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Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.

Browse Source 
Bug: webrtc:11567
Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31649}
This commit is contained in:
Markus Handell
2020-07-07 15:53:34 +02:00
committed by Commit Bot
parent 1e257cacbf
commit 0df0faefd5
16 changed files with 314 additions and 312 deletions
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64
modules/audio_processing/audio_processing_impl.cc
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@ -313,8 +313,8 @@ AudioProcessingImpl::~AudioProcessingImpl() = default;
int AudioProcessingImpl::Initialize() {
// Run in a single-threaded manner during initialization.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
return InitializeLocked();
}
@ -340,8 +340,8 @@ int AudioProcessingImpl::Initialize(int capture_input_sample_rate_hz,
int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
// Run in a single-threaded manner during initialization.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
return InitializeLocked(processing_config);
}
@ -352,7 +352,7 @@ int AudioProcessingImpl::MaybeInitializeRender(
return kNoError;
}
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
return InitializeLocked(processing_config);
}
@ -526,8 +526,8 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
// Run in a single-threaded manner when applying the settings.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
const bool pipeline_config_changed =
config_.pipeline.multi_channel_render !=
@ -643,7 +643,7 @@ void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {}
void AudioProcessingImpl::OverrideSubmoduleCreationForTesting(
const ApmSubmoduleCreationOverrides& overrides) {
rtc::CritScope cs(&crit_capture_);
MutexLock lock(&mutex_capture_);
submodule_creation_overrides_ = overrides;
}
@ -689,7 +689,7 @@ size_t AudioProcessingImpl::num_output_channels() const {
}
void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
rtc::CritScope cs(&crit_capture_);
MutexLock lock(&mutex_capture_);
capture_.output_will_be_muted = muted;
if (submodules_.agc_manager.get()) {
submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted);
@ -751,7 +751,7 @@ int AudioProcessingImpl::MaybeInitializeCapture(
// Acquire the capture lock in order to access api_format. The lock is
// released immediately, as we may need to acquire the render lock as part
// of the conditional reinitialization.
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
@ -767,8 +767,8 @@ int AudioProcessingImpl::MaybeInitializeCapture(
}
if (reinitialization_required) {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
return kNoError;
@ -785,7 +785,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
if (aec_dump_) {
RecordUnprocessedCaptureStream(src);
@ -991,7 +991,7 @@ void AudioProcessingImpl::AllocateRenderQueue() {
}
void AudioProcessingImpl::EmptyQueuedRenderAudio() {
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
EmptyQueuedRenderAudioLocked();
}
@ -1023,7 +1023,7 @@ int AudioProcessingImpl::ProcessStream(const int16_t* const src,
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
if (aec_dump_) {
RecordUnprocessedCaptureStream(src, input_config);
@ -1320,7 +1320,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(
const float* const* data,
const StreamConfig& reverse_config) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_StreamConfig");
rtc::CritScope cs(&crit_render_);
MutexLock lock(&mutex_render_);
return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
}
@ -1329,7 +1329,7 @@ int AudioProcessingImpl::ProcessReverseStream(const float* const* src,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
rtc::CritScope cs(&crit_render_);
MutexLock lock(&mutex_render_);
RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, input_config, output_config));
if (submodule_states_.RenderMultiBandProcessingActive() ||
submodule_states_.RenderFullBandProcessingActive()) {
@ -1390,7 +1390,7 @@ int AudioProcessingImpl::ProcessReverseStream(const int16_t* const src,
return AudioProcessing::Error::kBadNumberChannelsError;
}
rtc::CritScope cs(&crit_render_);
MutexLock lock(&mutex_render_);
ProcessingConfig processing_config = formats_.api_format;
processing_config.reverse_input_stream().set_sample_rate_hz(
input_config.sample_rate_hz());
@ -1457,7 +1457,7 @@ int AudioProcessingImpl::ProcessRenderStreamLocked() {
}
int AudioProcessingImpl::set_stream_delay_ms(int delay) {
rtc::CritScope cs(&crit_capture_);
MutexLock lock(&mutex_capture_);
Error retval = kNoError;
capture_.was_stream_delay_set = true;
@ -1478,7 +1478,7 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
bool AudioProcessingImpl::GetLinearAecOutput(
rtc::ArrayView<std::array<float, 160>> linear_output) const {
rtc::CritScope cs(&crit_capture_);
MutexLock lock(&mutex_capture_);
AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
RTC_DCHECK(linear_aec_buffer);
@ -1507,12 +1507,12 @@ int AudioProcessingImpl::stream_delay_ms() const {
}
void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
rtc::CritScope cs(&crit_capture_);
MutexLock lock(&mutex_capture_);
capture_.key_pressed = key_pressed;
}
void AudioProcessingImpl::set_stream_analog_level(int level) {
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
if (submodules_.agc_manager) {
submodules_.agc_manager->set_stream_analog_level(level);
@ -1527,7 +1527,7 @@ void AudioProcessingImpl::set_stream_analog_level(int level) {
}
int AudioProcessingImpl::recommended_stream_analog_level() const {
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_capture(&mutex_capture_);
return recommended_stream_analog_level_locked();
}
@ -1569,8 +1569,8 @@ bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle,
void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
RTC_DCHECK(aec_dump);
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
// The previously attached AecDump will be destroyed with the
// 'aec_dump' parameter, which is after locks are released.
@ -1585,23 +1585,23 @@ void AudioProcessingImpl::DetachAecDump() {
// the render and capture locks.
std::unique_ptr<AecDump> aec_dump = nullptr;
{
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
aec_dump = std::move(aec_dump_);
}
}
void AudioProcessingImpl::MutateConfig(
rtc::FunctionView<void(AudioProcessing::Config*)> mutator) {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
mutator(&config_);
ApplyConfig(config_);
}
AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
return config_;
}
@ -2054,7 +2054,7 @@ AudioProcessingImpl::ApmStatsReporter::ApmStatsReporter()
AudioProcessingImpl::ApmStatsReporter::~ApmStatsReporter() = default;
AudioProcessingStats AudioProcessingImpl::ApmStatsReporter::GetStatistics() {
rtc::CritScope cs_stats(&crit_stats_);
MutexLock lock_stats(&mutex_stats_);
bool new_stats_available = stats_message_queue_.Remove(&cached_stats_);
// If the message queue is full, return the cached stats.
static_cast<void>(new_stats_available);