| /* |
| ** |
| ** Copyright 2007, 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 "AudioMixer" |
| //#define LOG_NDEBUG 0 |
| |
| #include <sstream> |
| #include <stdint.h> |
| #include <string.h> |
| #include <stdlib.h> |
| #include <math.h> |
| #include <sys/types.h> |
| |
| #include <utils/Errors.h> |
| #include <utils/Log.h> |
| |
| #include <system/audio.h> |
| |
| #include <audio_utils/primitives.h> |
| #include <audio_utils/format.h> |
| #include <media/AudioMixer.h> |
| |
| #include "AudioMixerOps.h" |
| |
| // The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer. |
| #ifndef FCC_2 |
| #define FCC_2 2 |
| #endif |
| |
| // Look for MONO_HACK for any Mono hack involving legacy mono channel to |
| // stereo channel conversion. |
| |
| /* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is |
| * being used. This is a considerable amount of log spam, so don't enable unless you |
| * are verifying the hook based code. |
| */ |
| //#define VERY_VERY_VERBOSE_LOGGING |
| #ifdef VERY_VERY_VERBOSE_LOGGING |
| #define ALOGVV ALOGV |
| //define ALOGVV printf // for test-mixer.cpp |
| #else |
| #define ALOGVV(a...) do { } while (0) |
| #endif |
| |
| // Set to default copy buffer size in frames for input processing. |
| static constexpr size_t kCopyBufferFrameCount = 256; |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const { |
| return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible. |
| } |
| |
| // Called when channel masks have changed for a track name |
| // TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format, |
| // which will simplify this logic. |
| bool AudioMixer::setChannelMasks(int name, |
| audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<Track> &track = getTrack(name); |
| |
| if (trackChannelMask == (track->channelMask | track->mHapticChannelMask) |
| && mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) { |
| return false; // no need to change |
| } |
| const audio_channel_mask_t hapticChannelMask = |
| static_cast<audio_channel_mask_t>(trackChannelMask & AUDIO_CHANNEL_HAPTIC_ALL); |
| trackChannelMask = static_cast<audio_channel_mask_t>( |
| trackChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL); |
| const audio_channel_mask_t mixerHapticChannelMask = static_cast<audio_channel_mask_t>( |
| mixerChannelMask & AUDIO_CHANNEL_HAPTIC_ALL); |
| mixerChannelMask = static_cast<audio_channel_mask_t>( |
| mixerChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL); |
| // always recompute for both channel masks even if only one has changed. |
| const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask); |
| const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask); |
| const uint32_t hapticChannelCount = audio_channel_count_from_out_mask(hapticChannelMask); |
| const uint32_t mixerHapticChannelCount = |
| audio_channel_count_from_out_mask(mixerHapticChannelMask); |
| |
| ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) |
| && trackChannelCount |
| && mixerChannelCount); |
| track->channelMask = trackChannelMask; |
| track->channelCount = trackChannelCount; |
| track->mMixerChannelMask = mixerChannelMask; |
| track->mMixerChannelCount = mixerChannelCount; |
| track->mHapticChannelMask = hapticChannelMask; |
| track->mHapticChannelCount = hapticChannelCount; |
| track->mMixerHapticChannelMask = mixerHapticChannelMask; |
| track->mMixerHapticChannelCount = mixerHapticChannelCount; |
| |
| if (track->mHapticChannelCount > 0) { |
| track->mAdjustInChannelCount = track->channelCount + track->mHapticChannelCount; |
| track->mAdjustOutChannelCount = track->channelCount; |
| track->mKeepContractedChannels = track->mHapticPlaybackEnabled; |
| } else { |
| track->mAdjustInChannelCount = 0; |
| track->mAdjustOutChannelCount = 0; |
| track->mKeepContractedChannels = false; |
| } |
| |
| track->mInputFrameSize = audio_bytes_per_frame( |
| track->channelCount + track->mHapticChannelCount, track->mFormat); |
| |
| // channel masks have changed, does this track need a downmixer? |
| // update to try using our desired format (if we aren't already using it) |
| const status_t status = track->prepareForDownmix(); |
| ALOGE_IF(status != OK, |
| "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x", |
| status, track->channelMask, track->mMixerChannelMask); |
| |
| // always do reformat since channel mask changed, |
| // do it after downmix since track format may change! |
| track->prepareForReformat(); |
| |
| track->prepareForAdjustChannels(mFrameCount); |
| |
| // Resampler channels may have changed. |
| track->recreateResampler(mSampleRate); |
| return true; |
| } |
| |
| void AudioMixer::Track::unprepareForDownmix() { |
| ALOGV("AudioMixer::unprepareForDownmix(%p)", this); |
| |
| if (mPostDownmixReformatBufferProvider.get() != nullptr) { |
| // release any buffers held by the mPostDownmixReformatBufferProvider |
| // before deallocating the mDownmixerBufferProvider. |
| mPostDownmixReformatBufferProvider->reset(); |
| } |
| |
| mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; |
| if (mDownmixerBufferProvider.get() != nullptr) { |
| // this track had previously been configured with a downmixer, delete it |
| mDownmixerBufferProvider.reset(nullptr); |
| reconfigureBufferProviders(); |
| } else { |
| ALOGV(" nothing to do, no downmixer to delete"); |
| } |
| } |
| |
| status_t AudioMixer::Track::prepareForDownmix() |
| { |
| ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x", |
| this, channelMask); |
| |
| // discard the previous downmixer if there was one |
| unprepareForDownmix(); |
| // MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks |
| // are not the same and not handled internally, as mono for channel position masks is. |
| if (channelMask == mMixerChannelMask |
| || (channelMask == AUDIO_CHANNEL_OUT_MONO |
| && isAudioChannelPositionMask(mMixerChannelMask))) { |
| return NO_ERROR; |
| } |
| // DownmixerBufferProvider is only used for position masks. |
| if (audio_channel_mask_get_representation(channelMask) |
| == AUDIO_CHANNEL_REPRESENTATION_POSITION |
| && DownmixerBufferProvider::isMultichannelCapable()) { |
| |
| // Check if we have a float or int16 downmixer, in that order. |
| for (const audio_format_t format : { AUDIO_FORMAT_PCM_FLOAT, AUDIO_FORMAT_PCM_16_BIT }) { |
| mDownmixerBufferProvider.reset(new DownmixerBufferProvider( |
| channelMask, mMixerChannelMask, |
| format, |
| sampleRate, sessionId, kCopyBufferFrameCount)); |
| if (static_cast<DownmixerBufferProvider *>(mDownmixerBufferProvider.get()) |
| ->isValid()) { |
| mDownmixRequiresFormat = format; |
| reconfigureBufferProviders(); |
| return NO_ERROR; |
| } |
| } |
| // mDownmixerBufferProvider reset below. |
| } |
| |
| // See if we should use our built-in non-effect downmixer. |
| if (mMixerInFormat == AUDIO_FORMAT_PCM_FLOAT |
| && ChannelMixBufferProvider::isOutputChannelMaskSupported(mMixerChannelMask) |
| && audio_channel_mask_get_representation(channelMask) |
| == AUDIO_CHANNEL_REPRESENTATION_POSITION) { |
| mDownmixerBufferProvider.reset(new ChannelMixBufferProvider(channelMask, |
| mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount)); |
| if (static_cast<ChannelMixBufferProvider *>(mDownmixerBufferProvider.get()) |
| ->isValid()) { |
| mDownmixRequiresFormat = mMixerInFormat; |
| reconfigureBufferProviders(); |
| ALOGD("%s: Fallback using ChannelMix", __func__); |
| return NO_ERROR; |
| } else { |
| ALOGD("%s: ChannelMix not supported for channel mask %#x", __func__, channelMask); |
| } |
| } |
| |
| // Effect downmixer does not accept the channel conversion. Let's use our remixer. |
| mDownmixerBufferProvider.reset(new RemixBufferProvider(channelMask, |
| mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount)); |
| // Remix always finds a conversion whereas Downmixer effect above may fail. |
| reconfigureBufferProviders(); |
| return NO_ERROR; |
| } |
| |
| void AudioMixer::Track::unprepareForReformat() { |
| ALOGV("AudioMixer::unprepareForReformat(%p)", this); |
| bool requiresReconfigure = false; |
| if (mReformatBufferProvider.get() != nullptr) { |
| mReformatBufferProvider.reset(nullptr); |
| requiresReconfigure = true; |
| } |
| if (mPostDownmixReformatBufferProvider.get() != nullptr) { |
| mPostDownmixReformatBufferProvider.reset(nullptr); |
| requiresReconfigure = true; |
| } |
| if (requiresReconfigure) { |
| reconfigureBufferProviders(); |
| } |
| } |
| |
| status_t AudioMixer::Track::prepareForReformat() |
| { |
| ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat); |
| // discard previous reformatters |
| unprepareForReformat(); |
| // only configure reformatters as needed |
| const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID |
| ? mDownmixRequiresFormat : mMixerInFormat; |
| bool requiresReconfigure = false; |
| if (mFormat != targetFormat) { |
| mReformatBufferProvider.reset(new ReformatBufferProvider( |
| audio_channel_count_from_out_mask(channelMask), |
| mFormat, |
| targetFormat, |
| kCopyBufferFrameCount)); |
| requiresReconfigure = true; |
| } else if (mFormat == AUDIO_FORMAT_PCM_FLOAT) { |
| // Input and output are floats, make sure application did not provide > 3db samples |
| // that would break volume application (b/68099072) |
| // TODO: add a trusted source flag to avoid the overhead |
| mReformatBufferProvider.reset(new ClampFloatBufferProvider( |
| audio_channel_count_from_out_mask(channelMask), |
| kCopyBufferFrameCount)); |
| requiresReconfigure = true; |
| } |
| if (targetFormat != mMixerInFormat) { |
| mPostDownmixReformatBufferProvider.reset(new ReformatBufferProvider( |
| audio_channel_count_from_out_mask(mMixerChannelMask), |
| targetFormat, |
| mMixerInFormat, |
| kCopyBufferFrameCount)); |
| requiresReconfigure = true; |
| } |
| if (requiresReconfigure) { |
| reconfigureBufferProviders(); |
| } |
| return NO_ERROR; |
| } |
| |
| void AudioMixer::Track::unprepareForAdjustChannels() |
| { |
| ALOGV("AUDIOMIXER::unprepareForAdjustChannels"); |
| if (mAdjustChannelsBufferProvider.get() != nullptr) { |
| mAdjustChannelsBufferProvider.reset(nullptr); |
| reconfigureBufferProviders(); |
| } |
| } |
| |
| status_t AudioMixer::Track::prepareForAdjustChannels(size_t frames) |
| { |
| ALOGV("AudioMixer::prepareForAdjustChannels(%p) with inChannelCount: %u, outChannelCount: %u", |
| this, mAdjustInChannelCount, mAdjustOutChannelCount); |
| unprepareForAdjustChannels(); |
| if (mAdjustInChannelCount != mAdjustOutChannelCount) { |
| uint8_t* buffer = mKeepContractedChannels |
| ? (uint8_t*)mainBuffer + frames * audio_bytes_per_frame( |
| mMixerChannelCount, mMixerFormat) |
| : nullptr; |
| mAdjustChannelsBufferProvider.reset(new AdjustChannelsBufferProvider( |
| mFormat, mAdjustInChannelCount, mAdjustOutChannelCount, frames, |
| mKeepContractedChannels ? mMixerFormat : AUDIO_FORMAT_INVALID, |
| buffer, mMixerHapticChannelCount)); |
| reconfigureBufferProviders(); |
| } |
| return NO_ERROR; |
| } |
| |
| void AudioMixer::Track::unprepareForTee() { |
| ALOGV("AudioMixer::%s", __func__); |
| if (mTeeBufferProvider.get() != nullptr) { |
| mTeeBufferProvider.reset(nullptr); |
| reconfigureBufferProviders(); |
| } |
| } |
| |
| status_t AudioMixer::Track::prepareForTee() { |
| ALOGV("AudioMixer::%s(%p) teeBuffer=%p", __func__, this, teeBuffer); |
| unprepareForTee(); |
| if (teeBuffer != nullptr) { |
| mTeeBufferProvider.reset(new TeeBufferProvider( |
| mInputFrameSize, mInputFrameSize, kCopyBufferFrameCount, |
| (uint8_t*)teeBuffer, mTeeBufferFrameCount)); |
| reconfigureBufferProviders(); |
| } |
| return NO_ERROR; |
| } |
| |
| void AudioMixer::Track::clearContractedBuffer() |
| { |
| if (mAdjustChannelsBufferProvider.get() != nullptr) { |
| static_cast<AdjustChannelsBufferProvider*>( |
| mAdjustChannelsBufferProvider.get())->clearContractedFrames(); |
| } |
| } |
| |
| void AudioMixer::Track::clearTeeFrameCopied() { |
| if (mTeeBufferProvider.get() != nullptr) { |
| static_cast<TeeBufferProvider*>(mTeeBufferProvider.get())->clearFramesCopied(); |
| } |
| } |
| |
| void AudioMixer::Track::reconfigureBufferProviders() |
| { |
| // configure from upstream to downstream buffer providers. |
| bufferProvider = mInputBufferProvider; |
| if (mTeeBufferProvider != nullptr) { |
| mTeeBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mTeeBufferProvider.get(); |
| } |
| if (mAdjustChannelsBufferProvider.get() != nullptr) { |
| mAdjustChannelsBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mAdjustChannelsBufferProvider.get(); |
| } |
| if (mReformatBufferProvider.get() != nullptr) { |
| mReformatBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mReformatBufferProvider.get(); |
| } |
| if (mDownmixerBufferProvider.get() != nullptr) { |
| mDownmixerBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mDownmixerBufferProvider.get(); |
| } |
| if (mPostDownmixReformatBufferProvider.get() != nullptr) { |
| mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mPostDownmixReformatBufferProvider.get(); |
| } |
| if (mTimestretchBufferProvider.get() != nullptr) { |
| mTimestretchBufferProvider->setBufferProvider(bufferProvider); |
| bufferProvider = mTimestretchBufferProvider.get(); |
| } |
| } |
| |
| void AudioMixer::setParameter(int name, int target, int param, void *value) |
| { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<Track> &track = getTrack(name); |
| |
| int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value)); |
| int32_t *valueBuf = reinterpret_cast<int32_t*>(value); |
| |
| switch (target) { |
| |
| case TRACK: |
| switch (param) { |
| case CHANNEL_MASK: { |
| const audio_channel_mask_t trackChannelMask = |
| static_cast<audio_channel_mask_t>(valueInt); |
| if (setChannelMasks(name, trackChannelMask, |
| static_cast<audio_channel_mask_t>( |
| track->mMixerChannelMask | track->mMixerHapticChannelMask))) { |
| ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask); |
| invalidate(); |
| } |
| } break; |
| case MAIN_BUFFER: |
| if (track->mainBuffer != valueBuf) { |
| track->mainBuffer = valueBuf; |
| ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); |
| if (track->mKeepContractedChannels) { |
| track->prepareForAdjustChannels(mFrameCount); |
| } |
| invalidate(); |
| } |
| break; |
| case AUX_BUFFER: |
| AudioMixerBase::setParameter(name, target, param, value); |
| break; |
| case FORMAT: { |
| audio_format_t format = static_cast<audio_format_t>(valueInt); |
| if (track->mFormat != format) { |
| ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format); |
| track->mFormat = format; |
| ALOGV("setParameter(TRACK, FORMAT, %#x)", format); |
| track->prepareForReformat(); |
| invalidate(); |
| } |
| } break; |
| // FIXME do we want to support setting the downmix type from AudioFlinger? |
| // for a specific track? or per mixer? |
| /* case DOWNMIX_TYPE: |
| break */ |
| case MIXER_FORMAT: { |
| audio_format_t format = static_cast<audio_format_t>(valueInt); |
| if (track->mMixerFormat != format) { |
| track->mMixerFormat = format; |
| ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format); |
| if (track->mKeepContractedChannels) { |
| track->prepareForAdjustChannels(mFrameCount); |
| } |
| } |
| } break; |
| case MIXER_CHANNEL_MASK: { |
| const audio_channel_mask_t mixerChannelMask = |
| static_cast<audio_channel_mask_t>(valueInt); |
| if (setChannelMasks(name, static_cast<audio_channel_mask_t>( |
| track->channelMask | track->mHapticChannelMask), |
| mixerChannelMask)) { |
| ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask); |
| invalidate(); |
| } |
| } break; |
| case HAPTIC_ENABLED: { |
| const bool hapticPlaybackEnabled = static_cast<bool>(valueInt); |
| if (track->mHapticPlaybackEnabled != hapticPlaybackEnabled) { |
| track->mHapticPlaybackEnabled = hapticPlaybackEnabled; |
| track->mKeepContractedChannels = hapticPlaybackEnabled; |
| track->prepareForAdjustChannels(mFrameCount); |
| } |
| } break; |
| case HAPTIC_INTENSITY: { |
| const os::HapticScale hapticIntensity = static_cast<os::HapticScale>(valueInt); |
| if (track->mHapticIntensity != hapticIntensity) { |
| track->mHapticIntensity = hapticIntensity; |
| } |
| } break; |
| case HAPTIC_MAX_AMPLITUDE: { |
| const float hapticMaxAmplitude = *reinterpret_cast<float*>(value); |
| if (track->mHapticMaxAmplitude != hapticMaxAmplitude) { |
| track->mHapticMaxAmplitude = hapticMaxAmplitude; |
| } |
| } break; |
| case TEE_BUFFER: |
| if (track->teeBuffer != valueBuf) { |
| track->teeBuffer = valueBuf; |
| ALOGV("setParameter(TRACK, TEE_BUFFER, %p)", valueBuf); |
| track->prepareForTee(); |
| } |
| break; |
| case TEE_BUFFER_FRAME_COUNT: |
| if (track->mTeeBufferFrameCount != valueInt) { |
| track->mTeeBufferFrameCount = valueInt; |
| ALOGV("setParameter(TRACK, TEE_BUFFER_FRAME_COUNT, %i)", valueInt); |
| track->prepareForTee(); |
| } |
| break; |
| default: |
| LOG_ALWAYS_FATAL("setParameter track: bad param %d", param); |
| } |
| break; |
| |
| case RESAMPLE: |
| case RAMP_VOLUME: |
| case VOLUME: |
| AudioMixerBase::setParameter(name, target, param, value); |
| break; |
| case TIMESTRETCH: |
| switch (param) { |
| case PLAYBACK_RATE: { |
| const AudioPlaybackRate *playbackRate = |
| reinterpret_cast<AudioPlaybackRate*>(value); |
| ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate), |
| "bad parameters speed %f, pitch %f", |
| playbackRate->mSpeed, playbackRate->mPitch); |
| if (track->setPlaybackRate(*playbackRate)) { |
| ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE " |
| "%f %f %d %d", |
| playbackRate->mSpeed, |
| playbackRate->mPitch, |
| playbackRate->mStretchMode, |
| playbackRate->mFallbackMode); |
| // invalidate(); (should not require reconfigure) |
| } |
| } break; |
| default: |
| LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param); |
| } |
| break; |
| |
| default: |
| LOG_ALWAYS_FATAL("setParameter: bad target %d", target); |
| } |
| } |
| |
| bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate) |
| { |
| if ((mTimestretchBufferProvider.get() == nullptr && |
| fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA && |
| fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) || |
| isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { |
| return false; |
| } |
| mPlaybackRate = playbackRate; |
| if (mTimestretchBufferProvider.get() == nullptr) { |
| // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer |
| // but if none exists, it is the channel count (1 for mono). |
| const int timestretchChannelCount = getOutputChannelCount(); |
| mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount, |
| mMixerInFormat, sampleRate, playbackRate)); |
| reconfigureBufferProviders(); |
| } else { |
| static_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider.get()) |
| ->setPlaybackRate(playbackRate); |
| } |
| return true; |
| } |
| |
| void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) |
| { |
| LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name); |
| const std::shared_ptr<Track> &track = getTrack(name); |
| |
| if (track->mInputBufferProvider == bufferProvider) { |
| return; // don't reset any buffer providers if identical. |
| } |
| // reset order from downstream to upstream buffer providers. |
| if (track->mTimestretchBufferProvider.get() != nullptr) { |
| track->mTimestretchBufferProvider->reset(); |
| } else if (track->mPostDownmixReformatBufferProvider.get() != nullptr) { |
| track->mPostDownmixReformatBufferProvider->reset(); |
| } else if (track->mDownmixerBufferProvider != nullptr) { |
| track->mDownmixerBufferProvider->reset(); |
| } else if (track->mReformatBufferProvider.get() != nullptr) { |
| track->mReformatBufferProvider->reset(); |
| } else if (track->mAdjustChannelsBufferProvider.get() != nullptr) { |
| track->mAdjustChannelsBufferProvider->reset(); |
| } else if (track->mTeeBufferProvider.get() != nullptr) { |
| track->mTeeBufferProvider->reset(); |
| } |
| |
| track->mInputBufferProvider = bufferProvider; |
| track->reconfigureBufferProviders(); |
| } |
| |
| /*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT; |
| |
| /*static*/ void AudioMixer::sInitRoutine() |
| { |
| DownmixerBufferProvider::init(); // for the downmixer |
| } |
| |
| std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack() |
| { |
| return std::make_shared<Track>(); |
| } |
| |
| status_t AudioMixer::postCreateTrack(TrackBase *track) |
| { |
| Track* t = static_cast<Track*>(track); |
| |
| audio_channel_mask_t channelMask = t->channelMask; |
| t->mHapticChannelMask = static_cast<audio_channel_mask_t>( |
| channelMask & AUDIO_CHANNEL_HAPTIC_ALL); |
| t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask); |
| channelMask = static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL); |
| t->channelCount = audio_channel_count_from_out_mask(channelMask); |
| ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO, |
| "Non-stereo channel mask: %d\n", channelMask); |
| t->channelMask = channelMask; |
| t->mInputBufferProvider = NULL; |
| t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required |
| t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; |
| // haptic |
| t->mHapticPlaybackEnabled = false; |
| t->mHapticIntensity = os::HapticScale::NONE; |
| t->mHapticMaxAmplitude = NAN; |
| t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE; |
| t->mMixerHapticChannelCount = 0; |
| t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount; |
| t->mAdjustOutChannelCount = t->channelCount; |
| t->mKeepContractedChannels = false; |
| t->mInputFrameSize = audio_bytes_per_frame( |
| t->channelCount + t->mHapticChannelCount, t->mFormat); |
| // Check the downmixing (or upmixing) requirements. |
| status_t status = t->prepareForDownmix(); |
| if (status != OK) { |
| ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask); |
| return BAD_VALUE; |
| } |
| // prepareForDownmix() may change mDownmixRequiresFormat |
| ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat); |
| t->prepareForReformat(); |
| t->prepareForAdjustChannels(mFrameCount); |
| return OK; |
| } |
| |
| void AudioMixer::preProcess() |
| { |
| for (const auto &pair : mTracks) { |
| // Clear contracted buffer before processing if contracted channels are saved |
| const std::shared_ptr<TrackBase> &tb = pair.second; |
| Track *t = static_cast<Track*>(tb.get()); |
| if (t->mKeepContractedChannels) { |
| t->clearContractedBuffer(); |
| } |
| t->clearTeeFrameCopied(); |
| } |
| } |
| |
| void AudioMixer::postProcess() |
| { |
| // Process haptic data. |
| // Need to keep consistent with VibrationEffect.scale(int, float, int) |
| for (const auto &pair : mGroups) { |
| // process by group of tracks with same output main buffer. |
| const auto &group = pair.second; |
| for (const int name : group) { |
| const std::shared_ptr<Track> &t = getTrack(name); |
| if (t->mHapticPlaybackEnabled) { |
| size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount; |
| uint8_t* buffer = (uint8_t*)pair.first + mFrameCount * audio_bytes_per_frame( |
| t->mMixerChannelCount, t->mMixerFormat); |
| switch (t->mMixerFormat) { |
| // Mixer format should be AUDIO_FORMAT_PCM_FLOAT. |
| case AUDIO_FORMAT_PCM_FLOAT: { |
| os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity, |
| t->mHapticMaxAmplitude); |
| } break; |
| default: |
| LOG_ALWAYS_FATAL("bad mMixerFormat: %#x", t->mMixerFormat); |
| break; |
| } |
| break; |
| } |
| if (t->teeBuffer != nullptr && t->volumeRL == 0) { |
| // Need to mute tee |
| memset(t->teeBuffer, 0, t->mTeeBufferFrameCount * t->mInputFrameSize); |
| } |
| } |
| } |
| } |
| |
| // ---------------------------------------------------------------------------- |
| } // namespace android |