsyncWrite) {
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->exit();
}
PlaybackThread::threadLoop_exit();
}
PlaybackThread::mixer_state OffloadThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove
)
{
size_t count = mActiveTracks.size();
mixer_state mixerStatus = MIXER_IDLE;
bool doHwPause = false;
bool doHwResume = false;
ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
// find out which tracks need to be processed
for (const sp<IAfTrack>& t : mActiveTracks) {
IAfTrack* const track = t.get();
#ifdef VERY_VERY_VERBOSE_LOGGING
audio_track_cblk_t* cblk = track->cblk();
#endif
// Only consider last track started for volume and mixer state control.
// In theory an older track could underrun and restart after the new one starts
// but as we only care about the transition phase between two tracks on a
// direct output, it is not a problem to ignore the underrun case.
sp<IAfTrack> l = mActiveTracks.getLatest();
bool last = l.get() == track;
if (track->isInvalid()) {
ALOGW("An invalidated track shouldn't be in active list");
tracksToRemove->add(track);
continue;
}
if (track->state() == IAfTrackBase::IDLE) {
ALOGW("An idle track shouldn't be in active list");
continue;
}
if (track->isPausePending()) {
track->pauseAck();
// It is possible a track might have been flushed or stopped.
// Other operations such as flush pending might occur on the next prepare.
if (track->isPausing()) {
track->setPaused();
}
// Always perform pause if last, as an immediate flush will change
// the pause state to be no longer isPausing().
if (last) {
if (mHwSupportsPause && !mHwPaused) {
doHwPause = true;
mHwPaused = true;
}
// If we were part way through writing the mixbuffer to
// the HAL we must save this until we resume
// BUG - this will be wrong if a different track is made active,
// in that case we want to discard the pending data in the
// mixbuffer and tell the client to present it again when the
// track is resumed
mPausedWriteLength = mCurrentWriteLength;
mPausedBytesRemaining = mBytesRemaining;
mBytesRemaining = 0; // stop writing
}
tracksToRemove->add(track);
} else if (track->isFlushPending()) {
if (track->isStopping_1()) {
track->retryCount() = kMaxTrackStopRetriesOffload;
} else {
track->retryCount() = kMaxTrackRetriesOffload;
}
track->flushAck();
if (last) {
mFlushPending = true;
}
} else if (track->isResumePending()){
track->resumeAck();
if (last) {
if (mPausedBytesRemaining) {
// Need to continue write that was interrupted
mCurrentWriteLength = mPausedWriteLength;
mBytesRemaining = mPausedBytesRemaining;
mPausedBytesRemaining = 0;
}
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
// threadLoop_mix() will handle the case that we need to
// resume an interrupted write
}
// enable write to audio HAL
mSleepTimeUs = 0;
mLeftVolFloat = mRightVolFloat = -1.0;
// Do not handle new data in this iteration even if track->framesReady()
mixerStatus = MIXER_TRACKS_ENABLED;
}
} else if (track->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
ALOGVV("OffloadThread: track(%d) s=%08x [OK]", track->id(), cblk->mServer);
if (track->fillingStatus() == IAfTrack::FS_FILLED) {
track->fillingStatus() = IAfTrack::FS_ACTIVE;
if (last) {
// make sure processVolume_l() will apply new volume even if 0
mLeftVolFloat = mRightVolFloat = -1.0;
}
}
if (last) {
sp<IAfTrack> previousTrack = mPreviousTrack.promote();
if (previousTrack != 0) {
if (track != previousTrack.get()) {
// Flush any data still being written from last track
mBytesRemaining = 0;
if (mPausedBytesRemaining) {
// Last track was paused so we also need to flush saved
// mixbuffer state and invalidate track so that it will
// re-submit that unwritten data when it is next resumed
mPausedBytesRemaining = 0;
// Invalidate is a bit drastic - would be more efficient
// to have a flag to tell client that some of the
// previously written data was lost
previousTrack->invalidate();
}
// flush data already sent to the DSP if changing audio session as audio
// comes from a different source. Also invalidate previous track to force a
// seek when resuming.
if (previousTrack->sessionId() != track->sessionId()) {
previousTrack->invalidate();
}
}
}
mPreviousTrack = track;
// reset retry count
if (track->isStopping_1()) {
track->retryCount() = kMaxTrackStopRetriesOffload;
} else {
track->retryCount() = kMaxTrackRetriesOffload;
}
mActiveTrack = t;
mixerStatus = MIXER_TRACKS_READY;
}
} else {
ALOGVV("OffloadThread: track(%d) s=%08x [NOT READY]", track->id(), cblk->mServer);
if (track->isStopping_1()) {
if (--(track->retryCount()) <= 0) {
// Hardware buffer can hold a large amount of audio so we must
// wait for all current track's data to drain before we say
// that the track is stopped.
if (mBytesRemaining == 0) {
// Only start draining when all data in mixbuffer
// has been written
ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
track->setState(IAfTrackBase::STOPPING_2);
// so presentation completes after
// drain do not drain if no data was ever sent to HAL (mStandby == true)
if (last && !mStandby) {
// do not modify drain sequence if we are already draining. This happens
// when resuming from pause after drain.
if ((mDrainSequence & 1) == 0) {
mSleepTimeUs = 0;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
mixerStatus = MIXER_DRAIN_TRACK;
mDrainSequence += 2;
}
if (mHwPaused) {
// It is possible to move from PAUSED to STOPPING_1 without
// a resume so we must ensure hardware is running
doHwResume = true;
mHwPaused = false;
}
}
}
} else if (last) {
ALOGV("stopping1 underrun retries left %d", track->retryCount());
mixerStatus = MIXER_TRACKS_ENABLED;
}
} else if (track->isStopping_2()) {
// Drain has completed or we are in standby, signal presentation complete
if (!(mDrainSequence & 1) || !last || mStandby) {
track->setState(IAfTrackBase::STOPPED);
mOutput->presentationComplete();
track->presentationComplete(latency_l()); // always returns true
track->reset();
tracksToRemove->add(track);
// OFFLOADED stop resets frame counts.
if (!mUseAsyncWrite) {
// If we don't get explicit drain notification we must
// register discontinuity regardless of whether this is
// the previous (!last) or the upcoming (last) track
// to avoid skipping the discontinuity.
mTimestampVerifier.discontinuity(
mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
}
}
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
bool isTimestampAdvancing = mIsTimestampAdvancing.check(mOutput);
if (!isTunerStream() // tuner streams remain active in underrun
&& --(track->retryCount()) <= 0) {
if (isTimestampAdvancing) { // HAL is still playing audio, give us more time.
track->retryCount() = kMaxTrackRetriesOffload;
} else {
ALOGV("OffloadThread: BUFFER TIMEOUT: remove track(%d) from active list",
track->id());
tracksToRemove->add(track);
// tell client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
track->disable();
}
} else if (last){
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
// compute volume for this track
if (track->isReady()) { // check ready to prevent premature start.
processVolume_l(track, last);
}
}
// make sure the pause/flush/resume sequence is executed in the right order.
// If a flush is pending and a track is active but the HW is not paused, force a HW pause
// before flush and then resume HW. This can happen in case of pause/flush/resume
// if resume is received before pause is executed.
if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
status_t result = mOutput->stream->pause();
ALOGE_IF(result != OK, "Error when pausing output stream: %d", result);
doHwResume = !doHwPause; // resume if pause is due to flush.
}
if (mFlushPending) {
flushHw_l();
}
if (!mStandby && doHwResume) {
status_t result = mOutput->stream->resume();
ALOGE_IF(result != OK, "Error when resuming output stream: %d", result);
}
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
return mixerStatus;
}
// must be called with thread mutex locked
bool OffloadThread::waitingAsyncCallback_l()
{
ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
mWriteAckSequence, mDrainSequence);
if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
return true;
}
return false;
}
bool OffloadThread::waitingAsyncCallback()
{
audio_utils::lock_guard _l(mutex());
return waitingAsyncCallback_l();
}
void OffloadThread::flushHw_l()
{
DirectOutputThread::flushHw_l();
// Flush anything still waiting in the mixbuffer
mCurrentWriteLength = 0;
mBytesRemaining = 0;
mPausedWriteLength = 0;
mPausedBytesRemaining = 0;
// reset bytes written count to reflect that DSP buffers are empty after flush.
mBytesWritten = 0;
if (mUseAsyncWrite) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
mDrainSequence = (mDrainSequence + 2) & ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
mCallbackThread->setDraining(mDrainSequence);
}
}
void OffloadThread::invalidateTracks(audio_stream_type_t streamType)
{
audio_utils::lock_guard _l(mutex());
if (PlaybackThread::invalidateTracks_l(streamType)) {
mFlushPending = true;
}
}
void OffloadThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
audio_utils::lock_guard _l(mutex());
if (PlaybackThread::invalidateTracks_l(portIds)) {
mFlushPending = true;
}
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfDuplicatingThread> IAfDuplicatingThread::create(
const sp<IAfThreadCallback>& afThreadCallback,
IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady) {
return sp<DuplicatingThread>::make(afThreadCallback, mainThread, id, systemReady);
}
DuplicatingThread::DuplicatingThread(const sp<IAfThreadCallback>& afThreadCallback,
IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady)
: MixerThread(afThreadCallback, mainThread->getOutput(), id,
systemReady, DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
addOutputTrack(mainThread);
}
DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
}
void DuplicatingThread::threadLoop_mix()
{
// mix buffers...
if (outputsReady()) {
mAudioMixer->process();
} else {
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
} else {
memset(mSinkBuffer, 0, mSinkBufferSize);
}
}
mSleepTimeUs = 0;
writeFrames = mNormalFrameCount;
mCurrentWriteLength = mSinkBufferSize;
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
}
void DuplicatingThread::threadLoop_sleepTime()
{
if (mSleepTimeUs == 0) {
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
mSleepTimeUs = mActiveSleepTimeUs;
} else {
mSleepTimeUs = mIdleSleepTimeUs;
}
} else if (mBytesWritten != 0) {
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
writeFrames = mNormalFrameCount;
memset(mSinkBuffer, 0, mSinkBufferSize);
} else {
// flush remaining overflow buffers in output tracks
writeFrames = 0;
}
mSleepTimeUs = 0;
}
}
ssize_t DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
const ssize_t actualWritten = outputTracks[i]->write(mSinkBuffer, writeFrames);
// Consider the first OutputTrack for timestamp and frame counting.
// The threadLoop() generally assumes writing a full sink buffer size at a time.
// Here, we correct for writeFrames of 0 (a stop) or underruns because
// we always claim success.
if (i == 0) {
const ssize_t correction = mSinkBufferSize / mFrameSize - actualWritten;
ALOGD_IF(correction != 0 && writeFrames != 0,
"%s: writeFrames:%u actualWritten:%zd correction:%zd mFramesWritten:%lld",
__func__, writeFrames, actualWritten, correction, (long long)mFramesWritten);
mFramesWritten -= correction;
}
// TODO: Report correction for the other output tracks and show in the dump.
}
if (mStandby) {
mThreadMetrics.logBeginInterval();
mThreadSnapshot.onBegin();
mStandby = false;
}
return (ssize_t)mSinkBufferSize;
}
void DuplicatingThread::threadLoop_standby()
{
// DuplicatingThread implements standby by stopping all tracks
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->stop();
}
}
void DuplicatingThread::threadLoop_exit()
{
// Prevent calling the OutputTrack dtor in the DuplicatingThread dtor
// where other mutexes (i.e. AudioPolicyService_Mutex) may be held.
// Do so here in the threadLoop_exit().
SortedVector <sp<IAfOutputTrack>> localTracks;
{
audio_utils::lock_guard l(mutex());
localTracks = std::move(mOutputTracks);
mOutputTracks.clear();
}
localTracks.clear();
outputTracks.clear();
PlaybackThread::threadLoop_exit();
}
void DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MixerThread::dumpInternals_l(fd, args);
std::stringstream ss;
const size_t numTracks = mOutputTracks.size();
ss << " " << numTracks << " OutputTracks";
if (numTracks > 0) {
ss << ":";
for (const auto &track : mOutputTracks) {
const auto thread = track->thread().promote();
ss << " (" << track->id() << " : ";
if (thread.get() != nullptr) {
ss << thread.get() << ", " << thread->id();
} else {
ss << "null";
}
ss << ")";
}
}
ss << "\n";
std::string result = ss.str();
write(fd, result.c_str(), result.size());
}
void DuplicatingThread::saveOutputTracks()
{
outputTracks = mOutputTracks;
}
void DuplicatingThread::clearOutputTracks()
{
outputTracks.clear();
}
void DuplicatingThread::addOutputTrack(IAfPlaybackThread* thread)
{
audio_utils::lock_guard _l(mutex());
// The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
// Adjust for thread->sampleRate() to determine minimum buffer frame count.
// Then triple buffer because Threads do not run synchronously and may not be clock locked.
const size_t frameCount =
3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
// TODO: Consider asynchronous sample rate conversion to handle clock disparity
// from different OutputTracks and their associated MixerThreads (e.g. one may
// nearly empty and the other may be dropping data).
// TODO b/182392769: use attribution source util, move to server edge
AttributionSourceState attributionSource = AttributionSourceState();
attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
IPCThreadState::self()->getCallingUid()));
attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(
IPCThreadState::self()->getCallingPid()));
attributionSource.token = sp<BBinder>::make();
sp<IAfOutputTrack> outputTrack = IAfOutputTrack::create(thread,
this,
mSampleRate,
mFormat,
mChannelMask,
frameCount,
attributionSource);
status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY;
if (status != NO_ERROR) {
ALOGE("addOutputTrack() initCheck failed %d", status);
return;
}
thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
mOutputTracks.add(outputTrack);
ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
}
void DuplicatingThread::removeOutputTrack(IAfPlaybackThread* thread)
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mOutputTracks.size(); i++) {
if (mOutputTracks[i]->thread() == thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
updateWaitTime_l();
// NO_THREAD_SAFETY_ANALYSIS
// Lambda workaround: as thread != this
// we can safely call the remote thread getOutput.
const bool equalOutput =
[&](){ return thread->getOutput() == mOutput; }();
if (equalOutput) {
mOutput = nullptr;
}
return;
}
}
ALOGV("removeOutputTrack(): unknown thread: %p", thread);
}
// caller must hold mutex()
void DuplicatingThread::updateWaitTime_l()
{
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
const auto strong = mOutputTracks[i]->thread().promote();
if (strong != 0) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
}
}
}
}
bool DuplicatingThread::outputsReady()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
const auto thread = outputTracks[i]->thread().promote();
if (thread == 0) {
ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
outputTracks[i].get());
return false;
}
IAfPlaybackThread* const playbackThread = thread->asIAfPlaybackThread().get();
// see note at standby() declaration
if (playbackThread->inStandby() && !playbackThread->isSuspended()) {
ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
thread.get());
return false;
}
}
return true;
}
void DuplicatingThread::sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata)
{
for (auto& outputTrack : outputTracks) { // not mOutputTracks
outputTrack->setMetadatas(metadata.tracks);
}
}
uint32_t DuplicatingThread::activeSleepTimeUs() const
{
// return half the wait time in microseconds.
return std::min(mWaitTimeMs * 500ULL, (unsigned long long)UINT32_MAX); // prevent overflow.
}
void DuplicatingThread::cacheParameters_l()
{
// updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
updateWaitTime_l();
MixerThread::cacheParameters_l();
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfPlaybackThread> IAfPlaybackThread::createSpatializerThread(
const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
audio_config_base_t* mixerConfig) {
return sp<SpatializerThread>::make(afThreadCallback, output, id, systemReady, mixerConfig);
}
SpatializerThread::SpatializerThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
audio_config_base_t *mixerConfig)
: MixerThread(afThreadCallback, output, id, systemReady, SPATIALIZER, mixerConfig)
{
}
void SpatializerThread::setHalLatencyMode_l() {
// if mSupportedLatencyModes is empty, the HAL stream does not support
// latency mode control and we can exit.
if (mSupportedLatencyModes.empty()) {
return;
}
audio_latency_mode_t latencyMode = AUDIO_LATENCY_MODE_FREE;
if (mSupportedLatencyModes.size() == 1) {
// If the HAL only support one latency mode currently, confirm the choice
latencyMode = mSupportedLatencyModes[0];
} else if (mSupportedLatencyModes.size() > 1) {
// Request low latency if:
// - The low latency mode is requested by the spatializer controller
// (mRequestedLatencyMode = AUDIO_LATENCY_MODE_LOW)
// AND
// - At least one active track is spatialized
for (const auto& track : mActiveTracks) {
if (track->isSpatialized()) {
latencyMode = mRequestedLatencyMode;
break;
}
}
}
if (latencyMode != mSetLatencyMode) {
status_t status = mOutput->stream->setLatencyMode(latencyMode);
ALOGD("%s: thread(%d) setLatencyMode(%s) returned %d",
__func__, mId, toString(latencyMode).c_str(), status);
if (status == NO_ERROR) {
mSetLatencyMode = latencyMode;
}
}
}
status_t SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
if (mode < 0 || mode >= AUDIO_LATENCY_MODE_CNT) {
return BAD_VALUE;
}
audio_utils::lock_guard _l(mutex());
mRequestedLatencyMode = mode;
return NO_ERROR;
}
void SpatializerThread::checkOutputStageEffects()
NO_THREAD_SAFETY_ANALYSIS
// 'createEffect_l' requires holding mutex 'AudioFlinger_Mutex' exclusively
{
bool hasVirtualizer = false;
bool hasDownMixer = false;
sp<IAfEffectHandle> finalDownMixer;
{
audio_utils::lock_guard _l(mutex());
sp<IAfEffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
if (chain != 0) {
hasVirtualizer = chain->getEffectFromType_l(FX_IID_SPATIALIZER) != nullptr;
hasDownMixer = chain->getEffectFromType_l(EFFECT_UIID_DOWNMIX) != nullptr;
}
finalDownMixer = mFinalDownMixer;
mFinalDownMixer.clear();
}
if (hasVirtualizer) {
if (finalDownMixer != nullptr) {
int32_t ret;
finalDownMixer->asIEffect()->disable(&ret);
}
finalDownMixer.clear();
} else if (!hasDownMixer) {
std::vector<effect_descriptor_t> descriptors;
status_t status = mAfThreadCallback->getEffectsFactoryHal()->getDescriptors(
EFFECT_UIID_DOWNMIX, &descriptors);
if (status != NO_ERROR) {
return;
}
ALOG_ASSERT(!descriptors.empty(),
"%s getDescriptors() returned no error but empty list", __func__);
finalDownMixer = createEffect_l(nullptr /*client*/, nullptr /*effectClient*/,
0 /*priority*/, AUDIO_SESSION_OUTPUT_STAGE, &descriptors[0], nullptr /*enabled*/,
&status, false /*pinned*/, false /*probe*/, false /*notifyFramesProcessed*/);
if (finalDownMixer == nullptr || (status != NO_ERROR && status != ALREADY_EXISTS)) {
ALOGW("%s error creating downmixer %d", __func__, status);
finalDownMixer.clear();
} else {
int32_t ret;
finalDownMixer->asIEffect()->enable(&ret);
}
}
{
audio_utils::lock_guard _l(mutex());
mFinalDownMixer = finalDownMixer;
}
}
void SpatializerThread::threadLoop_exit()
{
// The Spatializer EffectHandle must be released on the PlaybackThread
// threadLoop() to prevent lock inversion in the SpatializerThread dtor.
mFinalDownMixer.clear();
PlaybackThread::threadLoop_exit();
}
// ----------------------------------------------------------------------------
// Record
// ----------------------------------------------------------------------------
sp<IAfRecordThread> IAfRecordThread::create(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamIn* input,
audio_io_handle_t id,
bool systemReady) {
return sp<RecordThread>::make(afThreadCallback, input, id, systemReady);
}
RecordThread::RecordThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamIn *input,
audio_io_handle_t id,
bool systemReady
) :
ThreadBase(afThreadCallback, id, RECORD, systemReady, false /* isOut */),
mInput(input),
mSource(mInput),
mActiveTracks(&this->mLocalLog),
mRsmpInBuffer(NULL),
// mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l()
mRsmpInRear(0)
, mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
"RecordThreadRO", MemoryHeapBase::READ_ONLY))
// mFastCapture below
, mFastCaptureFutex(0)
// mInputSource
// mPipeSink
// mPipeSource
, mPipeFramesP2(0)
// mPipeMemory
// mFastCaptureNBLogWriter
, mFastTrackAvail(false)
, mBtNrecSuspended(false)
{
snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
mNBLogWriter = afThreadCallback->newWriter_l(kLogSize, mThreadName);
if (mInput->audioHwDev != nullptr) {
mIsMsdDevice = strcmp(
mInput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
}
readInputParameters_l();
// TODO: We may also match on address as well as device type for
// AUDIO_DEVICE_IN_BUS, AUDIO_DEVICE_IN_BLUETOOTH_A2DP, AUDIO_DEVICE_IN_REMOTE_SUBMIX
// TODO: This property should be ensure that only contains one single device type.
mTimestampCorrectedDevice = (audio_devices_t)property_get_int64(
"audio.timestamp.corrected_input_device",
(int64_t)(mIsMsdDevice ? AUDIO_DEVICE_IN_BUS // turn on by default for MSD
: AUDIO_DEVICE_NONE));
// create an NBAIO source for the HAL input stream, and negotiate
mInputSource = new AudioStreamInSource(input->stream);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
#if !LOG_NDEBUG
[[maybe_unused]] ssize_t index =
#else
(void)
#endif
mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
// initialize fast capture depending on configuration
bool initFastCapture;
switch (kUseFastCapture) {
case FastCapture_Never:
initFastCapture = false;
ALOGV("%p kUseFastCapture = Never, initFastCapture = false", this);
break;
case FastCapture_Always:
initFastCapture = true;
ALOGV("%p kUseFastCapture = Always, initFastCapture = true", this);
break;
case FastCapture_Static:
initFastCapture = !mIsMsdDevice // Disable fast capture for MSD BUS devices.
&& audio_is_linear_pcm(mFormat)
&& (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;
ALOGV("%p kUseFastCapture = Static, format = 0x%x, (%lld * 1000) / %u vs %u, "
"initFastCapture = %d, mIsMsdDevice = %d", this, mFormat, (long long)mFrameCount,
mSampleRate, kMinNormalCaptureBufferSizeMs, initFastCapture, mIsMsdDevice);
break;
// case FastCapture_Dynamic:
}
if (initFastCapture) {
// create a Pipe for FastCapture to write to, and for us and fast tracks to read from
NBAIO_Format format = mInputSource->format();
// quadruple-buffering of 20 ms each; this ensures we can sleep for 20ms in RecordThread
size_t pipeFramesP2 = roundup(4 * FMS_20 * mSampleRate / 1000);
size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
void *pipeBuffer = nullptr;
const sp<MemoryDealer> roHeap(readOnlyHeap());
sp<IMemory> pipeMemory;
if ((roHeap == 0) ||
(pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
(pipeBuffer = pipeMemory->unsecurePointer()) == nullptr) {
ALOGE("not enough memory for pipe buffer size=%zu; "
"roHeap=%p, pipeMemory=%p, pipeBuffer=%p; roHeapSize: %lld",
pipeSize, roHeap.get(), pipeMemory.get(), pipeBuffer,
(long long)kRecordThreadReadOnlyHeapSize);
goto failed;
}
// pipe will be shared directly with fast clients, so clear to avoid leaking old information
memset(pipeBuffer, 0, pipeSize);
Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
const NBAIO_Format offersFast[1] = {format};
size_t numCounterOffersFast = 0;
[[maybe_unused]] ssize_t index2 = pipe->negotiate(offersFast, std::size(offersFast),
nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index2 == 0);
mPipeSink = pipe;
PipeReader *pipeReader = new PipeReader(*pipe);
numCounterOffersFast = 0;
index2 = pipeReader->negotiate(offersFast, std::size(offersFast),
nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index2 == 0);
mPipeSource = pipeReader;
mPipeFramesP2 = pipeFramesP2;
mPipeMemory = pipeMemory;
// create fast capture
mFastCapture = new FastCapture();
FastCaptureStateQueue *sq = mFastCapture->sq();
#ifdef STATE_QUEUE_DUMP
// FIXME
#endif
FastCaptureState *state = sq->begin();
state->mCblk = NULL;
state->mInputSource = mInputSource.get();
state->mInputSourceGen++;
state->mPipeSink = pipe;
state->mPipeSinkGen++;
state->mFrameCount = mFrameCount;
state->mCommand = FastCaptureState::COLD_IDLE;
// already done in constructor initialization list
//mFastCaptureFutex = 0;
state->mColdFutexAddr = &mFastCaptureFutex;
state->mColdGen++;
state->mDumpState = &mFastCaptureDumpState;
#ifdef TEE_SINK
// FIXME
#endif
mFastCaptureNBLogWriter =
afThreadCallback->newWriter_l(kFastCaptureLogSize, "FastCapture");
state->mNBLogWriter = mFastCaptureNBLogWriter.get();
sq->end();
sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
// start the fast capture
mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
pid_t tid = mFastCapture->getTid();
sendPrioConfigEvent(getpid(), tid, kPriorityFastCapture, false /*forApp*/);
stream()->setHalThreadPriority(kPriorityFastCapture);
#ifdef AUDIO_WATCHDOG
// FIXME
#endif
mFastTrackAvail = true;
}
#ifdef TEE_SINK
mTee.set(mInputSource->format(), NBAIO_Tee::TEE_FLAG_INPUT_THREAD);
mTee.setId(std::string("_") + std::to_string(mId) + "_C");
#endif
failed: ;
// FIXME mNormalSource
}
RecordThread::~RecordThread()
{
if (mFastCapture != 0) {
FastCaptureStateQueue *sq = mFastCapture->sq();
FastCaptureState *state = sq->begin();
if (state->mCommand == FastCaptureState::COLD_IDLE) {
int32_t old = android_atomic_inc(&mFastCaptureFutex);
if (old == -1) {
(void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
}
}
state->mCommand = FastCaptureState::EXIT;
sq->end();
sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
mFastCapture->join();
mFastCapture.clear();
}
mAfThreadCallback->unregisterWriter(mFastCaptureNBLogWriter);
mAfThreadCallback->unregisterWriter(mNBLogWriter);
free(mRsmpInBuffer);
}
void RecordThread::onFirstRef()
{
run(mThreadName, PRIORITY_URGENT_AUDIO);
}
void RecordThread::preExit()
{
ALOGV(" preExit()");
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); i++) {
sp<IAfRecordTrack> track = mTracks[i];
track->invalidate();
}
mActiveTracks.clear();
mStartStopCV.notify_all();
}
bool RecordThread::threadLoop()
{
nsecs_t lastWarning = 0;
inputStandBy();
reacquire_wakelock:
sp<IAfRecordTrack> activeTrack;
{
audio_utils::lock_guard _l(mutex());
acquireWakeLock_l();
}
// used to request a deferred sleep, to be executed later while mutex is unlocked
uint32_t sleepUs = 0;
// timestamp correction enable is determined under lock, used in processing step.
bool timestampCorrectionEnabled = false;
int64_t lastLoopCountRead = -2; // never matches "previous" loop, when loopCount = 0.
// loop while there is work to do
for (int64_t loopCount = 0;; ++loopCount) { // loopCount used for statistics tracking
Vector<sp<IAfEffectChain>> effectChains;
// activeTracks accumulates a copy of a subset of mActiveTracks
Vector<sp<IAfRecordTrack>> activeTracks;
// reference to the (first and only) active fast track
sp<IAfRecordTrack> fastTrack;
// reference to a fast track which is about to be removed
sp<IAfRecordTrack> fastTrackToRemove;
bool silenceFastCapture = false;
{ // scope for mutex()
audio_utils::unique_lock _l(mutex());
processConfigEvents_l();
// check exitPending here because checkForNewParameters_l() and
// checkForNewParameters_l() can temporarily release mutex()
if (exitPending()) {
break;
}
// sleep with mutex unlocked
if (sleepUs > 0) {
ATRACE_BEGIN("sleepC");
(void)mWaitWorkCV.wait_for(_l, std::chrono::microseconds(sleepUs));
ATRACE_END();
sleepUs = 0;
continue;
}
// if no active track(s), then standby and release wakelock
size_t size = mActiveTracks.size();
if (size == 0) {
standbyIfNotAlreadyInStandby();
// exitPending() can't become true here
releaseWakeLock_l();
ALOGV("RecordThread: loop stopping");
// go to sleep
mWaitWorkCV.wait(_l);
ALOGV("RecordThread: loop starting");
goto reacquire_wakelock;
}
bool doBroadcast = false;
bool allStopped = true;
for (size_t i = 0; i < size; ) {
activeTrack = mActiveTracks[i];
if (activeTrack->isTerminated()) {
if (activeTrack->isFastTrack()) {
ALOG_ASSERT(fastTrackToRemove == 0);
fastTrackToRemove = activeTrack;
}
removeTrack_l(activeTrack);
mActiveTracks.remove(activeTrack);
size--;
continue;
}
IAfTrackBase::track_state activeTrackState = activeTrack->state();
switch (activeTrackState) {
case IAfTrackBase::PAUSING:
mActiveTracks.remove(activeTrack);
activeTrack->setState(IAfTrackBase::PAUSED);
if (activeTrack->isFastTrack()) {
ALOGV("%s fast track is paused, thus removed from active list", __func__);
// Keep a ref on fast track to wait for FastCapture thread to get updated
// state before potential track removal
fastTrackToRemove = activeTrack;
}
doBroadcast = true;
size--;
continue;
case IAfTrackBase::STARTING_1:
sleepUs = 10000;
i++;
allStopped = false;
continue;
case IAfTrackBase::STARTING_2:
doBroadcast = true;
if (mStandby) {
mThreadMetrics.logBeginInterval();
mThreadSnapshot.onBegin();
mStandby = false;
}
activeTrack->setState(IAfTrackBase::ACTIVE);
allStopped = false;
break;
case IAfTrackBase::ACTIVE:
allStopped = false;
break;
case IAfTrackBase::IDLE: // cannot be on ActiveTracks if idle
case IAfTrackBase::PAUSED: // cannot be on ActiveTracks if paused
case IAfTrackBase::STOPPED: // cannot be on ActiveTracks if destroyed/terminated
default:
LOG_ALWAYS_FATAL("%s: Unexpected active track state:%d, id:%d, tracks:%zu",
__func__, activeTrackState, activeTrack->id(), size);
}
if (activeTrack->isFastTrack()) {
ALOG_ASSERT(!mFastTrackAvail);
ALOG_ASSERT(fastTrack == 0);
// if the active fast track is silenced either:
// 1) silence the whole capture from fast capture buffer if this is
// the only active track
// 2) invalidate this track: this will cause the client to reconnect and possibly
// be invalidated again until unsilenced
bool invalidate = false;
if (activeTrack->isSilenced()) {
if (size > 1) {
invalidate = true;
} else {
silenceFastCapture = true;
}
}
// Invalidate fast tracks if access to audio history is required as this is not
// possible with fast tracks. Once the fast track has been invalidated, no new
// fast track will be created until mMaxSharedAudioHistoryMs is cleared.
if (mMaxSharedAudioHistoryMs != 0) {
invalidate = true;
}
if (invalidate) {
activeTrack->invalidate();
ALOG_ASSERT(fastTrackToRemove == 0);
fastTrackToRemove = activeTrack;
removeTrack_l(activeTrack);
mActiveTracks.remove(activeTrack);
size--;
continue;
}
fastTrack = activeTrack;
}
activeTracks.add(activeTrack);
i++;
}
mActiveTracks.updatePowerState_l(this);
updateMetadata_l();
if (allStopped) {
standbyIfNotAlreadyInStandby();
}
if (doBroadcast) {
mStartStopCV.notify_all();
}
// sleep if there are no active tracks to process
if (activeTracks.isEmpty()) {
if (sleepUs == 0) {
sleepUs = kRecordThreadSleepUs;
}
continue;
}
sleepUs = 0;
timestampCorrectionEnabled = isTimestampCorrectionEnabled_l();
lockEffectChains_l(effectChains);
}
// thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
size_t size = effectChains.size();
for (size_t i = 0; i < size; i++) {
// thread mutex is not locked, but effect chain is locked
effectChains[i]->process_l();
}
// Push a new fast capture state if fast capture is not already running, or cblk change
if (mFastCapture != 0) {
FastCaptureStateQueue *sq = mFastCapture->sq();
FastCaptureState *state = sq->begin();
bool didModify = false;
FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
(kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
if (state->mCommand == FastCaptureState::COLD_IDLE) {
int32_t old = android_atomic_inc(&mFastCaptureFutex);
if (old == -1) {
(void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
}
}
state->mCommand = FastCaptureState::READ_WRITE;
#if 0 // FIXME
mFastCaptureDumpState.increaseSamplingN(mAfThreadCallback->isLowRamDevice() ?
FastThreadDumpState::kSamplingNforLowRamDevice :
FastThreadDumpState::kSamplingN);
#endif
didModify = true;
}
audio_track_cblk_t *cblkOld = state->mCblk;
audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
if (cblkNew != cblkOld) {
state->mCblk = cblkNew;
// block until acked if removing a fast track
if (cblkOld != NULL) {
block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
}
didModify = true;
}
AudioBufferProvider* abp = (fastTrack != 0 && fastTrack->isPatchTrack()) ?
reinterpret_cast<AudioBufferProvider*>(fastTrack.get()) : nullptr;
if (state->mFastPatchRecordBufferProvider != abp) {
state->mFastPatchRecordBufferProvider = abp;
state->mFastPatchRecordFormat = fastTrack == 0 ?
AUDIO_FORMAT_INVALID : fastTrack->format();
didModify = true;
}
if (state->mSilenceCapture != silenceFastCapture) {
state->mSilenceCapture = silenceFastCapture;
didModify = true;
}
sq->end(didModify);
if (didModify) {
sq->push(block);
#if 0
if (kUseFastCapture == FastCapture_Dynamic) {
mNormalSource = mPipeSource;
}
#endif
}
}
// now run the fast track destructor with thread mutex unlocked
fastTrackToRemove.clear();
// Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
// Only the client(s) that are too slow will overrun. But if even the fastest client is too
// slow, then this RecordThread will overrun by not calling HAL read often enough.
// If destination is non-contiguous, first read past the nominal end of buffer, then
// copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
ssize_t framesRead = 0; // not needed, remove clang-tidy warning.
const int64_t lastIoBeginNs = systemTime(); // start IO timing
// If an NBAIO source is present, use it to read the normal capture's data
if (mPipeSource != 0) {
size_t framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
// The audio fifo read() returns OVERRUN on overflow, and advances the read pointer
// to the full buffer point (clearing the overflow condition). Upon OVERRUN error,
// we immediately retry the read() to get data and prevent another overflow.
for (int retries = 0; retries <= 2; ++retries) {
ALOGW_IF(retries > 0, "overrun on read from pipe, retry #%d", retries);
framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
framesToRead);
if (framesRead != OVERRUN) break;
}
const ssize_t availableToRead = mPipeSource->availableToRead();
if (availableToRead >= 0) {
mMonopipePipeDepthStats.add(availableToRead);
// PipeSource is the primary clock. It is up to the AudioRecord client to keep up.
LOG_ALWAYS_FATAL_IF((size_t)availableToRead > mPipeFramesP2,
"more frames to read than fifo size, %zd > %zu",
availableToRead, mPipeFramesP2);
const size_t pipeFramesFree = mPipeFramesP2 - availableToRead;
const size_t sleepFrames = min(pipeFramesFree, mRsmpInFramesP2) / 2;
ALOGVV("mPipeFramesP2:%zu mRsmpInFramesP2:%zu sleepFrames:%zu availableToRead:%zd",
mPipeFramesP2, mRsmpInFramesP2, sleepFrames, availableToRead);
sleepUs = (sleepFrames * 1000000LL) / mSampleRate;
}
if (framesRead < 0) {
status_t status = (status_t) framesRead;
switch (status) {
case OVERRUN:
ALOGW("overrun on read from pipe");
framesRead = 0;
break;
case NEGOTIATE:
ALOGE("re-negotiation is needed");
framesRead = -1; // Will cause an attempt to recover.
break;
default:
ALOGE("unknown error %d on read from pipe", status);
break;
}
}
// otherwise use the HAL / AudioStreamIn directly
} else {
ATRACE_BEGIN("read");
size_t bytesRead;
status_t result = mSource->read(
(uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize, &bytesRead);
ATRACE_END();
if (result < 0) {
framesRead = result;
} else {
framesRead = bytesRead / mFrameSize;
}
}
const int64_t lastIoEndNs = systemTime(); // end IO timing
// Update server timestamp with server stats
// systemTime() is optional if the hardware supports timestamps.
if (framesRead >= 0) {
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead;
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = lastIoEndNs;
}
// Update server timestamp with kernel stats
if (mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) {
int64_t position, time;
if (mStandby) {
mTimestampVerifier.discontinuity(audio_is_linear_pcm(mFormat) ?
mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS :
mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
} else if (mSource->getCapturePosition(&position, &time) == NO_ERROR
&& time > mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]) {
mTimestampVerifier.add(position, time, mSampleRate);
if (timestampCorrectionEnabled) {
ALOGVV("TS_BEFORE: %d %lld %lld",
id(), (long long)time, (long long)position);
auto correctedTimestamp = mTimestampVerifier.getLastCorrectedTimestamp();
position = correctedTimestamp.mFrames;
time = correctedTimestamp.mTimeNs;
ALOGVV("TS_AFTER: %d %lld %lld",
id(), (long long)time, (long long)position);
}
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position;
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time;
// Note: In general record buffers should tend to be empty in
// a properly running pipeline.
//
// Also, it is not advantageous to call get_presentation_position during the read
// as the read obtains a lock, preventing the timestamp call from executing.
} else {
mTimestampVerifier.error();
}
}
// From the timestamp, input read latency is negative output write latency.
const audio_input_flags_t flags = mInput != NULL ? mInput->flags : AUDIO_INPUT_FLAG_NONE;
const double latencyMs = IAfRecordTrack::checkServerLatencySupported(mFormat, flags)
? - mTimestamp.getOutputServerLatencyMs(mSampleRate) : 0.;
if (latencyMs != 0.) { // note 0. means timestamp is empty.
mLatencyMs.add(latencyMs);
}
// Use this to track timestamp information
// ALOGD("%s", mTimestamp.toString().c_str());
if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
ALOGE("read failed: framesRead=%zd", framesRead);
// Force input into standby so that it tries to recover at next read attempt
inputStandBy();
sleepUs = kRecordThreadSleepUs;
}
if (framesRead <= 0) {
goto unlock;
}
ALOG_ASSERT(framesRead > 0);
mFramesRead += framesRead;
#ifdef TEE_SINK
(void)mTee.write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
#endif
// If destination is non-contiguous, we now correct for reading past end of buffer.
{
size_t part1 = mRsmpInFramesP2 - rear;
if ((size_t) framesRead > part1) {
memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
(framesRead - part1) * mFrameSize);
}
}
mRsmpInRear = audio_utils::safe_add_overflow(mRsmpInRear, (int32_t)framesRead);
size = activeTracks.size();
// loop over each active track
for (size_t i = 0; i < size; i++) {
activeTrack = activeTracks[i];
// skip fast tracks, as those are handled directly by FastCapture
if (activeTrack->isFastTrack()) {
continue;
}
// TODO: This code probably should be moved to RecordTrack.
// TODO: Update the activeTrack buffer converter in case of reconfigure.
enum {
OVERRUN_UNKNOWN,
OVERRUN_TRUE,
OVERRUN_FALSE
} overrun = OVERRUN_UNKNOWN;
// loop over getNextBuffer to handle circular sink
for (;;) {
activeTrack->sinkBuffer().frameCount = ~0;
status_t status = activeTrack->getNextBuffer(&activeTrack->sinkBuffer());
size_t framesOut = activeTrack->sinkBuffer().frameCount;
LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
// check available frames and handle overrun conditions
// if the record track isn't draining fast enough.
bool hasOverrun;
size_t framesIn;
activeTrack->resamplerBufferProvider()->sync(&framesIn, &hasOverrun);
if (hasOverrun) {
overrun = OVERRUN_TRUE;
}
if (framesOut == 0 || framesIn == 0) {
break;
}
// Don't allow framesOut to be larger than what is possible with resampling
// from framesIn.
// This isn't strictly necessary but helps limit buffer resizing in
// RecordBufferConverter. TODO: remove when no longer needed.
if (audio_is_linear_pcm(activeTrack->format())) {
framesOut = min(framesOut,
destinationFramesPossible(
framesIn, mSampleRate, activeTrack->sampleRate()));
}
if (activeTrack->isDirect()) {
// No RecordBufferConverter used for direct streams. Pass
// straight from RecordThread buffer to RecordTrack buffer.
AudioBufferProvider::Buffer buffer;
buffer.frameCount = framesOut;
const status_t getNextBufferStatus =
activeTrack->resamplerBufferProvider()->getNextBuffer(&buffer);
if (getNextBufferStatus == OK && buffer.frameCount != 0) {
ALOGV_IF(buffer.frameCount != framesOut,
"%s() read less than expected (%zu vs %zu)",
__func__, buffer.frameCount, framesOut);
framesOut = buffer.frameCount;
memcpy(activeTrack->sinkBuffer().raw,
buffer.raw, buffer.frameCount * mFrameSize);
activeTrack->resamplerBufferProvider()->releaseBuffer(&buffer);
} else {
framesOut = 0;
ALOGE("%s() cannot fill request, status: %d, frameCount: %zu",
__func__, getNextBufferStatus, buffer.frameCount);
}
} else {
// process frames from the RecordThread buffer provider to the RecordTrack
// buffer
framesOut = activeTrack->recordBufferConverter()->convert(
activeTrack->sinkBuffer().raw,
activeTrack->resamplerBufferProvider(),
framesOut);
}
if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
overrun = OVERRUN_FALSE;
}
// MediaSyncEvent handling: Synchronize AudioRecord to AudioTrack completion.
const ssize_t framesToDrop =
activeTrack->synchronizedRecordState().updateRecordFrames(framesOut);
if (framesToDrop == 0) {
// no sync event, process normally, otherwise ignore.
if (framesOut > 0) {
activeTrack->sinkBuffer().frameCount = framesOut;
// Sanitize before releasing if the track has no access to the source data
// An idle UID receives silence from non virtual devices until active
if (activeTrack->isSilenced()) {
memset(activeTrack->sinkBuffer().raw,
0, framesOut * activeTrack->frameSize());
}
activeTrack->releaseBuffer(&activeTrack->sinkBuffer());
}
}
if (framesOut == 0) {
break;
}
}
switch (overrun) {
case OVERRUN_TRUE:
// client isn't retrieving buffers fast enough
if (!activeTrack->setOverflow()) {
nsecs_t now = systemTime();
// FIXME should lastWarning per track?
if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("RecordThread: buffer overflow");
lastWarning = now;
}
}
break;
case OVERRUN_FALSE:
activeTrack->clearOverflow();
break;
case OVERRUN_UNKNOWN:
break;
}
// update frame information and push timestamp out
activeTrack->updateTrackFrameInfo(
activeTrack->serverProxy()->framesReleased(),
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
mSampleRate, mTimestamp);
}
unlock:
// enable changes in effect chain
unlockEffectChains(effectChains);
// effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
if (audio_has_proportional_frames(mFormat)
&& loopCount == lastLoopCountRead + 1) {
const int64_t readPeriodNs = lastIoEndNs - mLastIoEndNs;
const double jitterMs =
TimestampVerifier<int64_t, int64_t>::computeJitterMs(
{framesRead, readPeriodNs},
{0, 0} /* lastTimestamp */, mSampleRate);
const double processMs = (lastIoBeginNs - mLastIoEndNs) * 1e-6;
audio_utils::lock_guard _l(mutex());
mIoJitterMs.add(jitterMs);
mProcessTimeMs.add(processMs);
}
// update timing info.
mLastIoBeginNs = lastIoBeginNs;
mLastIoEndNs = lastIoEndNs;
lastLoopCountRead = loopCount;
}
standbyIfNotAlreadyInStandby();
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); i++) {
sp<IAfRecordTrack> track = mTracks[i];
track->invalidate();
}
mActiveTracks.clear();
mStartStopCV.notify_all();
}
releaseWakeLock();
ALOGV("RecordThread %p exiting", this);
return false;
}
void RecordThread::standbyIfNotAlreadyInStandby()
{
if (!mStandby) {
inputStandBy();
mThreadMetrics.logEndInterval();
mThreadSnapshot.onEnd();
mStandby = true;
}
}
void RecordThread::inputStandBy()
{
// Idle the fast capture if it's currently running
if (mFastCapture != 0) {
FastCaptureStateQueue *sq = mFastCapture->sq();
FastCaptureState *state = sq->begin();
if (!(state->mCommand & FastCaptureState::IDLE)) {
state->mCommand = FastCaptureState::COLD_IDLE;
state->mColdFutexAddr = &mFastCaptureFutex;
state->mColdGen++;
mFastCaptureFutex = 0;
sq->end();
// BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
#if 0
if (kUseFastCapture == FastCapture_Dynamic) {
// FIXME
}
#endif
#ifdef AUDIO_WATCHDOG
// FIXME
#endif
} else {
sq->end(false /*didModify*/);
}
}
status_t result = mSource->standby();
ALOGE_IF(result != OK, "Error when putting input stream into standby: %d", result);
// If going into standby, flush the pipe source.
if (mPipeSource.get() != nullptr) {
const ssize_t flushed = mPipeSource->flush();
if (flushed > 0) {
ALOGV("Input standby flushed PipeSource %zd frames", flushed);
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += flushed;
mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
}
}
}
// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mutex() held
sp<IAfRecordTrack> RecordThread::createRecordTrack_l(
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
audio_session_t sessionId,
size_t *pNotificationFrameCount,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
audio_input_flags_t *flags,
pid_t tid,
status_t *status,
audio_port_handle_t portId,
int32_t maxSharedAudioHistoryMs)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
sp<IAfRecordTrack> track;
status_t lStatus;
audio_input_flags_t inputFlags = mInput->flags;
audio_input_flags_t requestedFlags = *flags;
uint32_t sampleRate;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createRecordTrack_l() audio driver not initialized");
goto Exit;
}
if (!audio_is_linear_pcm(mFormat) && (*flags & AUDIO_INPUT_FLAG_DIRECT) == 0) {
ALOGE("createRecordTrack_l() on an encoded stream requires AUDIO_INPUT_FLAG_DIRECT");
lStatus = BAD_VALUE;
goto Exit;
}
if (maxSharedAudioHistoryMs != 0) {
if (!captureHotwordAllowed(attributionSource)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (maxSharedAudioHistoryMs < 0
|| maxSharedAudioHistoryMs > kMaxSharedAudioHistoryMs) {
lStatus = BAD_VALUE;
goto Exit;
}
}
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
sampleRate = *pSampleRate;
// special case for FAST flag considered OK if fast capture is present and access to
// audio history is not required
if (hasFastCapture() && mMaxSharedAudioHistoryMs == 0) {
inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST);
}
// Check if requested flags are compatible with input stream flags
if ((*flags & inputFlags) != *flags) {
ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and"
" input flags (%08x)",
*flags, inputFlags);
*flags = (audio_input_flags_t)(*flags & inputFlags);
}
// client expresses a preference for FAST and no access to audio history,
// but we get the final say
if (*flags & AUDIO_INPUT_FLAG_FAST && maxSharedAudioHistoryMs == 0) {
if (
// we formerly checked for a callback handler (non-0 tid),
// but that is no longer required for TRANSFER_OBTAIN mode
// No need to match hardware format, format conversion will be done in client side.
//
// Frame count is not specified (0), or is less than or equal the pipe depth.
// It is OK to provide a higher capacity than requested.
// We will force it to mPipeFramesP2 below.
(frameCount <= mPipeFramesP2) &&
// PCM data
audio_is_linear_pcm(format) &&
// hardware channel mask
(channelMask == mChannelMask) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// record thread has an associated fast capture
hasFastCapture() &&
// there are sufficient fast track slots available
mFastTrackAvail
) {
// check compatibility with audio effects.
audio_utils::lock_guard _l(mutex());
// Do not accept FAST flag if the session has software effects
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
audio_input_flags_t old = *flags;
chain->checkInputFlagCompatibility(flags);
if (old != *flags) {
ALOGV("%p AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x",
this, (int)old, (int)*flags);
}
}
ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0,
"%p AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
this, frameCount, mFrameCount);
} else {
ALOGV("%p AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
"format=%#x isLinear=%d mFormat=%#x channelMask=%#x sampleRate=%u mSampleRate=%u "
"hasFastCapture=%d tid=%d mFastTrackAvail=%d",
this, frameCount, mFrameCount, mPipeFramesP2,
format, audio_is_linear_pcm(format), mFormat, channelMask, sampleRate, mSampleRate,
hasFastCapture(), tid, mFastTrackAvail);
*flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST);
}
}
// If FAST or RAW flags were corrected, ask caller to request new input from audio policy
if ((*flags & AUDIO_INPUT_FLAG_FAST) !=
(requestedFlags & AUDIO_INPUT_FLAG_FAST)) {
*flags = (audio_input_flags_t) (*flags & ~(AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW));
lStatus = BAD_TYPE;
goto Exit;
}
// compute track buffer size in frames, and suggest the notification frame count
if (*flags & AUDIO_INPUT_FLAG_FAST) {
// fast track: frame count is exactly the pipe depth
frameCount = mPipeFramesP2;
// ignore requested notificationFrames, and always notify exactly once every HAL buffer
notificationFrameCount = mFrameCount;
} else {
// not fast track: max notification period is resampled equivalent of one HAL buffer time
// or 20 ms if there is a fast capture
// TODO This could be a roundupRatio inline, and const
size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
* sampleRate + mSampleRate - 1) / mSampleRate;
// minimum number of notification periods is at least kMinNotifications,
// and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
static const size_t kMinNotifications = 3;
static const uint32_t kMinMs = 30;
// TODO This could be a roundupRatio inline
const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
// TODO This could be a roundupRatio inline
const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
maxNotificationFrames;
const size_t minFrameCount = maxNotificationFrames *
max(kMinNotifications, minNotificationsByMs);
frameCount = max(frameCount, minFrameCount);
if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
notificationFrameCount = maxNotificationFrames;
}
}
*pFrameCount = frameCount;
*pNotificationFrameCount = notificationFrameCount;
{ // scope for mutex()
audio_utils::lock_guard _l(mutex());
int32_t startFrames = -1;
if (!mSharedAudioPackageName.empty()
&& mSharedAudioPackageName == attributionSource.packageName
&& mSharedAudioSessionId == sessionId
&& captureHotwordAllowed(attributionSource)) {
startFrames = mSharedAudioStartFrames;
}
track = IAfRecordTrack::create(this, client, attr, sampleRate,
format, channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid,
attributionSource, *flags, IAfTrackBase::TYPE_DEFAULT, portId,
startFrames);
lStatus = track->initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) {
pid_t callingPid = IPCThreadState::self()->getCallingPid();
// we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
// so ask activity manager to do this on our behalf
sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/);
}
if (maxSharedAudioHistoryMs != 0) {
sendResizeBufferConfigEvent_l(maxSharedAudioHistoryMs);
}
}
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}
status_t RecordThread::start(IAfRecordTrack* recordTrack,
AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
sp<ThreadBase> strongMe = this;
status_t status = NO_ERROR;
if (event == AudioSystem::SYNC_EVENT_NONE) {
recordTrack->clearSyncStartEvent();
} else if (event != AudioSystem::SYNC_EVENT_SAME) {
recordTrack->synchronizedRecordState().startRecording(
mAfThreadCallback->createSyncEvent(
event, triggerSession,
recordTrack->sessionId(), syncStartEventCallback, recordTrack));
}
{
// This section is a rendezvous between binder thread executing start() and RecordThread
audio_utils::lock_guard lock(mutex());
if (recordTrack->isInvalid()) {
recordTrack->clearSyncStartEvent();
ALOGW("%s track %d: invalidated before startInput", __func__, recordTrack->portId());
return DEAD_OBJECT;
}
if (mActiveTracks.indexOf(recordTrack) >= 0) {
if (recordTrack->state() == IAfTrackBase::PAUSING) {
// We haven't stopped yet (moved to PAUSED and not in mActiveTracks)
// so no need to startInput().
ALOGV("active record track PAUSING -> ACTIVE");
recordTrack->setState(IAfTrackBase::ACTIVE);
} else {
ALOGV("active record track state %d", (int)recordTrack->state());
}
return status;
}
// TODO consider other ways of handling this, such as changing the state to :STARTING and
// adding the track to mActiveTracks after returning from AudioSystem::startInput(),
// or using a separate command thread
recordTrack->setState(IAfTrackBase::STARTING_1);
mActiveTracks.add(recordTrack);
if (recordTrack->isExternalTrack()) {
mutex().unlock();
status = AudioSystem::startInput(recordTrack->portId());
mutex().lock();
if (recordTrack->isInvalid()) {
recordTrack->clearSyncStartEvent();
if (status == NO_ERROR && recordTrack->state() == IAfTrackBase::STARTING_1) {
recordTrack->setState(IAfTrackBase::STARTING_2);
// STARTING_2 forces destroy to call stopInput.
}
ALOGW("%s track %d: invalidated after startInput", __func__, recordTrack->portId());
return DEAD_OBJECT;
}
if (recordTrack->state() != IAfTrackBase::STARTING_1) {
ALOGW("%s(%d): unsynchronized mState:%d change",
__func__, recordTrack->id(), (int)recordTrack->state());
// Someone else has changed state, let them take over,
// leave mState in the new state.
recordTrack->clearSyncStartEvent();
return INVALID_OPERATION;
}
// we're ok, but perhaps startInput has failed
if (status != NO_ERROR) {
ALOGW("%s(%d): startInput failed, status %d",
__func__, recordTrack->id(), status);
// We are in ActiveTracks if STARTING_1 and valid, so remove from ActiveTracks,
// leave in STARTING_1, so destroy() will not call stopInput.
mActiveTracks.remove(recordTrack);
recordTrack->clearSyncStartEvent();
return status;
}
sendIoConfigEvent_l(
AUDIO_CLIENT_STARTED, recordTrack->creatorPid(), recordTrack->portId());
}
recordTrack->logBeginInterval(patchSourcesToString(&mPatch)); // log to MediaMetrics
// Catch up with current buffer indices if thread is already running.
// This is what makes a new client discard all buffered data. If the track's mRsmpInFront
// was initialized to some value closer to the thread's mRsmpInFront, then the track could
// see previously buffered data before it called start(), but with greater risk of overrun.
recordTrack->resamplerBufferProvider()->reset();
if (!recordTrack->isDirect()) {
// clear any converter state as new data will be discontinuous
recordTrack->recordBufferConverter()->reset();
}
recordTrack->setState(IAfTrackBase::STARTING_2);
// signal thread to start
mWaitWorkCV.notify_all();
return status;
}
}
void RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
{
const sp<SyncEvent> strongEvent = event.promote();
if (strongEvent != 0) {
sp<IAfTrackBase> ptr =
std::any_cast<const wp<IAfTrackBase>>(strongEvent->cookie()).promote();
if (ptr != nullptr) {
// TODO(b/291317898) handleSyncStartEvent is in IAfTrackBase not IAfRecordTrack.
ptr->handleSyncStartEvent(strongEvent);
}
}
}
bool RecordThread::stop(IAfRecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
audio_utils::unique_lock _l(mutex());
// if we're invalid, we can't be on the ActiveTracks.
if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->state() == IAfTrackBase::PAUSING) {
return false;
}
// note that threadLoop may still be processing the track at this point [without lock]
recordTrack->setState(IAfTrackBase::PAUSING);
// NOTE: Waiting here is important to keep stop synchronous.
// This is needed for proper patchRecord peer release.
while (recordTrack->state() == IAfTrackBase::PAUSING && !recordTrack->isInvalid()) {
mWaitWorkCV.notify_all(); // signal thread to stop
mStartStopCV.wait(_l);
}
if (recordTrack->state() == IAfTrackBase::PAUSED) { // successful stop
ALOGV("Record stopped OK");
return true;
}
// don't handle anything - we've been invalidated or restarted and in a different state
ALOGW_IF("%s(%d): unsynchronized stop, state: %d",
__func__, recordTrack->id(), recordTrack->state());
return false;
}
bool RecordThread::isValidSyncEvent(const sp<SyncEvent>& /* event */) const
{
return false;
}
status_t RecordThread::setSyncEvent(const sp<SyncEvent>& /* event */)
{
#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
if (!isValidSyncEvent(event)) {
return BAD_VALUE;
}
audio_session_t eventSession = event->triggerSession();
status_t ret = NAME_NOT_FOUND;
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size(); i++) {
sp<IAfRecordTrack> track = mTracks[i];
if (eventSession == track->sessionId()) {
(void) track->setSyncEvent(event);
ret = NO_ERROR;
}
}
return ret;
#else
return BAD_VALUE;
#endif
}
status_t RecordThread::getActiveMicrophones(
std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
{
ALOGV("RecordThread::getActiveMicrophones");
audio_utils::lock_guard _l(mutex());
if (!isStreamInitialized()) {
return NO_INIT;
}
status_t status = mInput->stream->getActiveMicrophones(activeMicrophones);
return status;
}
status_t RecordThread::setPreferredMicrophoneDirection(
audio_microphone_direction_t direction)
{
ALOGV("setPreferredMicrophoneDirection(%d)", direction);
audio_utils::lock_guard _l(mutex());
if (!isStreamInitialized()) {
return NO_INIT;
}
return mInput->stream->setPreferredMicrophoneDirection(direction);
}
status_t RecordThread::setPreferredMicrophoneFieldDimension(float zoom)
{
ALOGV("setPreferredMicrophoneFieldDimension(%f)", zoom);
audio_utils::lock_guard _l(mutex());
if (!isStreamInitialized()) {
return NO_INIT;
}
return mInput->stream->setPreferredMicrophoneFieldDimension(zoom);
}
status_t RecordThread::shareAudioHistory(
const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
int64_t sharedAudioStartMs) {
audio_utils::lock_guard _l(mutex());
return shareAudioHistory_l(sharedAudioPackageName, sharedSessionId, sharedAudioStartMs);
}
status_t RecordThread::shareAudioHistory_l(
const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
int64_t sharedAudioStartMs) {
if ((hasAudioSession_l(sharedSessionId) & ThreadBase::TRACK_SESSION) == 0) {
return BAD_VALUE;
}
if (sharedAudioStartMs < 0
|| sharedAudioStartMs > INT64_MAX / mSampleRate) {
return BAD_VALUE;
}
// Current implementation of the input resampling buffer wraps around indexes at 32 bit.
// As we cannot detect more than one wraparound, only accept values up current write position
// after one wraparound
// We assume recent wraparounds on mRsmpInRear only given it is unlikely that the requesting
// app waits several hours after the start time was computed.
int64_t sharedAudioStartFrames = sharedAudioStartMs * mSampleRate / 1000;
const int32_t sharedOffset = audio_utils::safe_sub_overflow(mRsmpInRear,
(int32_t)sharedAudioStartFrames);
// Bring the start frame position within the input buffer to match the documented
// "best effort" behavior of the API.
if (sharedOffset < 0) {
sharedAudioStartFrames = mRsmpInRear;
} else if (sharedOffset > static_cast<signed>(mRsmpInFrames)) {
sharedAudioStartFrames =
audio_utils::safe_sub_overflow(mRsmpInRear, (int32_t)mRsmpInFrames);
}
mSharedAudioPackageName = sharedAudioPackageName;
if (mSharedAudioPackageName.empty()) {
resetAudioHistory_l();
} else {
mSharedAudioSessionId = sharedSessionId;
mSharedAudioStartFrames = (int32_t)sharedAudioStartFrames;
}
return NO_ERROR;
}
void RecordThread::resetAudioHistory_l() {
mSharedAudioSessionId = AUDIO_SESSION_NONE;
mSharedAudioStartFrames = -1;
mSharedAudioPackageName = "";
}
ThreadBase::MetadataUpdate RecordThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
}
StreamInHalInterface::SinkMetadata metadata;
auto backInserter = std::back_inserter(metadata.tracks);
for (const sp<IAfRecordTrack>& track : mActiveTracks) {
track->copyMetadataTo(backInserter);
}
mInput->stream->updateSinkMetadata(metadata);
MetadataUpdate change;
change.recordMetadataUpdate = metadata.tracks;
return change;
}
// destroyTrack_l() must be called with ThreadBase::mutex() held
void RecordThread::destroyTrack_l(const sp<IAfRecordTrack>& track)
{
track->terminate();
track->setState(IAfTrackBase::STOPPED);
// active tracks are removed by threadLoop()
if (mActiveTracks.indexOf(track) < 0) {
removeTrack_l(track);
}
}
void RecordThread::removeTrack_l(const sp<IAfRecordTrack>& track)
{
String8 result;
track->appendDump(result, false /* active */);
mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.c_str());
mTracks.remove(track);
// need anything related to effects here?
if (track->isFastTrack()) {
ALOG_ASSERT(!mFastTrackAvail);
mFastTrackAvail = true;
}
}
void RecordThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
{
AudioStreamIn *input = mInput;
audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE;
dprintf(fd, " AudioStreamIn: %p flags %#x (%s)\n",
input, flags, toString(flags).c_str());
dprintf(fd, " Frames read: %lld\n", (long long)mFramesRead);
if (mActiveTracks.isEmpty()) {
dprintf(fd, " No active record clients\n");
}
if (input != nullptr) {
dprintf(fd, " Hal stream dump:\n");
(void)input->stream->dump(fd);
}
dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
// Make a non-atomic copy of fast capture dump state so it won't change underneath us
// while we are dumping it. It may be inconsistent, but it won't mutate!
// This is a large object so we place it on the heap.
// FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
const std::unique_ptr<FastCaptureDumpState> copy =
std::make_unique<FastCaptureDumpState>(mFastCaptureDumpState);
copy->dump(fd);
}
void RecordThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
size_t numtracks = mTracks.size();
size_t numactive = mActiveTracks.size();
size_t numactiveseen = 0;
dprintf(fd, " %zu Tracks", numtracks);
const char *prefix = " ";
if (numtracks) {
dprintf(fd, " of which %zu are active\n", numactive);
result.append(prefix);
mTracks[0]->appendDumpHeader(result);
for (size_t i = 0; i < numtracks ; ++i) {
sp<IAfRecordTrack> track = mTracks[i];
if (track != 0) {
bool active = mActiveTracks.indexOf(track) >= 0;
if (active) {
numactiveseen++;
}
result.append(prefix);
track->appendDump(result, active);
}
}
} else {
dprintf(fd, "\n");
}
if (numactiveseen != numactive) {
result.append(" The following tracks are in the active list but"
" not in the track list\n");
result.append(prefix);
mActiveTracks[0]->appendDumpHeader(result);
for (size_t i = 0; i < numactive; ++i) {
sp<IAfRecordTrack> track = mActiveTracks[i];
if (mTracks.indexOf(track) < 0) {
result.append(prefix);
track->appendDump(result, true /* active */);
}
}
}
write(fd, result.c_str(), result.size());
}
void RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mTracks.size() ; i++) {
sp<IAfRecordTrack> track = mTracks[i];
if (track != 0 && track->portId() == portId) {
track->setSilenced(silenced);
}
}
}
void ResamplerBufferProvider::reset()
{
const auto threadBase = mRecordTrack->thread().promote();
auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
mRsmpInUnrel = 0;
const int32_t rear = recordThread->mRsmpInRear;
ssize_t deltaFrames = 0;
if (mRecordTrack->startFrames() >= 0) {
int32_t startFrames = mRecordTrack->startFrames();
// Accept a recent wraparound of mRsmpInRear
if (startFrames <= rear) {
deltaFrames = rear - startFrames;
} else {
deltaFrames = (int32_t)((int64_t)rear + UINT32_MAX + 1 - startFrames);
}
// start frame cannot be further in the past than start of resampling buffer
if ((size_t) deltaFrames > recordThread->mRsmpInFrames) {
deltaFrames = recordThread->mRsmpInFrames;
}
}
mRsmpInFront = audio_utils::safe_sub_overflow(rear, static_cast<int32_t>(deltaFrames));
}
void ResamplerBufferProvider::sync(
size_t *framesAvailable, bool *hasOverrun)
{
const auto threadBase = mRecordTrack->thread().promote();
auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
const int32_t rear = recordThread->mRsmpInRear;
const int32_t front = mRsmpInFront;
const ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
size_t framesIn;
bool overrun = false;
if (filled < 0) {
// should not happen, but treat like a massive overrun and re-sync
framesIn = 0;
mRsmpInFront = rear;
overrun = true;
} else if ((size_t) filled <= recordThread->mRsmpInFrames) {
framesIn = (size_t) filled;
} else {
// client is not keeping up with server, but give it latest data
framesIn = recordThread->mRsmpInFrames;
mRsmpInFront = /* front = */ audio_utils::safe_sub_overflow(
rear, static_cast<int32_t>(framesIn));
overrun = true;
}
if (framesAvailable != NULL) {
*framesAvailable = framesIn;
}
if (hasOverrun != NULL) {
*hasOverrun = overrun;
}
}
// AudioBufferProvider interface
status_t ResamplerBufferProvider::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
const auto threadBase = mRecordTrack->thread().promote();
if (threadBase == 0) {
buffer->frameCount = 0;
buffer->raw = NULL;
return NOT_ENOUGH_DATA;
}
auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
int32_t rear = recordThread->mRsmpInRear;
int32_t front = mRsmpInFront;
ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
// FIXME should not be P2 (don't want to increase latency)
// FIXME if client not keeping up, discard
LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
// 'filled' may be non-contiguous, so return only the first contiguous chunk
front &= recordThread->mRsmpInFramesP2 - 1;
size_t part1 = recordThread->mRsmpInFramesP2 - front;
if (part1 > (size_t) filled) {
part1 = filled;
}
size_t ask = buffer->frameCount;
ALOG_ASSERT(ask > 0);
if (part1 > ask) {
part1 = ask;
}
if (part1 == 0) {
// out of data is fine since the resampler will return a short-count.
buffer->raw = NULL;
buffer->frameCount = 0;
mRsmpInUnrel = 0;
return NOT_ENOUGH_DATA;
}
buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
buffer->frameCount = part1;
mRsmpInUnrel = part1;
return NO_ERROR;
}
// AudioBufferProvider interface
void ResamplerBufferProvider::releaseBuffer(
AudioBufferProvider::Buffer* buffer)
{
int32_t stepCount = static_cast<int32_t>(buffer->frameCount);
if (stepCount == 0) {
return;
}
ALOG_ASSERT(stepCount <= (int32_t)mRsmpInUnrel);
mRsmpInUnrel -= stepCount;
mRsmpInFront = audio_utils::safe_add_overflow(mRsmpInFront, stepCount);
buffer->raw = NULL;
buffer->frameCount = 0;
}
void RecordThread::checkBtNrec()
{
audio_utils::lock_guard _l(mutex());
checkBtNrec_l();
}
void RecordThread::checkBtNrec_l()
{
// disable AEC and NS if the device is a BT SCO headset supporting those
// pre processings
bool suspend = audio_is_bluetooth_sco_device(inDeviceType_l()) &&
mAfThreadCallback->btNrecIsOff();
if (mBtNrecSuspended.exchange(suspend) != suspend) {
for (size_t i = 0; i < mEffectChains.size(); i++) {
setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId());
setEffectSuspended_l(FX_IID_NS, suspend, mEffectChains[i]->sessionId());
}
}
}
bool RecordThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
status = NO_ERROR;
audio_format_t reqFormat = mFormat;
uint32_t samplingRate = mSampleRate;
// TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs).
[[maybe_unused]] audio_channel_mask_t channelMask =
audio_channel_in_mask_from_count(mChannelCount);
AudioParameter param = AudioParameter(keyValuePair);
int value;
// scope for AutoPark extends to end of method
AutoPark<FastCapture> park(mFastCapture);
// TODO Investigate when this code runs. Check with audio policy when a sample rate and
// channel count change can be requested. Do we mandate the first client defines the
// HAL sampling rate and channel count or do we allow changes on the fly?
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
samplingRate = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
if (!audio_is_linear_pcm((audio_format_t) value)) {
status = BAD_VALUE;
} else {
reqFormat = (audio_format_t) value;
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
audio_channel_mask_t mask = (audio_channel_mask_t) value;
if (!audio_is_input_channel(mask) ||
audio_channel_count_from_in_mask(mask) > FCC_LIMIT) {
status = BAD_VALUE;
} else {
channelMask = mask;
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (mActiveTracks.size() > 0) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
LOG_FATAL("Should not set routing device in RecordThread");
}
if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
mAudioSource != (audio_source_t)value) {
LOG_FATAL("Should not set audio source in RecordThread");
}
if (status == NO_ERROR) {
status = mInput->stream->setParameters(keyValuePair);
if (status == INVALID_OPERATION) {
inputStandBy();
status = mInput->stream->setParameters(keyValuePair);
}
if (reconfig) {
if (status == BAD_VALUE) {
audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
if (mInput->stream->getAudioProperties(&config) == OK &&
audio_is_linear_pcm(config.format) && audio_is_linear_pcm(reqFormat) &&
config.sample_rate <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate) &&
audio_channel_count_from_in_mask(config.channel_mask) <= FCC_LIMIT) {
status = NO_ERROR;
}
}
if (status == NO_ERROR) {
readInputParameters_l();
sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
}
}
}
return reconfig;
}
String8 RecordThread::getParameters(const String8& keys)
{
audio_utils::lock_guard _l(mutex());
if (initCheck() == NO_ERROR) {
String8 out_s8;
if (mInput->stream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
}
return {};
}
void RecordThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
sp<AudioIoDescriptor> desc;
switch (event) {
case AUDIO_INPUT_OPENED:
case AUDIO_INPUT_REGISTERED:
case AUDIO_INPUT_CONFIG_CHANGED:
desc = sp<AudioIoDescriptor>::make(mId, mPatch, true /*isInput*/,
mSampleRate, mFormat, mChannelMask, mFrameCount, mFrameCount);
break;
case AUDIO_CLIENT_STARTED:
desc = sp<AudioIoDescriptor>::make(mId, mPatch, portId);
break;
case AUDIO_INPUT_CLOSED:
default:
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
void RecordThread::readInputParameters_l()
{
const audio_config_base_t audioConfig = mInput->getAudioProperties();
mSampleRate = audioConfig.sample_rate;
mChannelMask = audioConfig.channel_mask;
if (!audio_is_input_channel(mChannelMask)) {
LOG_ALWAYS_FATAL("Channel mask %#x not valid for input", mChannelMask);
}
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
// Get actual HAL format.
status_t result = mInput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving input stream format: %d", result);
// Get format from the shim, which will be different than the HAL format
// if recording compressed audio from IEC61937 wrapped sources.
mFormat = audioConfig.format;
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("Format %#x not valid for input", mFormat);
}
if (audio_is_linear_pcm(mFormat)) {
LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_LIMIT, "HAL channel count %d > %d",
mChannelCount, FCC_LIMIT);
} else {
// Can have more that FCC_LIMIT channels in encoded streams.
ALOGI("HAL format %#x is not linear pcm", mFormat);
}
mFrameSize = mInput->getFrameSize();
LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
mFrameSize);
result = mInput->stream->getBufferSize(&mBufferSize);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
mFrameCount = mBufferSize / mFrameSize;
ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
"mBufferSize=%zu, mFrameCount=%zu",
this, mChannelCount, mFormat, mFrameSize, mBufferSize, mFrameCount);
// mRsmpInFrames must be 0 before calling resizeInputBuffer_l for the first time
mRsmpInFrames = 0;
resizeInputBuffer_l(0 /*maxSharedAudioHistoryMs*/);
// AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
// But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
audio_input_flags_t flags = mInput->flags;
mediametrics::LogItem item(mThreadMetrics.getMetricsId());
item.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_READPARAMETERS)
.set(AMEDIAMETRICS_PROP_ENCODING, IAfThreadBase::formatToString(mFormat).c_str())
.set(AMEDIAMETRICS_PROP_FLAGS, toString(flags).c_str())
.set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
.set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
.set(AMEDIAMETRICS_PROP_CHANNELCOUNT, (int32_t)mChannelCount)
.set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
.record();
}
uint32_t RecordThread::getInputFramesLost() const
{
audio_utils::lock_guard _l(mutex());
uint32_t result;
if (initCheck() == NO_ERROR && mInput->stream->getInputFramesLost(&result) == OK) {
return result;
}
return 0;
}
KeyedVector<audio_session_t, bool> RecordThread::sessionIds() const
{
KeyedVector<audio_session_t, bool> ids;
audio_utils::lock_guard _l(mutex());
for (size_t j = 0; j < mTracks.size(); ++j) {
sp<IAfRecordTrack> track = mTracks[j];
audio_session_t sessionId = track->sessionId();
if (ids.indexOfKey(sessionId) < 0) {
ids.add(sessionId, true);
}
}
return ids;
}
AudioStreamIn* RecordThread::clearInput()
{
audio_utils::lock_guard _l(mutex());
AudioStreamIn *input = mInput;
mInput = NULL;
mInputSource.clear();
return input;
}
// this method must always be called either with ThreadBase mutex() held or inside the thread loop
sp<StreamHalInterface> RecordThread::stream() const
{
if (mInput == NULL) {
return NULL;
}
return mInput->stream;
}
status_t RecordThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
chain->setThread(this);
chain->setInBuffer(NULL);
chain->setOutBuffer(NULL);
checkSuspendOnAddEffectChain_l(chain);
// make sure enabled pre processing effects state is communicated to the HAL as we
// just moved them to a new input stream.
chain->syncHalEffectsState_l();
mEffectChains.add(chain);
return NO_ERROR;
}
size_t RecordThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (chain == mEffectChains[i]) {
mEffectChains.removeAt(i);
break;
}
}
return mEffectChains.size();
}
status_t RecordThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
{
status_t status = NO_ERROR;
// store new device and send to effects
mInDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
mInDeviceTypeAddr.setAddress(patch->sources[0].ext.device.address);
audio_port_handle_t deviceId = patch->sources[0].id;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setInputDevice_l(inDeviceTypeAddr());
}
checkBtNrec_l();
// store new source and send to effects
if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
mAudioSource = patch->sinks[0].ext.mix.usecase.source;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setAudioSource_l(mAudioSource);
}
}
if (mInput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice();
status = hwDevice->createAudioPatch(patch->num_sources,
patch->sources,
patch->num_sinks,
patch->sinks,
handle);
} else {
status = mInput->stream->legacyCreateAudioPatch(patch->sources[0],
patch->sinks[0].ext.mix.usecase.source,
patch->sources[0].ext.device.type);
*handle = AUDIO_PATCH_HANDLE_NONE;
}
if ((mPatch.num_sources == 0) || (mPatch.sources[0].id != deviceId)) {
sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
mPatch = *patch;
}
const std::string pathSourcesAsString = patchSourcesToString(patch);
mThreadMetrics.logEndInterval();
mThreadMetrics.logCreatePatch(pathSourcesAsString, /* outDevices */ {});
mThreadMetrics.logBeginInterval();
// also dispatch to active AudioRecords
for (const auto &track : mActiveTracks) {
track->logEndInterval();
track->logBeginInterval(pathSourcesAsString);
}
// Force meteadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
status_t RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
mPatch = audio_patch{};
mInDeviceTypeAddr.reset();
if (mInput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice();
status = hwDevice->releaseAudioPatch(handle);
} else {
status = mInput->stream->legacyReleaseAudioPatch();
}
// Force meteadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
void RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
{
audio_utils::lock_guard _l(mutex());
mOutDevices = outDevices;
mOutDeviceTypeAddrs = deviceTypeAddrsFromDescriptors(mOutDevices);
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevices_l(outDeviceTypeAddrs());
}
}
int32_t RecordThread::getOldestFront_l()
{
if (mTracks.size() == 0) {
return mRsmpInRear;
}
int32_t oldestFront = mRsmpInRear;
int32_t maxFilled = 0;
for (size_t i = 0; i < mTracks.size(); i++) {
int32_t front = mTracks[i]->resamplerBufferProvider()->getFront();
int32_t filled;
(void)__builtin_sub_overflow(mRsmpInRear, front, &filled);
if (filled > maxFilled) {
oldestFront = front;
maxFilled = filled;
}
}
if (maxFilled > static_cast<signed>(mRsmpInFrames)) {
(void)__builtin_sub_overflow(mRsmpInRear, mRsmpInFrames, &oldestFront);
}
return oldestFront;
}
void RecordThread::updateFronts_l(int32_t offset)
{
if (offset == 0) {
return;
}
for (size_t i = 0; i < mTracks.size(); i++) {
int32_t front = mTracks[i]->resamplerBufferProvider()->getFront();
front = audio_utils::safe_sub_overflow(front, offset);
mTracks[i]->resamplerBufferProvider()->setFront(front);
}
}
void RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
{
// This is the formula for calculating the temporary buffer size.
// With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
// 1 full output buffer, regardless of the alignment of the available input.
// The value is somewhat arbitrary, and could probably be even larger.
// A larger value should allow more old data to be read after a track calls start(),
// without increasing latency.
//
// Note this is independent of the maximum downsampling ratio permitted for capture.
size_t minRsmpInFrames = mFrameCount * 7;
// maxSharedAudioHistoryMs != 0 indicates a request to possibly make some part of the audio
// capture history available to another client using the same session ID:
// dimension the resampler input buffer accordingly.
// Get oldest client read position: getOldestFront_l() must be called before altering
// mRsmpInRear, or mRsmpInFrames
int32_t previousFront = getOldestFront_l();
size_t previousRsmpInFramesP2 = mRsmpInFramesP2;
int32_t previousRear = mRsmpInRear;
mRsmpInRear = 0;
ALOG_ASSERT(maxSharedAudioHistoryMs >= 0
&& maxSharedAudioHistoryMs <= kMaxSharedAudioHistoryMs,
"resizeInputBuffer_l() called with invalid max shared history %d",
maxSharedAudioHistoryMs);
if (maxSharedAudioHistoryMs != 0) {
// resizeInputBuffer_l should never be called with a non zero shared history if the
// buffer was not already allocated
ALOG_ASSERT(mRsmpInBuffer != nullptr && mRsmpInFrames != 0,
"resizeInputBuffer_l() called with shared history and unallocated buffer");
size_t rsmpInFrames = (size_t)maxSharedAudioHistoryMs * mSampleRate / 1000;
// never reduce resampler input buffer size
if (rsmpInFrames <= mRsmpInFrames) {
return;
}
mRsmpInFrames = rsmpInFrames;
}
mMaxSharedAudioHistoryMs = maxSharedAudioHistoryMs;
// Note: mRsmpInFrames is 0 when called with maxSharedAudioHistoryMs equals to 0 so it is always
// initialized
if (mRsmpInFrames < minRsmpInFrames) {
mRsmpInFrames = minRsmpInFrames;
}
mRsmpInFramesP2 = roundup(mRsmpInFrames);
// TODO optimize audio capture buffer sizes ...
// Here we calculate the size of the sliding buffer used as a source
// for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7).
// For current HAL frame counts, this is usually 2048 = 40 ms. It would
// be better to have it derived from the pipe depth in the long term.
// The current value is higher than necessary. However it should not add to latency.
// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1;
void *rsmpInBuffer;
(void)posix_memalign(&rsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize);
// if posix_memalign fails, will segv here.
memset(rsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize);
// Copy audio history if any from old buffer before freeing it
if (previousRear != 0) {
ALOG_ASSERT(mRsmpInBuffer != nullptr,
"resizeInputBuffer_l() called with null buffer but frames already read from HAL");
ssize_t unread = audio_utils::safe_sub_overflow(previousRear, previousFront);
previousFront &= previousRsmpInFramesP2 - 1;
size_t part1 = previousRsmpInFramesP2 - previousFront;
if (part1 > (size_t) unread) {
part1 = unread;
}
if (part1 != 0) {
memcpy(rsmpInBuffer, (const uint8_t*)mRsmpInBuffer + previousFront * mFrameSize,
part1 * mFrameSize);
mRsmpInRear = part1;
part1 = unread - part1;
if (part1 != 0) {
memcpy((uint8_t*)rsmpInBuffer + mRsmpInRear * mFrameSize,
(const uint8_t*)mRsmpInBuffer, part1 * mFrameSize);
mRsmpInRear += part1;
}
}
// Update front for all clients according to new rear
updateFronts_l(audio_utils::safe_sub_overflow(previousRear, mRsmpInRear));
} else {
mRsmpInRear = 0;
}
free(mRsmpInBuffer);
mRsmpInBuffer = rsmpInBuffer;
}
void RecordThread::addPatchTrack(const sp<IAfPatchRecord>& record)
{
audio_utils::lock_guard _l(mutex());
mTracks.add(record);
if (record->getSource()) {
mSource = record->getSource();
}
}
void RecordThread::deletePatchTrack(const sp<IAfPatchRecord>& record)
{
audio_utils::lock_guard _l(mutex());
if (mSource == record->getSource()) {
mSource = mInput;
}
destroyTrack_l(record);
}
void RecordThread::toAudioPortConfig(struct audio_port_config* config)
{
ThreadBase::toAudioPortConfig(config);
config->role = AUDIO_PORT_ROLE_SINK;
config->ext.mix.hw_module = mInput->audioHwDev->handle();
config->ext.mix.usecase.source = mAudioSource;
if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
config->flags.input = mInput->flags;
}
}
// ----------------------------------------------------------------------------
// Mmap
// ----------------------------------------------------------------------------
// Mmap stream control interface implementation. Each MmapThreadHandle controls one
// MmapPlaybackThread or MmapCaptureThread instance.
class MmapThreadHandle : public MmapStreamInterface {
public:
explicit MmapThreadHandle(const sp<IAfMmapThread>& thread);
~MmapThreadHandle() override;
// MmapStreamInterface virtuals
status_t createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info* info) final;
status_t getMmapPosition(struct audio_mmap_position* position) final;
status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) final;
status_t start(const AudioClient& client,
const audio_attributes_t* attr, audio_port_handle_t* handle) final;
status_t stop(audio_port_handle_t handle) final;
status_t standby() final;
status_t reportData(const void* buffer, size_t frameCount) final;
private:
const sp<IAfMmapThread> mThread;
};
/* static */
sp<MmapStreamInterface> IAfMmapThread::createMmapStreamInterfaceAdapter(
const sp<IAfMmapThread>& mmapThread) {
return sp<MmapThreadHandle>::make(mmapThread);
}
MmapThreadHandle::MmapThreadHandle(const sp<IAfMmapThread>& thread)
: mThread(thread)
{
assert(thread != 0); // thread must start non-null and stay non-null
}
// MmapStreamInterface could be directly implemented by MmapThread excepting this
// special handling on adapter dtor.
MmapThreadHandle::~MmapThreadHandle()
{
mThread->disconnect();
}
status_t MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info)
{
return mThread->createMmapBuffer(minSizeFrames, info);
}
status_t MmapThreadHandle::getMmapPosition(struct audio_mmap_position* position)
{
return mThread->getMmapPosition(position);
}
status_t MmapThreadHandle::getExternalPosition(uint64_t* position,
int64_t *timeNanos) {
return mThread->getExternalPosition(position, timeNanos);
}
status_t MmapThreadHandle::start(const AudioClient& client,
const audio_attributes_t *attr, audio_port_handle_t *handle)
{
return mThread->start(client, attr, handle);
}
status_t MmapThreadHandle::stop(audio_port_handle_t handle)
{
return mThread->stop(handle);
}
status_t MmapThreadHandle::standby()
{
return mThread->standby();
}
status_t MmapThreadHandle::reportData(const void* buffer, size_t frameCount)
{
return mThread->reportData(buffer, frameCount);
}
MmapThread::MmapThread(
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady, bool isOut)
: ThreadBase(afThreadCallback, id, (isOut ? MMAP_PLAYBACK : MMAP_CAPTURE), systemReady, isOut),
mSessionId(AUDIO_SESSION_NONE),
mPortId(AUDIO_PORT_HANDLE_NONE),
mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev),
mActiveTracks(&this->mLocalLog),
mHalVolFloat(-1.0f), // Initialize to illegal value so it always gets set properly later.
mNoCallbackWarningCount(0)
{
mStandby = true;
readHalParameters_l();
}
void MmapThread::onFirstRef()
{
run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}
void MmapThread::disconnect()
{
ActiveTracks<IAfMmapTrack> activeTracks;
audio_port_handle_t localPortId;
{
audio_utils::lock_guard _l(mutex());
for (const sp<IAfMmapTrack>& t : mActiveTracks) {
activeTracks.add(t);
}
localPortId = mPortId;
}
for (const sp<IAfMmapTrack>& t : activeTracks) {
stop(t->portId());
}
// This will decrement references and may cause the destruction of this thread.
if (isOutput()) {
AudioSystem::releaseOutput(localPortId);
} else {
AudioSystem::releaseInput(localPortId);
}
}
void MmapThread::configure_l(const audio_attributes_t* attr,
audio_stream_type_t streamType __unused,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
audio_port_handle_t portId)
{
mAttr = *attr;
mSessionId = sessionId;
mCallback = callback;
mDeviceId = deviceId;
mPortId = portId;
}
status_t MmapThread::createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info)
{
audio_utils::lock_guard l(mutex());
if (mHalStream == 0) {
return NO_INIT;
}
mStandby = true;
return mHalStream->createMmapBuffer(minSizeFrames, info);
}
status_t MmapThread::getMmapPosition(struct audio_mmap_position* position) const
{
audio_utils::lock_guard l(mutex());
if (mHalStream == 0) {
return NO_INIT;
}
return mHalStream->getMmapPosition(position);
}
status_t MmapThread::exitStandby_l()
{
// The HAL must receive track metadata before starting the stream
updateMetadata_l();
status_t ret = mHalStream->start();
if (ret != NO_ERROR) {
ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret);
return ret;
}
if (mStandby) {
mThreadMetrics.logBeginInterval();
mThreadSnapshot.onBegin();
mStandby = false;
}
return NO_ERROR;
}
status_t MmapThread::start(const AudioClient& client,
const audio_attributes_t *attr,
audio_port_handle_t *handle)
{
audio_utils::lock_guard l(mutex());
ALOGV("%s clientUid %d mStandby %d mPortId %d *handle %d", __FUNCTION__,
client.attributionSource.uid, mStandby, mPortId, *handle);
if (mHalStream == 0) {
return NO_INIT;
}
status_t ret;
// For the first track, reuse portId and session allocated when the stream was opened.
if (*handle == mPortId) {
acquireWakeLock_l();
return NO_ERROR;
}
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
audio_io_handle_t io = mId;
const AttributionSourceState adjAttributionSource = afutils::checkAttributionSourcePackage(
client.attributionSource);
const auto localSessionId = mSessionId;
auto localAttr = mAttr;
if (isOutput()) {
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = mSampleRate;
config.channel_mask = mChannelMask;
config.format = mFormat;
audio_stream_type_t stream = streamType_l();
audio_output_flags_t flags =
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT);
audio_port_handle_t deviceId = mDeviceId;
std::vector<audio_io_handle_t> secondaryOutputs;
bool isSpatialized;
bool isBitPerfect;
mutex().unlock();
ret = AudioSystem::getOutputForAttr(&localAttr, &io,
localSessionId,
&stream,
adjAttributionSource,
&config,
flags,
&deviceId,
&portId,
&secondaryOutputs,
&isSpatialized,
&isBitPerfect);
mutex().lock();
mAttr = localAttr;
ALOGD_IF(!secondaryOutputs.empty(),
"MmapThread::start does not support secondary outputs, ignoring them");
} else {
audio_config_base_t config;
config.sample_rate = mSampleRate;
config.channel_mask = mChannelMask;
config.format = mFormat;
audio_port_handle_t deviceId = mDeviceId;
mutex().unlock();
ret = AudioSystem::getInputForAttr(&localAttr, &io,
RECORD_RIID_INVALID,
localSessionId,
adjAttributionSource,
&config,
AUDIO_INPUT_FLAG_MMAP_NOIRQ,
&deviceId,
&portId);
mutex().lock();
// localAttr is const for getInputForAttr.
}
// APM should not chose a different input or output stream for the same set of attributes
// and audo configuration
if (ret != NO_ERROR || io != mId) {
ALOGE("%s: error getting output or input from APM (error %d, io %d expected io %d)",
__FUNCTION__, ret, io, mId);
return BAD_VALUE;
}
if (isOutput()) {
mutex().unlock();
ret = AudioSystem::startOutput(portId);
mutex().lock();
} else {
{
// Add the track record before starting input so that the silent status for the
// client can be cached.
setClientSilencedState_l(portId, false /*silenced*/);
}
mutex().unlock();
ret = AudioSystem::startInput(portId);
mutex().lock();
}
// abort if start is rejected by audio policy manager
if (ret != NO_ERROR) {
ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret);
if (!mActiveTracks.isEmpty()) {
mutex().unlock();
if (isOutput()) {
AudioSystem::releaseOutput(portId);
} else {
AudioSystem::releaseInput(portId);
}
mutex().lock();
} else {
mHalStream->stop();
}
eraseClientSilencedState_l(portId);
return PERMISSION_DENIED;
}
// Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ?
sp<IAfMmapTrack> track = IAfMmapTrack::create(
this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
mChannelMask, mSessionId, isOutput(),
client.attributionSource,
IPCThreadState::self()->getCallingPid(), portId);
if (!isOutput()) {
track->setSilenced_l(isClientSilenced_l(portId));
}
if (isOutput()) {
// force volume update when a new track is added
mHalVolFloat = -1.0f;
} else if (!track->isSilenced_l()) {
for (const sp<IAfMmapTrack>& t : mActiveTracks) {
if (t->isSilenced_l()
&& t->uid() != static_cast<uid_t>(client.attributionSource.uid)) {
t->invalidate();
}
}
}
mActiveTracks.add(track);
sp<IAfEffectChain> chain = getEffectChain_l(mSessionId);
if (chain != 0) {
chain->setStrategy(getStrategyForStream(streamType_l()));
chain->incTrackCnt();
chain->incActiveTrackCnt();
}
track->logBeginInterval(patchSinksToString(&mPatch)); // log to MediaMetrics
*handle = portId;
if (mActiveTracks.size() == 1) {
ret = exitStandby_l();
}
broadcast_l();
ALOGV("%s DONE status %d handle %d stream %p", __FUNCTION__, ret, *handle, mHalStream.get());
return ret;
}
status_t MmapThread::stop(audio_port_handle_t handle)
{
ALOGV("%s handle %d", __FUNCTION__, handle);
audio_utils::lock_guard l(mutex());
if (mHalStream == 0) {
return NO_INIT;
}
if (handle == mPortId) {
releaseWakeLock_l();
return NO_ERROR;
}
sp<IAfMmapTrack> track;
for (const sp<IAfMmapTrack>& t : mActiveTracks) {
if (handle == t->portId()) {
track = t;
break;
}
}
if (track == 0) {
return BAD_VALUE;
}
mActiveTracks.remove(track);
eraseClientSilencedState_l(track->portId());
mutex().unlock();
if (isOutput()) {
AudioSystem::stopOutput(track->portId());
AudioSystem::releaseOutput(track->portId());
} else {
AudioSystem::stopInput(track->portId());
AudioSystem::releaseInput(track->portId());
}
mutex().lock();
sp<IAfEffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->decActiveTrackCnt();
chain->decTrackCnt();
}
if (mActiveTracks.isEmpty()) {
mHalStream->stop();
}
broadcast_l();
return NO_ERROR;
}
status_t MmapThread::standby()
NO_THREAD_SAFETY_ANALYSIS // clang bug
{
ALOGV("%s", __FUNCTION__);
audio_utils::lock_guard l_{mutex()};
if (mHalStream == 0) {
return NO_INIT;
}
if (!mActiveTracks.isEmpty()) {
return INVALID_OPERATION;
}
mHalStream->standby();
if (!mStandby) {
mThreadMetrics.logEndInterval();
mThreadSnapshot.onEnd();
mStandby = true;
}
releaseWakeLock_l();
return NO_ERROR;
}
status_t MmapThread::reportData(const void* /*buffer*/, size_t /*frameCount*/) {
// This is a stub implementation. The MmapPlaybackThread overrides this function.
return INVALID_OPERATION;
}
void MmapThread::readHalParameters_l()
{
status_t result = mHalStream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
mFormat = mHALFormat;
LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat);
result = mHalStream->getFrameSize(&mFrameSize);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
mFrameSize);
result = mHalStream->getBufferSize(&mBufferSize);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
mFrameCount = mBufferSize / mFrameSize;
// TODO: make a readHalParameters call?
mediametrics::LogItem item(mThreadMetrics.getMetricsId());
item.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_READPARAMETERS)
.set(AMEDIAMETRICS_PROP_ENCODING, IAfThreadBase::formatToString(mFormat).c_str())
.set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
.set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
.set(AMEDIAMETRICS_PROP_CHANNELCOUNT, (int32_t)mChannelCount)
.set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
/*
.set(AMEDIAMETRICS_PROP_FLAGS, toString(flags).c_str())
.set(AMEDIAMETRICS_PROP_PREFIX_HAPTIC AMEDIAMETRICS_PROP_CHANNELMASK,
(int32_t)mHapticChannelMask)
.set(AMEDIAMETRICS_PROP_PREFIX_HAPTIC AMEDIAMETRICS_PROP_CHANNELCOUNT,
(int32_t)mHapticChannelCount)
*/
.set(AMEDIAMETRICS_PROP_PREFIX_HAL AMEDIAMETRICS_PROP_ENCODING,
IAfThreadBase::formatToString(mHALFormat).c_str())
.set(AMEDIAMETRICS_PROP_PREFIX_HAL AMEDIAMETRICS_PROP_FRAMECOUNT,
(int32_t)mFrameCount) // sic - added HAL
.record();
}
bool MmapThread::threadLoop()
{
{
audio_utils::unique_lock _l(mutex());
checkSilentMode_l();
}
const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
while (!exitPending())
{
Vector<sp<IAfEffectChain>> effectChains;
{ // under Thread lock
audio_utils::unique_lock _l(mutex());
if (mSignalPending) {
// A signal was raised while we were unlocked
mSignalPending = false;
} else {
if (mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) {
break;
}
// wait until we have something to do...
ALOGV("%s going to sleep", myName.c_str());
mWaitWorkCV.wait(_l);
ALOGV("%s waking up", myName.c_str());
checkSilentMode_l();
continue;
}
}
processConfigEvents_l();
processVolume_l();
checkInvalidTracks_l();
mActiveTracks.updatePowerState_l(this);
updateMetadata_l();
lockEffectChains_l(effectChains);
} // release Thread lock
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l(); // Thread is not locked, but effect chain is locked
}
// enable changes in effect chain, including moving to another thread.
unlockEffectChains(effectChains);
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
}
threadLoop_exit();
if (!mStandby) {
threadLoop_standby();
mStandby = true;
}
ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
// checkForNewParameter_l() must be called with ThreadBase::mutex() held
bool MmapThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
AudioParameter param = AudioParameter(keyValuePair);
int value;
bool sendToHal = true;
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
LOG_FATAL("Should not happen set routing device in MmapThread");
}
if (sendToHal) {
status = mHalStream->setParameters(keyValuePair);
} else {
status = NO_ERROR;
}
return false;
}
String8 MmapThread::getParameters(const String8& keys)
{
audio_utils::lock_guard _l(mutex());
String8 out_s8;
if (initCheck() == NO_ERROR && mHalStream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
return {};
}
void MmapThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId __unused) {
sp<AudioIoDescriptor> desc;
bool isInput = false;
switch (event) {
case AUDIO_INPUT_OPENED:
case AUDIO_INPUT_REGISTERED:
case AUDIO_INPUT_CONFIG_CHANGED:
isInput = true;
FALLTHROUGH_INTENDED;
case AUDIO_OUTPUT_OPENED:
case AUDIO_OUTPUT_REGISTERED:
case AUDIO_OUTPUT_CONFIG_CHANGED:
desc = sp<AudioIoDescriptor>::make(mId, mPatch, isInput,
mSampleRate, mFormat, mChannelMask, mFrameCount, mFrameCount);
break;
case AUDIO_INPUT_CLOSED:
case AUDIO_OUTPUT_CLOSED:
default:
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
status_t MmapThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
NO_THREAD_SAFETY_ANALYSIS // elease and re-acquire mutex()
{
status_t status = NO_ERROR;
// store new device and send to effects
audio_devices_t type = AUDIO_DEVICE_NONE;
audio_port_handle_t deviceId;
AudioDeviceTypeAddrVector sinkDeviceTypeAddrs;
AudioDeviceTypeAddr sourceDeviceTypeAddr;
uint32_t numDevices = 0;
if (isOutput()) {
for (unsigned int i = 0; i < patch->num_sinks; i++) {
LOG_ALWAYS_FATAL_IF(popcount(patch->sinks[i].ext.device.type) > 1
&& !mAudioHwDev->supportsAudioPatches(),
"Enumerated device type(%#x) must not be used "
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
sinkDeviceTypeAddrs.emplace_back(patch->sinks[i].ext.device.type,
patch->sinks[i].ext.device.address);
}
deviceId = patch->sinks[0].id;
numDevices = mPatch.num_sinks;
} else {
type = patch->sources[0].ext.device.type;
deviceId = patch->sources[0].id;
numDevices = mPatch.num_sources;
sourceDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
sourceDeviceTypeAddr.setAddress(patch->sources[0].ext.device.address);
}
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (isOutput()) {
mEffectChains[i]->setDevices_l(sinkDeviceTypeAddrs);
} else {
mEffectChains[i]->setInputDevice_l(sourceDeviceTypeAddr);
}
}
if (!isOutput()) {
// store new source and send to effects
if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
mAudioSource = patch->sinks[0].ext.mix.usecase.source;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setAudioSource_l(mAudioSource);
}
}
}
// For mmap streams, once the routing has changed, they will be disconnected. It should be
// okay to notify the client earlier before the new patch creation.
if (mDeviceId != deviceId) {
if (const sp<MmapStreamCallback> callback = mCallback.promote()) {
// The aaudioservice handle the routing changed event asynchronously. In that case,
// it is safe to hold the lock here.
callback->onRoutingChanged(deviceId);
}
}
if (mAudioHwDev->supportsAudioPatches()) {
status = mHalDevice->createAudioPatch(patch->num_sources, patch->sources, patch->num_sinks,
patch->sinks, handle);
} else {
audio_port_config port;
std::optional<audio_source_t> source;
if (isOutput()) {
port = patch->sinks[0];
} else {
port = patch->sources[0];
source = patch->sinks[0].ext.mix.usecase.source;
}
status = mHalStream->legacyCreateAudioPatch(port, source, type);
*handle = AUDIO_PATCH_HANDLE_NONE;
}
if (numDevices == 0 || mDeviceId != deviceId) {
if (isOutput()) {
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
mOutDeviceTypeAddrs = sinkDeviceTypeAddrs;
checkSilentMode_l();
} else {
sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
mInDeviceTypeAddr = sourceDeviceTypeAddr;
}
mPatch = *patch;
mDeviceId = deviceId;
}
// Force meteadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
status_t MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
mPatch = audio_patch{};
mOutDeviceTypeAddrs.clear();
mInDeviceTypeAddr.reset();
bool supportsAudioPatches = mHalDevice->supportsAudioPatches(&supportsAudioPatches) == OK ?
supportsAudioPatches : false;
if (supportsAudioPatches) {
status = mHalDevice->releaseAudioPatch(handle);
} else {
status = mHalStream->legacyReleaseAudioPatch();
}
// Force meteadata update after a route change
mActiveTracks.setHasChanged();
return status;
}
void MmapThread::toAudioPortConfig(struct audio_port_config* config)
NO_THREAD_SAFETY_ANALYSIS // mAudioHwDev handle access
{
ThreadBase::toAudioPortConfig(config);
if (isOutput()) {
config->role = AUDIO_PORT_ROLE_SOURCE;
config->ext.mix.hw_module = mAudioHwDev->handle();
config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
} else {
config->role = AUDIO_PORT_ROLE_SINK;
config->ext.mix.hw_module = mAudioHwDev->handle();
config->ext.mix.usecase.source = mAudioSource;
}
}
status_t MmapThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
// Attach all tracks with same session ID to this chain.
// indicate all active tracks in the chain
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
if (session == track->sessionId()) {
chain->incTrackCnt();
chain->incActiveTrackCnt();
}
}
chain->setThread(this);
chain->setInBuffer(nullptr);
chain->setOutBuffer(nullptr);
chain->syncHalEffectsState_l();
mEffectChains.add(chain);
checkSuspendOnAddEffectChain_l(chain);
return NO_ERROR;
}
size_t MmapThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (chain == mEffectChains[i]) {
mEffectChains.removeAt(i);
// detach all active tracks from the chain
// detach all tracks with same session ID from this chain
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
if (session == track->sessionId()) {
chain->decActiveTrackCnt();
chain->decTrackCnt();
}
}
break;
}
}
return mEffectChains.size();
}
void MmapThread::threadLoop_standby()
{
mHalStream->standby();
}
void MmapThread::threadLoop_exit()
{
// Do not call callback->onTearDown() because it is redundant for thread exit
// and because it can cause a recursive mutex lock on stop().
}
status_t MmapThread::setSyncEvent(const sp<SyncEvent>& /* event */)
{
return BAD_VALUE;
}
bool MmapThread::isValidSyncEvent(
const sp<SyncEvent>& /* event */) const
{
return false;
}
status_t MmapThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global effect sessions on mmap threads
if (audio_is_global_session(sessionId)) {
ALOGW("checkEffectCompatibility_l(): global effect %s on MMAP thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
if (!isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC)) {
ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on capture mmap thread",
desc->name);
return BAD_VALUE;
}
if (isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap "
"thread", desc->name);
return BAD_VALUE;
}
// Only allow effects without processing load or latency
if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) != EFFECT_FLAG_NO_PROCESS) {
return BAD_VALUE;
}
if (IAfEffectModule::isHapticGenerator(&desc->type)) {
ALOGE("%s(): HapticGenerator is not supported for MmapThread", __func__);
return BAD_VALUE;
}
return NO_ERROR;
}
void MmapThread::checkInvalidTracks_l()
{
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
if (track->isInvalid()) {
if (const sp<MmapStreamCallback> callback = mCallback.promote()) {
// The aaudioservice handle the routing changed event asynchronously. In that case,
// it is safe to hold the lock here.
callback->onRoutingChanged(AUDIO_PORT_HANDLE_NONE);
} else if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
ALOGW("Could not notify MMAP stream tear down: no onRoutingChanged callback!");
mNoCallbackWarningCount++;
}
break;
}
}
}
void MmapThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
{
dprintf(fd, " Attributes: content type %d usage %d source %d\n",
mAttr.content_type, mAttr.usage, mAttr.source);
dprintf(fd, " Session: %d port Id: %d\n", mSessionId, mPortId);
if (mActiveTracks.isEmpty()) {
dprintf(fd, " No active clients\n");
}
}
void MmapThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
size_t numtracks = mActiveTracks.size();
dprintf(fd, " %zu Tracks\n", numtracks);
const char *prefix = " ";
if (numtracks) {
result.append(prefix);
mActiveTracks[0]->appendDumpHeader(result);
for (size_t i = 0; i < numtracks ; ++i) {
sp<IAfMmapTrack> track = mActiveTracks[i];
result.append(prefix);
track->appendDump(result, true /* active */);
}
} else {
dprintf(fd, "\n");
}
write(fd, result.c_str(), result.size());
}
/* static */
sp<IAfMmapPlaybackThread> IAfMmapPlaybackThread::create(
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady) {
return sp<MmapPlaybackThread>::make(afThreadCallback, id, hwDev, output, systemReady);
}
MmapPlaybackThread::MmapPlaybackThread(
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady)
: MmapThread(afThreadCallback, id, hwDev, output->stream, systemReady, true /* isOut */),
mStreamType(AUDIO_STREAM_MUSIC),
mOutput(output)
{
snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id);
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mMasterVolume = afThreadCallback->masterVolume_l();
mMasterMute = afThreadCallback->masterMute_l();
for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
mStreamTypes[stream].volume = 0.0f;
mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
}
// Audio patch and call assistant volume are always max
mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
if (mAudioHwDev) {
if (mAudioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
}
if (mAudioHwDev->canSetMasterMute()) {
mMasterMute = false;
}
}
}
void MmapPlaybackThread::configure(const audio_attributes_t* attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
audio_port_handle_t portId)
{
audio_utils::lock_guard l(mutex());
MmapThread::configure_l(attr, streamType, sessionId, callback, deviceId, portId);
mStreamType = streamType;
}
AudioStreamOut* MmapPlaybackThread::clearOutput()
{
audio_utils::lock_guard _l(mutex());
AudioStreamOut *output = mOutput;
mOutput = NULL;
return output;
}
void MmapPlaybackThread::setMasterVolume(float value)
{
audio_utils::lock_guard _l(mutex());
// Don't apply master volume in SW if our HAL can do it for us.
if (mAudioHwDev &&
mAudioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
} else {
mMasterVolume = value;
}
}
void MmapPlaybackThread::setMasterMute(bool muted)
{
audio_utils::lock_guard _l(mutex());
// Don't apply master mute in SW if our HAL can do it for us.
if (mAudioHwDev && mAudioHwDev->canSetMasterMute()) {
mMasterMute = false;
} else {
mMasterMute = muted;
}
}
void MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
audio_utils::lock_guard _l(mutex());
mStreamTypes[stream].volume = value;
if (stream == mStreamType) {
broadcast_l();
}
}
float MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
{
audio_utils::lock_guard _l(mutex());
return mStreamTypes[stream].volume;
}
void MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
audio_utils::lock_guard _l(mutex());
mStreamTypes[stream].mute = muted;
if (stream == mStreamType) {
broadcast_l();
}
}
void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
audio_utils::lock_guard _l(mutex());
if (streamType == mStreamType) {
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
track->invalidate();
}
broadcast_l();
}
}
void MmapPlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds)
{
audio_utils::lock_guard _l(mutex());
bool trackMatch = false;
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
if (portIds.find(track->portId()) != portIds.end()) {
track->invalidate();
trackMatch = true;
portIds.erase(track->portId());
}
if (portIds.empty()) {
break;
}
}
if (trackMatch) {
broadcast_l();
}
}
void MmapPlaybackThread::processVolume_l()
NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
{
float volume;
if (mMasterMute || streamMuted_l()) {
volume = 0;
} else {
volume = mMasterVolume * streamVolume_l();
}
if (volume != mHalVolFloat) {
// Convert volumes from float to 8.24
uint32_t vol = (uint32_t)(volume * (1 << 24));
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!mEffectChains.isEmpty()) {
mEffectChains[0]->setVolume(&vol, &vol);
volume = (float)vol / (1 << 24);
}
// Try to use HW volume control and fall back to SW control if not implemented
if (mOutput->stream->setVolume(volume, volume) == NO_ERROR) {
mHalVolFloat = volume; // HW volume control worked, so update value.
mNoCallbackWarningCount = 0;
} else {
sp<MmapStreamCallback> callback = mCallback.promote();
if (callback != 0) {
mHalVolFloat = volume; // SW volume control worked, so update value.
mNoCallbackWarningCount = 0;
mutex().unlock();
callback->onVolumeChanged(volume);
mutex().lock();
} else {
if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
ALOGW("Could not set MMAP stream volume: no volume callback!");
mNoCallbackWarningCount++;
}
}
}
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
track->setMetadataHasChanged();
track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
/*muteState=*/{mMasterMute,
streamVolume_l() == 0.f,
streamMuted_l(),
// TODO(b/241533526): adjust logic to include mute from AppOps
false /*muteFromPlaybackRestricted*/,
false /*muteFromClientVolume*/,
false /*muteFromVolumeShaper*/});
}
}
}
ThreadBase::MetadataUpdate MmapPlaybackThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
}
StreamOutHalInterface::SourceMetadata metadata;
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
// No track is invalid as this is called after prepareTrack_l in the same critical section
playback_track_metadata_v7_t trackMetadata;
trackMetadata.base = {
.usage = track->attributes().usage,
.content_type = track->attributes().content_type,
.gain = mHalVolFloat, // TODO: propagate from aaudio pre-mix volume
};
trackMetadata.channel_mask = track->channelMask(),
strncpy(trackMetadata.tags, track->attributes().tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
metadata.tracks.push_back(trackMetadata);
}
mOutput->stream->updateSourceMetadata(metadata);
MetadataUpdate change;
change.playbackMetadataUpdate = metadata.tracks;
return change;
};
void MmapPlaybackThread::checkSilentMode_l()
{
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
if (property_get("ro.audio.silent", value, "0") > 0) {
char *endptr;
unsigned long ul = strtoul(value, &endptr, 0);
if (*endptr == '\0' && ul != 0) {
ALOGW("%s: mute from ro.audio.silent. Silence is golden", __func__);
// The setprop command will not allow a property to be changed after
// the first time it is set, so we don't have to worry about un-muting.
setMasterMute_l(true);
}
}
}
}
void MmapPlaybackThread::toAudioPortConfig(struct audio_port_config* config)
{
MmapThread::toAudioPortConfig(config);
if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
config->flags.output = mOutput->flags;
}
}
status_t MmapPlaybackThread::getExternalPosition(uint64_t* position,
int64_t* timeNanos) const
{
if (mOutput == nullptr) {
return NO_INIT;
}
struct timespec timestamp;
status_t status = mOutput->getPresentationPosition(position, &timestamp);
if (status == NO_ERROR) {
*timeNanos = timestamp.tv_sec * NANOS_PER_SECOND + timestamp.tv_nsec;
}
return status;
}
status_t MmapPlaybackThread::reportData(const void* buffer, size_t frameCount) {
// Send to MelProcessor for sound dose measurement.
auto processor = mMelProcessor.load();
if (processor) {
processor->process(buffer, frameCount * mFrameSize);
}
return NO_ERROR;
}
// startMelComputation_l() must be called with AudioFlinger::mutex() held
void MmapPlaybackThread::startMelComputation_l(
const sp<audio_utils::MelProcessor>& processor)
{
ALOGV("%s: starting mel processor for thread %d", __func__, id());
mMelProcessor.store(processor);
if (processor) {
processor->resume();
}
// no need to update output format for MMapPlaybackThread since it is
// assigned constant for each thread
}
// stopMelComputation_l() must be called with AudioFlinger::mutex() held
void MmapPlaybackThread::stopMelComputation_l()
{
ALOGV("%s: pausing mel processor for thread %d", __func__, id());
auto melProcessor = mMelProcessor.load();
if (melProcessor != nullptr) {
melProcessor->pause();
}
}
void MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MmapThread::dumpInternals_l(fd, args);
dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
}
/* static */
sp<IAfMmapCaptureThread> IAfMmapCaptureThread::create(
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice* hwDev, AudioStreamIn* input, bool systemReady) {
return sp<MmapCaptureThread>::make(afThreadCallback, id, hwDev, input, systemReady);
}
MmapCaptureThread::MmapCaptureThread(
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady)
: MmapThread(afThreadCallback, id, hwDev, input->stream, systemReady, false /* isOut */),
mInput(input)
{
snprintf(mThreadName, kThreadNameLength, "AudioMmapIn_%X", id);
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
}
status_t MmapCaptureThread::exitStandby_l()
{
{
// mInput might have been cleared by clearInput()
if (mInput != nullptr && mInput->stream != nullptr) {
mInput->stream->setGain(1.0f);
}
}
return MmapThread::exitStandby_l();
}
AudioStreamIn* MmapCaptureThread::clearInput()
{
audio_utils::lock_guard _l(mutex());
AudioStreamIn *input = mInput;
mInput = NULL;
return input;
}
void MmapCaptureThread::processVolume_l()
{
bool changed = false;
bool silenced = false;
sp<MmapStreamCallback> callback = mCallback.promote();
if (callback == 0) {
if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
ALOGW("Could not set MMAP stream silenced: no onStreamSilenced callback!");
mNoCallbackWarningCount++;
}
}
// After a change occurred in track silenced state, mute capture in audio DSP if at least one
// track is silenced and unmute otherwise
for (size_t i = 0; i < mActiveTracks.size() && !silenced; i++) {
if (!mActiveTracks[i]->getAndSetSilencedNotified_l()) {
changed = true;
silenced = mActiveTracks[i]->isSilenced_l();
}
}
if (changed) {
mInput->stream->setGain(silenced ? 0.0f: 1.0f);
}
}
ThreadBase::MetadataUpdate MmapCaptureThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
}
StreamInHalInterface::SinkMetadata metadata;
for (const sp<IAfMmapTrack>& track : mActiveTracks) {
// No track is invalid as this is called after prepareTrack_l in the same critical section
record_track_metadata_v7_t trackMetadata;
trackMetadata.base = {
.source = track->attributes().source,
.gain = 1, // capture tracks do not have volumes
};
trackMetadata.channel_mask = track->channelMask(),
strncpy(trackMetadata.tags, track->attributes().tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
metadata.tracks.push_back(trackMetadata);
}
mInput->stream->updateSinkMetadata(metadata);
MetadataUpdate change;
change.recordMetadataUpdate = metadata.tracks;
return change;
}
void MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
{
audio_utils::lock_guard _l(mutex());
for (size_t i = 0; i < mActiveTracks.size() ; i++) {
if (mActiveTracks[i]->portId() == portId) {
mActiveTracks[i]->setSilenced_l(silenced);
broadcast_l();
}
}
setClientSilencedIfExists_l(portId, silenced);
}
void MmapCaptureThread::toAudioPortConfig(struct audio_port_config* config)
{
MmapThread::toAudioPortConfig(config);
if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
config->flags.input = mInput->flags;
}
}
status_t MmapCaptureThread::getExternalPosition(
uint64_t* position, int64_t* timeNanos) const
{
if (mInput == nullptr) {
return NO_INIT;
}
return mInput->getCapturePosition((int64_t*)position, timeNanos);
}
// ----------------------------------------------------------------------------
/* static */
sp<IAfPlaybackThread> IAfPlaybackThread::createBitPerfectThread(
const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output, audio_io_handle_t id, bool systemReady) {
return sp<BitPerfectThread>::make(afThreadCallback, output, id, systemReady);
}
BitPerfectThread::BitPerfectThread(const sp<IAfThreadCallback> &afThreadCallback,
AudioStreamOut *output, audio_io_handle_t id, bool systemReady)
: MixerThread(afThreadCallback, output, id, systemReady, BIT_PERFECT) {}
PlaybackThread::mixer_state BitPerfectThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove) {
mixer_state result = MixerThread::prepareTracks_l(tracksToRemove);
// If there is only one active track and it is bit-perfect, enable tee buffer.
float volumeLeft = 1.0f;
float volumeRight = 1.0f;
if (mActiveTracks.size() == 1 && mActiveTracks[0]->isBitPerfect()) {
const int trackId = mActiveTracks[0]->id();
mAudioMixer->setParameter(
trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER, (void *)mSinkBuffer);
mAudioMixer->setParameter(
trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER_FRAME_COUNT,
(void *)(uintptr_t)mNormalFrameCount);
mActiveTracks[0]->getFinalVolume(&volumeLeft, &volumeRight);
mIsBitPerfect = true;
} else {
mIsBitPerfect = false;
// No need to copy bit-perfect data directly to sink buffer given there are multiple tracks
// active.
for (const auto& track : mActiveTracks) {
const int trackId = track->id();
mAudioMixer->setParameter(
trackId, AudioMixer::TRACK, AudioMixer::TEE_BUFFER, nullptr);
}
}
if (mVolumeLeft != volumeLeft || mVolumeRight != volumeRight) {
mVolumeLeft = volumeLeft;
mVolumeRight = volumeRight;
setVolumeForOutput_l(volumeLeft, volumeRight);
}
return result;
}
void BitPerfectThread::threadLoop_mix() {
MixerThread::threadLoop_mix();
mHasDataCopiedToSinkBuffer = mIsBitPerfect;
}
} // namespace android