| /* |
| * Copyright (C) 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 "r_submix" |
| //#define LOG_NDEBUG 0 |
| |
| #include <errno.h> |
| #include <pthread.h> |
| #include <stdint.h> |
| #include <stdlib.h> |
| #include <sys/param.h> |
| #include <sys/time.h> |
| #include <sys/limits.h> |
| |
| #include <cutils/log.h> |
| #include <cutils/properties.h> |
| #include <cutils/str_parms.h> |
| |
| #include <hardware/audio.h> |
| #include <hardware/hardware.h> |
| #include <system/audio.h> |
| |
| #include <media/AudioParameter.h> |
| #include <media/AudioBufferProvider.h> |
| #include <media/nbaio/MonoPipe.h> |
| #include <media/nbaio/MonoPipeReader.h> |
| |
| #include <utils/String8.h> |
| |
| #define LOG_STREAMS_TO_FILES 0 |
| #if LOG_STREAMS_TO_FILES |
| #include <fcntl.h> |
| #include <stdio.h> |
| #include <sys/stat.h> |
| #endif // LOG_STREAMS_TO_FILES |
| |
| extern "C" { |
| |
| namespace android { |
| |
| // Set to 1 to enable extremely verbose logging in this module. |
| #define SUBMIX_VERBOSE_LOGGING 0 |
| #if SUBMIX_VERBOSE_LOGGING |
| #define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__) |
| #define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__) |
| #else |
| #define SUBMIX_ALOGV(...) |
| #define SUBMIX_ALOGE(...) |
| #endif // SUBMIX_VERBOSE_LOGGING |
| |
| // NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe(). |
| #define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*8) |
| // Value used to divide the MonoPipe() buffer into segments that are written to the source and |
| // read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer |
| // the minimum latency is the MonoPipe buffer size divided by this value. |
| #define DEFAULT_PIPE_PERIOD_COUNT 4 |
| // The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to |
| // the duration of a record buffer at the current record sample rate (of the device, not of |
| // the recording itself). Here we have: |
| // 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms |
| #define MAX_READ_ATTEMPTS 3 |
| #define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty |
| #define DEFAULT_SAMPLE_RATE_HZ 48000 // default sample rate |
| // See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h. |
| #define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT |
| // A legacy user of this device does not close the input stream when it shuts down, which |
| // results in the application opening a new input stream before closing the old input stream |
| // handle it was previously using. Setting this value to 1 allows multiple clients to open |
| // multiple input streams from this device. If this option is enabled, each input stream returned |
| // is *the same stream* which means that readers will race to read data from these streams. |
| #define ENABLE_LEGACY_INPUT_OPEN 1 |
| // Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled. |
| #define ENABLE_CHANNEL_CONVERSION 1 |
| // Whether resampling is enabled. |
| #define ENABLE_RESAMPLING 1 |
| #if LOG_STREAMS_TO_FILES |
| // Folder to save stream log files to. |
| #define LOG_STREAM_FOLDER "/data/misc/media" |
| // Log filenames for input and output streams. |
| #define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw" |
| #define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw" |
| // File permissions for stream log files. |
| #define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH) |
| #endif // LOG_STREAMS_TO_FILES |
| // limit for number of read error log entries to avoid spamming the logs |
| #define MAX_READ_ERROR_LOGS 5 |
| |
| // Common limits macros. |
| #ifndef min |
| #define min(a, b) ((a) < (b) ? (a) : (b)) |
| #endif // min |
| #ifndef max |
| #define max(a, b) ((a) > (b) ? (a) : (b)) |
| #endif // max |
| |
| // Set *result_variable_ptr to true if value_to_find is present in the array array_to_search, |
| // otherwise set *result_variable_ptr to false. |
| #define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \ |
| { \ |
| size_t i; \ |
| *(result_variable_ptr) = false; \ |
| for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \ |
| if ((value_to_find) == (array_to_search)[i]) { \ |
| *(result_variable_ptr) = true; \ |
| break; \ |
| } \ |
| } \ |
| } |
| |
| // Configuration of the submix pipe. |
| struct submix_config { |
| // Channel mask field in this data structure is set to either input_channel_mask or |
| // output_channel_mask depending upon the last stream to be opened on this device. |
| struct audio_config common; |
| // Input stream and output stream channel masks. This is required since input and output |
| // channel bitfields are not equivalent. |
| audio_channel_mask_t input_channel_mask; |
| audio_channel_mask_t output_channel_mask; |
| #if ENABLE_RESAMPLING |
| // Input stream and output stream sample rates. |
| uint32_t input_sample_rate; |
| uint32_t output_sample_rate; |
| #endif // ENABLE_RESAMPLING |
| size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. |
| size_t buffer_size_frames; // Size of the audio pipe in frames. |
| // Maximum number of frames buffered by the input and output streams. |
| size_t buffer_period_size_frames; |
| }; |
| |
| #define MAX_ROUTES 10 |
| typedef struct route_config { |
| struct submix_config config; |
| char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; |
| // Pipe variables: they handle the ring buffer that "pipes" audio: |
| // - from the submix virtual audio output == what needs to be played |
| // remotely, seen as an output for AudioFlinger |
| // - to the virtual audio source == what is captured by the component |
| // which "records" the submix / virtual audio source, and handles it as needed. |
| // A usecase example is one where the component capturing the audio is then sending it over |
| // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a |
| // TV with Wifi Display capabilities), or to a wireless audio player. |
| sp<MonoPipe> rsxSink; |
| sp<MonoPipeReader> rsxSource; |
| // Pointers to the current input and output stream instances. rsxSink and rsxSource are |
| // destroyed if both and input and output streams are destroyed. |
| struct submix_stream_out *output; |
| struct submix_stream_in *input; |
| #if ENABLE_RESAMPLING |
| // Buffer used as temporary storage for resampled data prior to returning data to the output |
| // stream. |
| int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; |
| #endif // ENABLE_RESAMPLING |
| } route_config_t; |
| |
| struct submix_audio_device { |
| struct audio_hw_device device; |
| route_config_t routes[MAX_ROUTES]; |
| // Device lock, also used to protect access to submix_audio_device from the input and output |
| // streams. |
| pthread_mutex_t lock; |
| }; |
| |
| struct submix_stream_out { |
| struct audio_stream_out stream; |
| struct submix_audio_device *dev; |
| int route_handle; |
| bool output_standby; |
| #if LOG_STREAMS_TO_FILES |
| int log_fd; |
| #endif // LOG_STREAMS_TO_FILES |
| }; |
| |
| struct submix_stream_in { |
| struct audio_stream_in stream; |
| struct submix_audio_device *dev; |
| int route_handle; |
| bool input_standby; |
| bool output_standby_rec_thr; // output standby state as seen from record thread |
| |
| // wall clock when recording starts |
| struct timespec record_start_time; |
| // how many frames have been requested to be read |
| int64_t read_counter_frames; |
| |
| #if ENABLE_LEGACY_INPUT_OPEN |
| // Number of references to this input stream. |
| volatile int32_t ref_count; |
| #endif // ENABLE_LEGACY_INPUT_OPEN |
| #if LOG_STREAMS_TO_FILES |
| int log_fd; |
| #endif // LOG_STREAMS_TO_FILES |
| |
| volatile int16_t read_error_count; |
| }; |
| |
| // Determine whether the specified sample rate is supported by the submix module. |
| static bool sample_rate_supported(const uint32_t sample_rate) |
| { |
| // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. |
| static const unsigned int supported_sample_rates[] = { |
| 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, |
| }; |
| bool return_value; |
| SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); |
| return return_value; |
| } |
| |
| // Determine whether the specified sample rate is supported, if it is return the specified sample |
| // rate, otherwise return the default sample rate for the submix module. |
| static uint32_t get_supported_sample_rate(uint32_t sample_rate) |
| { |
| return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; |
| } |
| |
| // Determine whether the specified channel in mask is supported by the submix module. |
| static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) |
| { |
| // Set of channel in masks supported by Format_from_SR_C() |
| // frameworks/av/media/libnbaio/NAIO.cpp. |
| static const audio_channel_mask_t supported_channel_in_masks[] = { |
| AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, |
| }; |
| bool return_value; |
| SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); |
| return return_value; |
| } |
| |
| // Determine whether the specified channel in mask is supported, if it is return the specified |
| // channel in mask, otherwise return the default channel in mask for the submix module. |
| static audio_channel_mask_t get_supported_channel_in_mask( |
| const audio_channel_mask_t channel_in_mask) |
| { |
| return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : |
| static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO); |
| } |
| |
| // Determine whether the specified channel out mask is supported by the submix module. |
| static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) |
| { |
| // Set of channel out masks supported by Format_from_SR_C() |
| // frameworks/av/media/libnbaio/NAIO.cpp. |
| static const audio_channel_mask_t supported_channel_out_masks[] = { |
| AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, |
| }; |
| bool return_value; |
| SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); |
| return return_value; |
| } |
| |
| // Determine whether the specified channel out mask is supported, if it is return the specified |
| // channel out mask, otherwise return the default channel out mask for the submix module. |
| static audio_channel_mask_t get_supported_channel_out_mask( |
| const audio_channel_mask_t channel_out_mask) |
| { |
| return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : |
| static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO); |
| } |
| |
| // Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the |
| // structure. |
| static struct submix_stream_out * audio_stream_out_get_submix_stream_out( |
| struct audio_stream_out * const stream) |
| { |
| ALOG_ASSERT(stream); |
| return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) - |
| offsetof(struct submix_stream_out, stream)); |
| } |
| |
| // Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. |
| static struct submix_stream_out * audio_stream_get_submix_stream_out( |
| struct audio_stream * const stream) |
| { |
| ALOG_ASSERT(stream); |
| return audio_stream_out_get_submix_stream_out( |
| reinterpret_cast<struct audio_stream_out *>(stream)); |
| } |
| |
| // Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the |
| // structure. |
| static struct submix_stream_in * audio_stream_in_get_submix_stream_in( |
| struct audio_stream_in * const stream) |
| { |
| ALOG_ASSERT(stream); |
| return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) - |
| offsetof(struct submix_stream_in, stream)); |
| } |
| |
| // Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. |
| static struct submix_stream_in * audio_stream_get_submix_stream_in( |
| struct audio_stream * const stream) |
| { |
| ALOG_ASSERT(stream); |
| return audio_stream_in_get_submix_stream_in( |
| reinterpret_cast<struct audio_stream_in *>(stream)); |
| } |
| |
| // Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within |
| // the structure. |
| static struct submix_audio_device * audio_hw_device_get_submix_audio_device( |
| struct audio_hw_device *device) |
| { |
| ALOG_ASSERT(device); |
| return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) - |
| offsetof(struct submix_audio_device, device)); |
| } |
| |
| // Compare an audio_config with input channel mask and an audio_config with output channel mask |
| // returning false if they do *not* match, true otherwise. |
| static bool audio_config_compare(const audio_config * const input_config, |
| const audio_config * const output_config) |
| { |
| #if !ENABLE_CHANNEL_CONVERSION |
| const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); |
| const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); |
| if (input_channels != output_channels) { |
| ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", |
| input_channels, output_channels); |
| return false; |
| } |
| #endif // !ENABLE_CHANNEL_CONVERSION |
| #if ENABLE_RESAMPLING |
| if (input_config->sample_rate != output_config->sample_rate && |
| audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { |
| #else |
| if (input_config->sample_rate != output_config->sample_rate) { |
| #endif // ENABLE_RESAMPLING |
| ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", |
| input_config->sample_rate, output_config->sample_rate); |
| return false; |
| } |
| if (input_config->format != output_config->format) { |
| ALOGE("audio_config_compare() format mismatch %x vs. %x", |
| input_config->format, output_config->format); |
| return false; |
| } |
| // This purposely ignores offload_info as it's not required for the submix device. |
| return true; |
| } |
| |
| // If one doesn't exist, create a pipe for the submix audio device rsxadev of size |
| // buffer_size_frames and optionally associate "in" or "out" with the submix audio device. |
| // Must be called with lock held on the submix_audio_device |
| static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, |
| const struct audio_config * const config, |
| const size_t buffer_size_frames, |
| const uint32_t buffer_period_count, |
| struct submix_stream_in * const in, |
| struct submix_stream_out * const out, |
| const char *address, |
| int route_idx) |
| { |
| ALOG_ASSERT(in || out); |
| ALOG_ASSERT(route_idx > -1); |
| ALOG_ASSERT(route_idx < MAX_ROUTES); |
| ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); |
| |
| // Save a reference to the specified input or output stream and the associated channel |
| // mask. |
| if (in) { |
| in->route_handle = route_idx; |
| rsxadev->routes[route_idx].input = in; |
| rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; |
| #if ENABLE_RESAMPLING |
| rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; |
| // If the output isn't configured yet, set the output sample rate to the maximum supported |
| // sample rate such that the smallest possible input buffer is created, and put a default |
| // value for channel count |
| if (!rsxadev->routes[route_idx].output) { |
| rsxadev->routes[route_idx].config.output_sample_rate = 48000; |
| rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; |
| } |
| #endif // ENABLE_RESAMPLING |
| } |
| if (out) { |
| out->route_handle = route_idx; |
| rsxadev->routes[route_idx].output = out; |
| rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; |
| #if ENABLE_RESAMPLING |
| rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; |
| #endif // ENABLE_RESAMPLING |
| } |
| // Save the address |
| strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); |
| ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); |
| // If a pipe isn't associated with the device, create one. |
| if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) |
| { |
| struct submix_config * const device_config = &rsxadev->routes[route_idx].config; |
| uint32_t channel_count; |
| if (out) |
| channel_count = audio_channel_count_from_out_mask(config->channel_mask); |
| else |
| channel_count = audio_channel_count_from_in_mask(config->channel_mask); |
| #if ENABLE_CHANNEL_CONVERSION |
| // If channel conversion is enabled, allocate enough space for the maximum number of |
| // possible channels stored in the pipe for the situation when the number of channels in |
| // the output stream don't match the number in the input stream. |
| const uint32_t pipe_channel_count = max(channel_count, 2); |
| #else |
| const uint32_t pipe_channel_count = channel_count; |
| #endif // ENABLE_CHANNEL_CONVERSION |
| const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, |
| config->format); |
| const NBAIO_Format offers[1] = {format}; |
| size_t numCounterOffers = 0; |
| // Create a MonoPipe with optional blocking set to true. |
| MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); |
| // Negotiation between the source and sink cannot fail as the device open operation |
| // creates both ends of the pipe using the same audio format. |
| ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); |
| ALOG_ASSERT(index == 0); |
| MonoPipeReader* source = new MonoPipeReader(sink); |
| numCounterOffers = 0; |
| index = source->negotiate(offers, 1, NULL, numCounterOffers); |
| ALOG_ASSERT(index == 0); |
| ALOGV("submix_audio_device_create_pipe_l(): created pipe"); |
| |
| // Save references to the source and sink. |
| ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); |
| ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); |
| rsxadev->routes[route_idx].rsxSink = sink; |
| rsxadev->routes[route_idx].rsxSource = source; |
| // Store the sanitized audio format in the device so that it's possible to determine |
| // the format of the pipe source when opening the input device. |
| memcpy(&device_config->common, config, sizeof(device_config->common)); |
| device_config->buffer_size_frames = sink->maxFrames(); |
| device_config->buffer_period_size_frames = device_config->buffer_size_frames / |
| buffer_period_count; |
| if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); |
| if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); |
| #if ENABLE_CHANNEL_CONVERSION |
| // Calculate the pipe frame size based upon the number of channels. |
| device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / |
| channel_count; |
| #endif // ENABLE_CHANNEL_CONVERSION |
| SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " |
| "period size %zd", device_config->pipe_frame_size, |
| device_config->buffer_size_frames, device_config->buffer_period_size_frames); |
| } |
| } |
| |
| // Release references to the sink and source. Input and output threads may maintain references |
| // to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use |
| // before they shutdown. |
| // Must be called with lock held on the submix_audio_device |
| static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, |
| int route_idx) |
| { |
| ALOG_ASSERT(route_idx > -1); |
| ALOG_ASSERT(route_idx < MAX_ROUTES); |
| ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, |
| rsxadev->routes[route_idx].address); |
| if (rsxadev->routes[route_idx].rsxSink != 0) { |
| rsxadev->routes[route_idx].rsxSink.clear(); |
| rsxadev->routes[route_idx].rsxSink = 0; |
| } |
| if (rsxadev->routes[route_idx].rsxSource != 0) { |
| rsxadev->routes[route_idx].rsxSource.clear(); |
| rsxadev->routes[route_idx].rsxSource = 0; |
| } |
| memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); |
| #ifdef ENABLE_RESAMPLING |
| memset(rsxadev->routes[route_idx].resampler_buffer, 0, |
| sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); |
| #endif |
| } |
| |
| // Remove references to the specified input and output streams. When the device no longer |
| // references input and output streams destroy the associated pipe. |
| // Must be called with lock held on the submix_audio_device |
| static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, |
| const struct submix_stream_in * const in, |
| const struct submix_stream_out * const out) |
| { |
| MonoPipe* sink; |
| ALOGV("submix_audio_device_destroy_pipe_l()"); |
| int route_idx = -1; |
| if (in != NULL) { |
| #if ENABLE_LEGACY_INPUT_OPEN |
| const_cast<struct submix_stream_in*>(in)->ref_count--; |
| route_idx = in->route_handle; |
| ALOG_ASSERT(rsxadev->routes[route_idx].input == in); |
| if (in->ref_count == 0) { |
| rsxadev->routes[route_idx].input = NULL; |
| } |
| ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); |
| #else |
| rsxadev->input = NULL; |
| #endif // ENABLE_LEGACY_INPUT_OPEN |
| } |
| if (out != NULL) { |
| route_idx = out->route_handle; |
| ALOG_ASSERT(rsxadev->routes[route_idx].output == out); |
| rsxadev->routes[route_idx].output = NULL; |
| } |
| if (route_idx != -1 && |
| rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { |
| submix_audio_device_release_pipe_l(rsxadev, route_idx); |
| ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); |
| } |
| } |
| |
| // Sanitize the user specified audio config for a submix input / output stream. |
| static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) |
| { |
| config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : |
| get_supported_channel_out_mask(config->channel_mask); |
| config->sample_rate = get_supported_sample_rate(config->sample_rate); |
| config->format = DEFAULT_FORMAT; |
| } |
| |
| // Verify a submix input or output stream can be opened. |
| // Must be called with lock held on the submix_audio_device |
| static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, |
| int route_idx, |
| const struct audio_config * const config, |
| const bool opening_input) |
| { |
| bool input_open; |
| bool output_open; |
| audio_config pipe_config; |
| |
| // Query the device for the current audio config and whether input and output streams are open. |
| output_open = rsxadev->routes[route_idx].output != NULL; |
| input_open = rsxadev->routes[route_idx].input != NULL; |
| memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); |
| |
| // If the stream is already open, don't open it again. |
| if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { |
| ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : |
| "Output"); |
| return false; |
| } |
| |
| SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " |
| "%s_channel_mask=%x", config->sample_rate, config->format, |
| opening_input ? "in" : "out", config->channel_mask); |
| |
| // If either stream is open, verify the existing audio config the pipe matches the user |
| // specified config. |
| if (input_open || output_open) { |
| const audio_config * const input_config = opening_input ? config : &pipe_config; |
| const audio_config * const output_config = opening_input ? &pipe_config : config; |
| // Get the channel mask of the open device. |
| pipe_config.channel_mask = |
| opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : |
| rsxadev->routes[route_idx].config.input_channel_mask; |
| if (!audio_config_compare(input_config, output_config)) { |
| ALOGE("submix_open_validate_l(): Unsupported format."); |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| // Must be called with lock held on the submix_audio_device |
| static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, |
| const char* address, /*in*/ |
| int *idx /*out*/) |
| { |
| // Do we already have a route for this address |
| int route_idx = -1; |
| int route_empty_idx = -1; // index of an empty route slot that can be used if needed |
| for (int i=0 ; i < MAX_ROUTES ; i++) { |
| if (strcmp(rsxadev->routes[i].address, "") == 0) { |
| route_empty_idx = i; |
| } |
| if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { |
| route_idx = i; |
| break; |
| } |
| } |
| |
| if ((route_idx == -1) && (route_empty_idx == -1)) { |
| ALOGE("Cannot create new route for address %s, max number of routes reached", address); |
| return -ENOMEM; |
| } |
| if (route_idx == -1) { |
| route_idx = route_empty_idx; |
| } |
| *idx = route_idx; |
| return OK; |
| } |
| |
| |
| // Calculate the maximum size of the pipe buffer in frames for the specified stream. |
| static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, |
| const struct submix_config *config, |
| const size_t pipe_frames, |
| const size_t stream_frame_size) |
| { |
| const size_t pipe_frame_size = config->pipe_frame_size; |
| const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); |
| return (pipe_frames * config->pipe_frame_size) / max_frame_size; |
| } |
| |
| /* audio HAL functions */ |
| |
| static uint32_t out_get_sample_rate(const struct audio_stream *stream) |
| { |
| const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( |
| const_cast<struct audio_stream *>(stream)); |
| #if ENABLE_RESAMPLING |
| const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; |
| #else |
| const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; |
| #endif // ENABLE_RESAMPLING |
| SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", |
| out_rate, out->dev->routes[out->route_handle].address); |
| return out_rate; |
| } |
| |
| static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) |
| { |
| struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); |
| #if ENABLE_RESAMPLING |
| // The sample rate of the stream can't be changed once it's set since this would change the |
| // output buffer size and hence break playback to the shared pipe. |
| if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { |
| ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " |
| "%u to %u for addr %s", |
| out->dev->routes[out->route_handle].config.output_sample_rate, rate, |
| out->dev->routes[out->route_handle].address); |
| return -ENOSYS; |
| } |
| #endif // ENABLE_RESAMPLING |
| if (!sample_rate_supported(rate)) { |
| ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); |
| return -ENOSYS; |
| } |
| SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); |
| out->dev->routes[out->route_handle].config.common.sample_rate = rate; |
| return 0; |
| } |
| |
| static size_t out_get_buffer_size(const struct audio_stream *stream) |
| { |
| const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( |
| const_cast<struct audio_stream *>(stream)); |
| const struct submix_config * const config = &out->dev->routes[out->route_handle].config; |
| const size_t stream_frame_size = |
| audio_stream_out_frame_size((const struct audio_stream_out *)stream); |
| const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( |
| stream, config, config->buffer_period_size_frames, stream_frame_size); |
| const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; |
| SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", |
| buffer_size_bytes, buffer_size_frames); |
| return buffer_size_bytes; |
| } |
| |
| static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) |
| { |
| const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( |
| const_cast<struct audio_stream *>(stream)); |
| uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; |
| SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); |
| return channel_mask; |
| } |
| |
| static audio_format_t out_get_format(const struct audio_stream *stream) |
| { |
| const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( |
| const_cast<struct audio_stream *>(stream)); |
| const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; |
| SUBMIX_ALOGV("out_get_format() returns %x", format); |
| return format; |
| } |
| |
| static int out_set_format(struct audio_stream *stream, audio_format_t format) |
| { |
| const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); |
| if (format != out->dev->routes[out->route_handle].config.common.format) { |
| ALOGE("out_set_format(format=%x) format unsupported", format); |
| return -ENOSYS; |
| } |
| SUBMIX_ALOGV("out_set_format(format=%x)", format); |
| return 0; |
| } |
| |
| static int out_standby(struct audio_stream *stream) |
| { |
| ALOGI("out_standby()"); |
| struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); |
| struct submix_audio_device * const rsxadev = out->dev; |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| out->output_standby = true; |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| return 0; |
| } |
| |
| static int out_dump(const struct audio_stream *stream, int fd) |
| { |
| (void)stream; |
| (void)fd; |
| return 0; |
| } |
| |
| static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) |
| { |
| int exiting = -1; |
| AudioParameter parms = AudioParameter(String8(kvpairs)); |
| SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); |
| |
| // FIXME this is using hard-coded strings but in the future, this functionality will be |
| // converted to use audio HAL extensions required to support tunneling |
| if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { |
| struct submix_audio_device * const rsxadev = |
| audio_stream_get_submix_stream_out(stream)->dev; |
| pthread_mutex_lock(&rsxadev->lock); |
| { // using the sink |
| sp<MonoPipe> sink = |
| rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] |
| .rsxSink; |
| if (sink == NULL) { |
| pthread_mutex_unlock(&rsxadev->lock); |
| return 0; |
| } |
| |
| ALOGD("out_set_parameters(): shutting down MonoPipe sink"); |
| sink->shutdown(true); |
| } // done using the sink |
| pthread_mutex_unlock(&rsxadev->lock); |
| } |
| return 0; |
| } |
| |
| static char * out_get_parameters(const struct audio_stream *stream, const char *keys) |
| { |
| (void)stream; |
| (void)keys; |
| return strdup(""); |
| } |
| |
| static uint32_t out_get_latency(const struct audio_stream_out *stream) |
| { |
| const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( |
| const_cast<struct audio_stream_out *>(stream)); |
| const struct submix_config * const config = &out->dev->routes[out->route_handle].config; |
| const size_t stream_frame_size = |
| audio_stream_out_frame_size(stream); |
| const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( |
| &stream->common, config, config->buffer_size_frames, stream_frame_size); |
| const uint32_t sample_rate = out_get_sample_rate(&stream->common); |
| const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; |
| SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", |
| latency_ms, buffer_size_frames, sample_rate); |
| return latency_ms; |
| } |
| |
| static int out_set_volume(struct audio_stream_out *stream, float left, |
| float right) |
| { |
| (void)stream; |
| (void)left; |
| (void)right; |
| return -ENOSYS; |
| } |
| |
| static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, |
| size_t bytes) |
| { |
| SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); |
| ssize_t written_frames = 0; |
| const size_t frame_size = audio_stream_out_frame_size(stream); |
| struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); |
| struct submix_audio_device * const rsxadev = out->dev; |
| const size_t frames = bytes / frame_size; |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| out->output_standby = false; |
| |
| sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink; |
| if (sink != NULL) { |
| if (sink->isShutdown()) { |
| sink.clear(); |
| pthread_mutex_unlock(&rsxadev->lock); |
| SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); |
| // the pipe has already been shutdown, this buffer will be lost but we must |
| // simulate timing so we don't drain the output faster than realtime |
| usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); |
| return bytes; |
| } |
| } else { |
| pthread_mutex_unlock(&rsxadev->lock); |
| ALOGE("out_write without a pipe!"); |
| ALOG_ASSERT("out_write without a pipe!"); |
| return 0; |
| } |
| |
| // If the write to the sink would block when no input stream is present, flush enough frames |
| // from the pipe to make space to write the most recent data. |
| { |
| const size_t availableToWrite = sink->availableToWrite(); |
| sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource; |
| if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { |
| static uint8_t flush_buffer[64]; |
| const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; |
| size_t frames_to_flush_from_source = frames - availableToWrite; |
| SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", |
| frames_to_flush_from_source); |
| while (frames_to_flush_from_source) { |
| const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); |
| frames_to_flush_from_source -= flush_size; |
| // read does not block |
| source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); |
| } |
| } |
| } |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| written_frames = sink->write(buffer, frames); |
| |
| #if LOG_STREAMS_TO_FILES |
| if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); |
| #endif // LOG_STREAMS_TO_FILES |
| |
| if (written_frames < 0) { |
| if (written_frames == (ssize_t)NEGOTIATE) { |
| ALOGE("out_write() write to pipe returned NEGOTIATE"); |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| sink.clear(); |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| written_frames = 0; |
| return 0; |
| } else { |
| // write() returned UNDERRUN or WOULD_BLOCK, retry |
| ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); |
| written_frames = sink->write(buffer, frames); |
| } |
| } |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| sink.clear(); |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| if (written_frames < 0) { |
| ALOGE("out_write() failed writing to pipe with %zd", written_frames); |
| return 0; |
| } |
| const ssize_t written_bytes = written_frames * frame_size; |
| SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); |
| return written_bytes; |
| } |
| |
| static int out_get_render_position(const struct audio_stream_out *stream, |
| uint32_t *dsp_frames) |
| { |
| (void)stream; |
| (void)dsp_frames; |
| return -EINVAL; |
| } |
| |
| static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| { |
| (void)stream; |
| (void)effect; |
| return 0; |
| } |
| |
| static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| { |
| (void)stream; |
| (void)effect; |
| return 0; |
| } |
| |
| static int out_get_next_write_timestamp(const struct audio_stream_out *stream, |
| int64_t *timestamp) |
| { |
| (void)stream; |
| (void)timestamp; |
| return -EINVAL; |
| } |
| |
| /** audio_stream_in implementation **/ |
| static uint32_t in_get_sample_rate(const struct audio_stream *stream) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( |
| const_cast<struct audio_stream*>(stream)); |
| #if ENABLE_RESAMPLING |
| const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; |
| #else |
| const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; |
| #endif // ENABLE_RESAMPLING |
| SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); |
| return rate; |
| } |
| |
| static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); |
| #if ENABLE_RESAMPLING |
| // The sample rate of the stream can't be changed once it's set since this would change the |
| // input buffer size and hence break recording from the shared pipe. |
| if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { |
| ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " |
| "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); |
| return -ENOSYS; |
| } |
| #endif // ENABLE_RESAMPLING |
| if (!sample_rate_supported(rate)) { |
| ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); |
| return -ENOSYS; |
| } |
| in->dev->routes[in->route_handle].config.common.sample_rate = rate; |
| SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); |
| return 0; |
| } |
| |
| static size_t in_get_buffer_size(const struct audio_stream *stream) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( |
| const_cast<struct audio_stream*>(stream)); |
| const struct submix_config * const config = &in->dev->routes[in->route_handle].config; |
| const size_t stream_frame_size = |
| audio_stream_in_frame_size((const struct audio_stream_in *)stream); |
| size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( |
| stream, config, config->buffer_period_size_frames, stream_frame_size); |
| #if ENABLE_RESAMPLING |
| // Scale the size of the buffer based upon the maximum number of frames that could be returned |
| // given the ratio of output to input sample rate. |
| buffer_size_frames = (size_t)(((float)buffer_size_frames * |
| (float)config->input_sample_rate) / |
| (float)config->output_sample_rate); |
| #endif // ENABLE_RESAMPLING |
| const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; |
| SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, |
| buffer_size_frames); |
| return buffer_size_bytes; |
| } |
| |
| static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( |
| const_cast<struct audio_stream*>(stream)); |
| const audio_channel_mask_t channel_mask = |
| in->dev->routes[in->route_handle].config.input_channel_mask; |
| SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); |
| return channel_mask; |
| } |
| |
| static audio_format_t in_get_format(const struct audio_stream *stream) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( |
| const_cast<struct audio_stream*>(stream)); |
| const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; |
| SUBMIX_ALOGV("in_get_format() returns %x", format); |
| return format; |
| } |
| |
| static int in_set_format(struct audio_stream *stream, audio_format_t format) |
| { |
| const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); |
| if (format != in->dev->routes[in->route_handle].config.common.format) { |
| ALOGE("in_set_format(format=%x) format unsupported", format); |
| return -ENOSYS; |
| } |
| SUBMIX_ALOGV("in_set_format(format=%x)", format); |
| return 0; |
| } |
| |
| static int in_standby(struct audio_stream *stream) |
| { |
| ALOGI("in_standby()"); |
| struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); |
| struct submix_audio_device * const rsxadev = in->dev; |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| in->input_standby = true; |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| return 0; |
| } |
| |
| static int in_dump(const struct audio_stream *stream, int fd) |
| { |
| (void)stream; |
| (void)fd; |
| return 0; |
| } |
| |
| static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) |
| { |
| (void)stream; |
| (void)kvpairs; |
| return 0; |
| } |
| |
| static char * in_get_parameters(const struct audio_stream *stream, |
| const char *keys) |
| { |
| (void)stream; |
| (void)keys; |
| return strdup(""); |
| } |
| |
| static int in_set_gain(struct audio_stream_in *stream, float gain) |
| { |
| (void)stream; |
| (void)gain; |
| return 0; |
| } |
| |
| static ssize_t in_read(struct audio_stream_in *stream, void* buffer, |
| size_t bytes) |
| { |
| struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); |
| struct submix_audio_device * const rsxadev = in->dev; |
| struct audio_config *format; |
| const size_t frame_size = audio_stream_in_frame_size(stream); |
| const size_t frames_to_read = bytes / frame_size; |
| |
| SUBMIX_ALOGV("in_read bytes=%zu", bytes); |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| const bool output_standby = rsxadev->routes[in->route_handle].output == NULL |
| ? true : rsxadev->routes[in->route_handle].output->output_standby; |
| const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); |
| in->output_standby_rec_thr = output_standby; |
| |
| if (in->input_standby || output_standby_transition) { |
| in->input_standby = false; |
| // keep track of when we exit input standby (== first read == start "real recording") |
| // or when we start recording silence, and reset projected time |
| int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); |
| if (rc == 0) { |
| in->read_counter_frames = 0; |
| } |
| } |
| |
| in->read_counter_frames += frames_to_read; |
| size_t remaining_frames = frames_to_read; |
| |
| { |
| // about to read from audio source |
| sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource; |
| if (source == NULL) { |
| in->read_error_count++;// ok if it rolls over |
| ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, |
| "no audio pipe yet we're trying to read! (not all errors will be logged)"); |
| pthread_mutex_unlock(&rsxadev->lock); |
| usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); |
| memset(buffer, 0, bytes); |
| return bytes; |
| } |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| |
| // read the data from the pipe (it's non blocking) |
| int attempts = 0; |
| char* buff = (char*)buffer; |
| #if ENABLE_CHANNEL_CONVERSION |
| // Determine whether channel conversion is required. |
| const uint32_t input_channels = audio_channel_count_from_in_mask( |
| rsxadev->routes[in->route_handle].config.input_channel_mask); |
| const uint32_t output_channels = audio_channel_count_from_out_mask( |
| rsxadev->routes[in->route_handle].config.output_channel_mask); |
| if (input_channels != output_channels) { |
| SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " |
| "input channels", output_channels, input_channels); |
| // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. |
| ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == |
| AUDIO_FORMAT_PCM_16_BIT); |
| ALOG_ASSERT((input_channels == 1 && output_channels == 2) || |
| (input_channels == 2 && output_channels == 1)); |
| } |
| #endif // ENABLE_CHANNEL_CONVERSION |
| |
| #if ENABLE_RESAMPLING |
| const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); |
| const uint32_t output_sample_rate = |
| rsxadev->routes[in->route_handle].config.output_sample_rate; |
| const size_t resampler_buffer_size_frames = |
| sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / |
| sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); |
| float resampler_ratio = 1.0f; |
| // Determine whether resampling is required. |
| if (input_sample_rate != output_sample_rate) { |
| resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; |
| // Only support 16-bit PCM mono resampling. |
| // NOTE: Resampling is performed after the channel conversion step. |
| ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == |
| AUDIO_FORMAT_PCM_16_BIT); |
| ALOG_ASSERT(audio_channel_count_from_in_mask( |
| rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); |
| } |
| #endif // ENABLE_RESAMPLING |
| |
| while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { |
| ssize_t frames_read = -1977; |
| size_t read_frames = remaining_frames; |
| #if ENABLE_RESAMPLING |
| char* const saved_buff = buff; |
| if (resampler_ratio != 1.0f) { |
| // Calculate the number of frames from the pipe that need to be read to generate |
| // the data for the input stream read. |
| const size_t frames_required_for_resampler = (size_t)( |
| (float)read_frames * (float)resampler_ratio); |
| read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); |
| // Read into the resampler buffer. |
| buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; |
| } |
| #endif // ENABLE_RESAMPLING |
| #if ENABLE_CHANNEL_CONVERSION |
| if (output_channels == 1 && input_channels == 2) { |
| // Need to read half the requested frames since the converted output |
| // data will take twice the space (mono->stereo). |
| read_frames /= 2; |
| } |
| #endif // ENABLE_CHANNEL_CONVERSION |
| |
| SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); |
| |
| frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); |
| |
| SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); |
| |
| #if ENABLE_CHANNEL_CONVERSION |
| // Perform in-place channel conversion. |
| // NOTE: In the following "input stream" refers to the data returned by this function |
| // and "output stream" refers to the data read from the pipe. |
| if (input_channels != output_channels && frames_read > 0) { |
| int16_t *data = (int16_t*)buff; |
| if (output_channels == 2 && input_channels == 1) { |
| // Offset into the output stream data in samples. |
| ssize_t output_stream_offset = 0; |
| for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; |
| input_stream_frame++, output_stream_offset += 2) { |
| // Average the content from both channels. |
| data[input_stream_frame] = ((int32_t)data[output_stream_offset] + |
| (int32_t)data[output_stream_offset + 1]) / 2; |
| } |
| } else if (output_channels == 1 && input_channels == 2) { |
| // Offset into the input stream data in samples. |
| ssize_t input_stream_offset = (frames_read - 1) * 2; |
| for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; |
| output_stream_frame--, input_stream_offset -= 2) { |
| const short sample = data[output_stream_frame]; |
| data[input_stream_offset] = sample; |
| data[input_stream_offset + 1] = sample; |
| } |
| } |
| } |
| #endif // ENABLE_CHANNEL_CONVERSION |
| |
| #if ENABLE_RESAMPLING |
| if (resampler_ratio != 1.0f) { |
| SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); |
| const int16_t * const data = (int16_t*)buff; |
| int16_t * const resampled_buffer = (int16_t*)saved_buff; |
| // Resample with *no* filtering - if the data from the ouptut stream was really |
| // sampled at a different rate this will result in very nasty aliasing. |
| const float output_stream_frames = (float)frames_read; |
| size_t input_stream_frame = 0; |
| for (float output_stream_frame = 0.0f; |
| output_stream_frame < output_stream_frames && |
| input_stream_frame < remaining_frames; |
| output_stream_frame += resampler_ratio, input_stream_frame++) { |
| resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; |
| } |
| ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); |
| SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); |
| frames_read = input_stream_frame; |
| buff = saved_buff; |
| } |
| #endif // ENABLE_RESAMPLING |
| |
| if (frames_read > 0) { |
| #if LOG_STREAMS_TO_FILES |
| if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); |
| #endif // LOG_STREAMS_TO_FILES |
| |
| remaining_frames -= frames_read; |
| buff += frames_read * frame_size; |
| SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", |
| attempts, frames_read, remaining_frames); |
| } else { |
| attempts++; |
| SUBMIX_ALOGE(" in_read read returned %zd", frames_read); |
| usleep(READ_ATTEMPT_SLEEP_MS * 1000); |
| } |
| } |
| // done using the source |
| pthread_mutex_lock(&rsxadev->lock); |
| source.clear(); |
| pthread_mutex_unlock(&rsxadev->lock); |
| } |
| |
| if (remaining_frames > 0) { |
| const size_t remaining_bytes = remaining_frames * frame_size; |
| SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); |
| memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); |
| } |
| |
| // compute how much we need to sleep after reading the data by comparing the wall clock with |
| // the projected time at which we should return. |
| struct timespec time_after_read;// wall clock after reading from the pipe |
| struct timespec record_duration;// observed record duration |
| int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); |
| const uint32_t sample_rate = in_get_sample_rate(&stream->common); |
| if (rc == 0) { |
| // for how long have we been recording? |
| record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; |
| record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; |
| if (record_duration.tv_nsec < 0) { |
| record_duration.tv_sec--; |
| record_duration.tv_nsec += 1000000000; |
| } |
| |
| // read_counter_frames contains the number of frames that have been read since the |
| // beginning of recording (including this call): it's converted to usec and compared to |
| // how long we've been recording for, which gives us how long we must wait to sync the |
| // projected recording time, and the observed recording time. |
| long projected_vs_observed_offset_us = |
| ((int64_t)(in->read_counter_frames |
| - (record_duration.tv_sec*sample_rate))) |
| * 1000000 / sample_rate |
| - (record_duration.tv_nsec / 1000); |
| |
| SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", |
| record_duration.tv_sec, record_duration.tv_nsec/1000000, |
| projected_vs_observed_offset_us); |
| if (projected_vs_observed_offset_us > 0) { |
| usleep(projected_vs_observed_offset_us); |
| } |
| } |
| |
| SUBMIX_ALOGV("in_read returns %zu", bytes); |
| return bytes; |
| |
| } |
| |
| static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) |
| { |
| (void)stream; |
| return 0; |
| } |
| |
| static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| { |
| (void)stream; |
| (void)effect; |
| return 0; |
| } |
| |
| static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) |
| { |
| (void)stream; |
| (void)effect; |
| return 0; |
| } |
| |
| static int adev_open_output_stream(struct audio_hw_device *dev, |
| audio_io_handle_t handle, |
| audio_devices_t devices, |
| audio_output_flags_t flags, |
| struct audio_config *config, |
| struct audio_stream_out **stream_out, |
| const char *address) |
| { |
| struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); |
| ALOGD("adev_open_output_stream(address=%s)", address); |
| struct submix_stream_out *out; |
| bool force_pipe_creation = false; |
| (void)handle; |
| (void)devices; |
| (void)flags; |
| |
| *stream_out = NULL; |
| |
| // Make sure it's possible to open the device given the current audio config. |
| submix_sanitize_config(config, false); |
| |
| int route_idx = -1; |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); |
| if (res != OK) { |
| ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); |
| pthread_mutex_unlock(&rsxadev->lock); |
| return res; |
| } |
| |
| if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { |
| ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); |
| pthread_mutex_unlock(&rsxadev->lock); |
| return -EINVAL; |
| } |
| |
| out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); |
| if (!out) { |
| pthread_mutex_unlock(&rsxadev->lock); |
| return -ENOMEM; |
| } |
| |
| // Initialize the function pointer tables (v-tables). |
| out->stream.common.get_sample_rate = out_get_sample_rate; |
| out->stream.common.set_sample_rate = out_set_sample_rate; |
| out->stream.common.get_buffer_size = out_get_buffer_size; |
| out->stream.common.get_channels = out_get_channels; |
| out->stream.common.get_format = out_get_format; |
| out->stream.common.set_format = out_set_format; |
| out->stream.common.standby = out_standby; |
| out->stream.common.dump = out_dump; |
| out->stream.common.set_parameters = out_set_parameters; |
| out->stream.common.get_parameters = out_get_parameters; |
| out->stream.common.add_audio_effect = out_add_audio_effect; |
| out->stream.common.remove_audio_effect = out_remove_audio_effect; |
| out->stream.get_latency = out_get_latency; |
| out->stream.set_volume = out_set_volume; |
| out->stream.write = out_write; |
| out->stream.get_render_position = out_get_render_position; |
| out->stream.get_next_write_timestamp = out_get_next_write_timestamp; |
| |
| #if ENABLE_RESAMPLING |
| // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits |
| // writes correctly. |
| force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate |
| != config->sample_rate; |
| #endif // ENABLE_RESAMPLING |
| |
| // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so |
| // that it's recreated. |
| if ((rsxadev->routes[route_idx].rsxSink != NULL |
| && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { |
| submix_audio_device_release_pipe_l(rsxadev, route_idx); |
| } |
| |
| // Store a pointer to the device from the output stream. |
| out->dev = rsxadev; |
| // Initialize the pipe. |
| ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); |
| submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, |
| DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); |
| #if LOG_STREAMS_TO_FILES |
| out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, |
| LOG_STREAM_FILE_PERMISSIONS); |
| ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", |
| strerror(errno)); |
| ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); |
| #endif // LOG_STREAMS_TO_FILES |
| // Return the output stream. |
| *stream_out = &out->stream; |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| return 0; |
| } |
| |
| static void adev_close_output_stream(struct audio_hw_device *dev, |
| struct audio_stream_out *stream) |
| { |
| struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( |
| const_cast<struct audio_hw_device*>(dev)); |
| struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); |
| submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); |
| #if LOG_STREAMS_TO_FILES |
| if (out->log_fd >= 0) close(out->log_fd); |
| #endif // LOG_STREAMS_TO_FILES |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| free(out); |
| } |
| |
| static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) |
| { |
| (void)dev; |
| (void)kvpairs; |
| return -ENOSYS; |
| } |
| |
| static char * adev_get_parameters(const struct audio_hw_device *dev, |
| const char *keys) |
| { |
| (void)dev; |
| (void)keys; |
| return strdup("");; |
| } |
| |
| static int adev_init_check(const struct audio_hw_device *dev) |
| { |
| ALOGI("adev_init_check()"); |
| (void)dev; |
| return 0; |
| } |
| |
| static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) |
| { |
| (void)dev; |
| (void)volume; |
| return -ENOSYS; |
| } |
| |
| static int adev_set_master_volume(struct audio_hw_device *dev, float volume) |
| { |
| (void)dev; |
| (void)volume; |
| return -ENOSYS; |
| } |
| |
| static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) |
| { |
| (void)dev; |
| (void)volume; |
| return -ENOSYS; |
| } |
| |
| static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) |
| { |
| (void)dev; |
| (void)muted; |
| return -ENOSYS; |
| } |
| |
| static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) |
| { |
| (void)dev; |
| (void)muted; |
| return -ENOSYS; |
| } |
| |
| static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) |
| { |
| (void)dev; |
| (void)mode; |
| return 0; |
| } |
| |
| static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) |
| { |
| (void)dev; |
| (void)state; |
| return -ENOSYS; |
| } |
| |
| static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) |
| { |
| (void)dev; |
| (void)state; |
| return -ENOSYS; |
| } |
| |
| static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, |
| const struct audio_config *config) |
| { |
| if (audio_is_linear_pcm(config->format)) { |
| size_t max_buffer_period_size_frames = 0; |
| struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( |
| const_cast<struct audio_hw_device*>(dev)); |
| // look for the largest buffer period size |
| for (int i = 0 ; i < MAX_ROUTES ; i++) { |
| if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) |
| { |
| max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; |
| } |
| } |
| const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * |
| audio_bytes_per_sample(config->format); |
| const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; |
| SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", |
| buffer_size, buffer_period_size_frames); |
| return buffer_size; |
| } |
| return 0; |
| } |
| |
| static int adev_open_input_stream(struct audio_hw_device *dev, |
| audio_io_handle_t handle, |
| audio_devices_t devices, |
| struct audio_config *config, |
| struct audio_stream_in **stream_in, |
| audio_input_flags_t flags __unused, |
| const char *address, |
| audio_source_t source __unused) |
| { |
| struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); |
| struct submix_stream_in *in; |
| ALOGD("adev_open_input_stream(addr=%s)", address); |
| (void)handle; |
| (void)devices; |
| |
| *stream_in = NULL; |
| |
| // Do we already have a route for this address |
| int route_idx = -1; |
| |
| pthread_mutex_lock(&rsxadev->lock); |
| |
| status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); |
| if (res != OK) { |
| ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); |
| pthread_mutex_unlock(&rsxadev->lock); |
| return res; |
| } |
| |
| // Make sure it's possible to open the device given the current audio config. |
| submix_sanitize_config(config, true); |
| if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { |
| ALOGE("adev_open_input_stream(): Unable to open input stream."); |
| pthread_mutex_unlock(&rsxadev->lock); |
| return -EINVAL; |
| } |
| |
| #if ENABLE_LEGACY_INPUT_OPEN |
| in = rsxadev->routes[route_idx].input; |
| if (in) { |
| in->ref_count++; |
| sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; |
| ALOG_ASSERT(sink != NULL); |
| // If the sink has been shutdown, delete the pipe. |
| if (sink != NULL) { |
| if (sink->isShutdown()) { |
| ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", |
| in->ref_count); |
| submix_audio_device_release_pipe_l(rsxadev, in->route_handle); |
| } else { |
| ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); |
| } |
| } else { |
| ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); |
| } |
| } |
| #else |
| in = NULL; |
| #endif // ENABLE_LEGACY_INPUT_OPEN |
| |
| if (!in) { |
| in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); |
| if (!in) return -ENOMEM; |
| in->ref_count = 1; |
| |
| // Initialize the function pointer tables (v-tables). |
| in->stream.common.get_sample_rate = in_get_sample_rate; |
| in->stream.common.set_sample_rate = in_set_sample_rate; |
| in->stream.common.get_buffer_size = in_get_buffer_size; |
| in->stream.common.get_channels = in_get_channels; |
| in->stream.common.get_format = in_get_format; |
| in->stream.common.set_format = in_set_format; |
| in->stream.common.standby = in_standby; |
| in->stream.common.dump = in_dump; |
| in->stream.common.set_parameters = in_set_parameters; |
| in->stream.common.get_parameters = in_get_parameters; |
| in->stream.common.add_audio_effect = in_add_audio_effect; |
| in->stream.common.remove_audio_effect = in_remove_audio_effect; |
| in->stream.set_gain = in_set_gain; |
| in->stream.read = in_read; |
| in->stream.get_input_frames_lost = in_get_input_frames_lost; |
| |
| in->dev = rsxadev; |
| #if LOG_STREAMS_TO_FILES |
| in->log_fd = -1; |
| #endif |
| } |
| |
| // Initialize the input stream. |
| in->read_counter_frames = 0; |
| in->input_standby = true; |
| if (rsxadev->routes[route_idx].output != NULL) { |
| in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; |
| } else { |
| in->output_standby_rec_thr = true; |
| } |
| |
| in->read_error_count = 0; |
| // Initialize the pipe. |
| ALOGV("adev_open_input_stream(): about to create pipe"); |
| submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, |
| DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); |
| #if LOG_STREAMS_TO_FILES |
| if (in->log_fd >= 0) close(in->log_fd); |
| in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, |
| LOG_STREAM_FILE_PERMISSIONS); |
| ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", |
| strerror(errno)); |
| ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); |
| #endif // LOG_STREAMS_TO_FILES |
| // Return the input stream. |
| *stream_in = &in->stream; |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| return 0; |
| } |
| |
| static void adev_close_input_stream(struct audio_hw_device *dev, |
| struct audio_stream_in *stream) |
| { |
| struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); |
| |
| struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); |
| ALOGD("adev_close_input_stream()"); |
| pthread_mutex_lock(&rsxadev->lock); |
| submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); |
| #if LOG_STREAMS_TO_FILES |
| if (in->log_fd >= 0) close(in->log_fd); |
| #endif // LOG_STREAMS_TO_FILES |
| #if ENABLE_LEGACY_INPUT_OPEN |
| if (in->ref_count == 0) free(in); |
| #else |
| free(in); |
| #endif // ENABLE_LEGACY_INPUT_OPEN |
| |
| pthread_mutex_unlock(&rsxadev->lock); |
| } |
| |
| static int adev_dump(const audio_hw_device_t *device, int fd) |
| { |
| const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); |
| reinterpret_cast<const struct submix_audio_device *>( |
| reinterpret_cast<const uint8_t *>(device) - |
| offsetof(struct submix_audio_device, device)); |
| char msg[100]; |
| int n = sprintf(msg, "\nReroute submix audio module:\n"); |
| write(fd, &msg, n); |
| for (int i=0 ; i < MAX_ROUTES ; i++) { |
| n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i, |
| rsxadev->routes[i].config.input_sample_rate, |
| rsxadev->routes[i].config.output_sample_rate, |
| rsxadev->routes[i].address); |
| write(fd, &msg, n); |
| } |
| return 0; |
| } |
| |
| static int adev_close(hw_device_t *device) |
| { |
| ALOGI("adev_close()"); |
| free(device); |
| return 0; |
| } |
| |
| static int adev_open(const hw_module_t* module, const char* name, |
| hw_device_t** device) |
| { |
| ALOGI("adev_open(name=%s)", name); |
| struct submix_audio_device *rsxadev; |
| |
| if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) |
| return -EINVAL; |
| |
| rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); |
| if (!rsxadev) |
| return -ENOMEM; |
| |
| rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; |
| rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; |
| rsxadev->device.common.module = (struct hw_module_t *) module; |
| rsxadev->device.common.close = adev_close; |
| |
| rsxadev->device.init_check = adev_init_check; |
| rsxadev->device.set_voice_volume = adev_set_voice_volume; |
| rsxadev->device.set_master_volume = adev_set_master_volume; |
| rsxadev->device.get_master_volume = adev_get_master_volume; |
| rsxadev->device.set_master_mute = adev_set_master_mute; |
| rsxadev->device.get_master_mute = adev_get_master_mute; |
| rsxadev->device.set_mode = adev_set_mode; |
| rsxadev->device.set_mic_mute = adev_set_mic_mute; |
| rsxadev->device.get_mic_mute = adev_get_mic_mute; |
| rsxadev->device.set_parameters = adev_set_parameters; |
| rsxadev->device.get_parameters = adev_get_parameters; |
| rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; |
| rsxadev->device.open_output_stream = adev_open_output_stream; |
| rsxadev->device.close_output_stream = adev_close_output_stream; |
| rsxadev->device.open_input_stream = adev_open_input_stream; |
| rsxadev->device.close_input_stream = adev_close_input_stream; |
| rsxadev->device.dump = adev_dump; |
| |
| for (int i=0 ; i < MAX_ROUTES ; i++) { |
| memset(&rsxadev->routes[i], 0, sizeof(route_config)); |
| strcpy(rsxadev->routes[i].address, ""); |
| } |
| |
| *device = &rsxadev->device.common; |
| |
| return 0; |
| } |
| |
| static struct hw_module_methods_t hal_module_methods = { |
| /* open */ adev_open, |
| }; |
| |
| struct audio_module HAL_MODULE_INFO_SYM = { |
| /* common */ { |
| /* tag */ HARDWARE_MODULE_TAG, |
| /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, |
| /* hal_api_version */ HARDWARE_HAL_API_VERSION, |
| /* id */ AUDIO_HARDWARE_MODULE_ID, |
| /* name */ "Wifi Display audio HAL", |
| /* author */ "The Android Open Source Project", |
| /* methods */ &hal_module_methods, |
| /* dso */ NULL, |
| /* reserved */ { 0 }, |
| }, |
| }; |
| |
| } //namespace android |
| |
| } //extern "C" |