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Unified Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1422013002: Preparational work for an upcoming addition of a threadchecking scheme for APM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@bundling_of_state_CL
Patch Set: Fixed the final threadchecker refactoring issues (and merged from latest master) Created 5 years, 1 month ago
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« no previous file with comments | « webrtc/modules/audio_processing/audio_processing_impl.h ('k') | webrtc/modules/audio_processing/echo_cancellation_impl.h » ('j') | webrtc/modules/audio_processing/echo_cancellation_impl.cc » ('J')
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Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index f4977d4b01f8a507545dc46ccc17a16a1ea60c43..21629c53158a2fc800a735d4baf923a0e1c38003 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -223,13 +223,20 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
array_geometry_(config.Get<Beamforming>().array_geometry),
target_direction_(config.Get<Beamforming>().target_direction),
intelligibility_enabled_(config.Get<Intelligibility>().enabled) {
- echo_cancellation_ = new EchoCancellationImpl(this, crit_);
+ render_thread_checker_.DetachFromThread();
+ signal_thread_checker_.DetachFromThread();
+ capture_thread_checker_.DetachFromThread();
+
+ echo_cancellation_ =
+ new EchoCancellationImpl(this, crit_, &render_thread_checker_);
component_list_.push_back(echo_cancellation_);
- echo_control_mobile_ = new EchoControlMobileImpl(this, crit_);
+ echo_control_mobile_ =
+ new EchoControlMobileImpl(this, crit_, &render_thread_checker_);
component_list_.push_back(echo_control_mobile_);
- gain_control_ = new GainControlImpl(this, crit_);
+ gain_control_ = new GainControlImpl(this, crit_, &render_thread_checker_,
+ &capture_thread_checker_);
component_list_.push_back(gain_control_);
high_pass_filter_ = new HighPassFilterImpl(this, crit_);
@@ -274,6 +281,7 @@ AudioProcessingImpl::~AudioProcessingImpl() {
}
int AudioProcessingImpl::Initialize() {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
CriticalSectionScoped crit_scoped(crit_);
return InitializeLocked();
}
@@ -284,6 +292,7 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
const ProcessingConfig processing_config = {
{{input_sample_rate_hz,
ChannelsFromLayout(input_layout),
@@ -302,6 +311,7 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
}
int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
CriticalSectionScoped crit_scoped(crit_);
return InitializeLocked(processing_config);
}
@@ -310,6 +320,7 @@ int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
// their current values.
int AudioProcessingImpl::MaybeInitializeLocked(
const ProcessingConfig& processing_config) {
+ // Called from both threads. Thread check is therefore not possible.
the sun 2015/11/23 12:46:21 Uhm, yes it is possible: RTC_DCHECK(render_thread_
peah-webrtc 2015/11/23 13:49:40 Of course! Great! Added that! Done.
if (processing_config == shared_state_.api_format_) {
return kNoError;
}
@@ -379,6 +390,8 @@ int AudioProcessingImpl::InitializeLocked() {
}
int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
+ // Called from both render and capture threads, thread checks is therefore not
+ // possible.
for (const auto& stream : config.streams) {
if (stream.num_channels() < 0) {
return kBadNumberChannelsError;
@@ -453,9 +466,9 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
return InitializeLocked();
}
-
void AudioProcessingImpl::SetExtraOptions(const Config& config) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
for (auto item : component_list_) {
item->SetExtraOptions(config);
}
@@ -468,27 +481,33 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
int AudioProcessingImpl::proc_sample_rate_hz() const {
+ // TODO(peah): Add threadchecker when possible.
return fwd_proc_format_.sample_rate_hz();
}
int AudioProcessingImpl::proc_split_sample_rate_hz() const {
+ // TODO(peah): Add threadchecker when possible.
return split_rate_;
}
int AudioProcessingImpl::num_reverse_channels() const {
+ // TODO(peah): Add threadchecker when possible.
return rev_proc_format_.num_channels();
}
int AudioProcessingImpl::num_input_channels() const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
return shared_state_.api_format_.input_stream().num_channels();
}
int AudioProcessingImpl::num_output_channels() const {
+ // TODO(peah): Add appropriate thread checker when possible.
return shared_state_.api_format_.output_stream().num_channels();
}
void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
CriticalSectionScoped lock(crit_);
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
output_will_be_muted_ = muted;
if (agc_manager_.get()) {
agc_manager_->SetCaptureMuted(output_will_be_muted_);
@@ -504,6 +523,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
ChannelLayout output_layout,
float* const* dest) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
StreamConfig input_stream = shared_state_.api_format_.input_stream();
input_stream.set_sample_rate_hz(input_sample_rate_hz);
input_stream.set_num_channels(ChannelsFromLayout(input_layout));
@@ -525,6 +545,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& output_config,
float* const* dest) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
if (!src || !dest) {
return kNullPointerError;
}
@@ -576,6 +597,7 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
echo_cancellation_->ReadQueuedRenderData();
echo_control_mobile_->ReadQueuedRenderData();
gain_control_->ReadQueuedRenderData();
@@ -639,6 +661,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
}
int AudioProcessingImpl::ProcessStreamLocked() {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
audioproc::Stream* msg = event_msg_->mutable_stream();
@@ -720,6 +743,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int rev_sample_rate_hz,
ChannelLayout layout) {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
const StreamConfig reverse_config = {
rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
};
@@ -734,6 +758,7 @@ int AudioProcessingImpl::ProcessReverseStream(
const StreamConfig& reverse_input_config,
const StreamConfig& reverse_output_config,
float* const* dest) {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
RETURN_ON_ERR(
AnalyzeReverseStream(src, reverse_input_config, reverse_output_config));
if (is_rev_processed()) {
@@ -755,6 +780,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(
const StreamConfig& reverse_input_config,
const StreamConfig& reverse_output_config) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
if (src == NULL) {
return kNullPointerError;
}
@@ -792,6 +818,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(
}
int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
RETURN_ON_ERR(AnalyzeReverseStream(frame));
if (is_rev_processed()) {
render_audio_->InterleaveTo(frame, true);
@@ -801,6 +828,7 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
}
int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
CriticalSectionScoped crit_scoped(crit_);
if (frame == NULL) {
return kNullPointerError;
@@ -853,6 +881,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
}
int AudioProcessingImpl::ProcessReverseStreamLocked() {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
AudioBuffer* ra = render_audio_.get(); // For brevity.
if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
ra->SplitIntoFrequencyBands();
@@ -878,6 +907,7 @@ int AudioProcessingImpl::ProcessReverseStreamLocked() {
}
int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
Error retval = kNoError;
was_stream_delay_set_ = true;
delay += delay_offset_ms_;
@@ -898,29 +928,35 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
}
int AudioProcessingImpl::stream_delay_ms() const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
return stream_delay_ms_;
}
bool AudioProcessingImpl::was_stream_delay_set() const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
return was_stream_delay_set_;
}
void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
key_pressed_ = key_pressed;
}
void AudioProcessingImpl::set_delay_offset_ms(int offset) {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
CriticalSectionScoped crit_scoped(crit_);
delay_offset_ms_ = offset;
}
int AudioProcessingImpl::delay_offset_ms() const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
return delay_offset_ms_;
}
int AudioProcessingImpl::StartDebugRecording(
const char filename[AudioProcessing::kMaxFilenameSize]) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
if (filename == NULL) {
@@ -950,6 +986,7 @@ int AudioProcessingImpl::StartDebugRecording(
int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
if (handle == NULL) {
return kNullPointerError;
@@ -977,12 +1014,14 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
rtc::PlatformFile handle) {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
return StartDebugRecording(stream);
}
int AudioProcessingImpl::StopDebugRecording() {
CriticalSectionScoped crit_scoped(crit_);
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// We just return if recording hasn't started.
@@ -1029,6 +1068,7 @@ VoiceDetection* AudioProcessingImpl::voice_detection() const {
}
bool AudioProcessingImpl::is_data_processed() const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
if (beamformer_enabled_) {
return true;
}
@@ -1057,6 +1097,7 @@ bool AudioProcessingImpl::is_data_processed() const {
}
bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
// Check if we've upmixed or downmixed the audio.
return ((shared_state_.api_format_.output_stream().num_channels() !=
shared_state_.api_format_.input_stream().num_channels()) ||
@@ -1064,12 +1105,14 @@ bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
}
bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
return (is_data_processed &&
(fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz));
}
bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
if (!is_data_processed && !voice_detection_->is_enabled() &&
!transient_suppressor_enabled_) {
// Only level_estimator_ is enabled.
@@ -1083,15 +1126,18 @@ bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
}
bool AudioProcessingImpl::is_rev_processed() const {
+ RTC_DCHECK(render_thread_checker_.CalledOnValidThread());
return intelligibility_enabled_ && intelligibility_enhancer_->active();
}
bool AudioProcessingImpl::rev_conversion_needed() const {
+ // Called from several threads, thread check not possible.
return (shared_state_.api_format_.reverse_input_stream() !=
shared_state_.api_format_.reverse_output_stream());
}
void AudioProcessingImpl::InitializeExperimentalAgc() {
+ // Called from several threads, thread check not possible.
if (use_new_agc_) {
if (!agc_manager_.get()) {
agc_manager_.reset(new AgcManagerDirect(gain_control_,
@@ -1104,6 +1150,7 @@ void AudioProcessingImpl::InitializeExperimentalAgc() {
}
void AudioProcessingImpl::InitializeTransient() {
+ // Called from several threads, thread check not possible.
if (transient_suppressor_enabled_) {
if (!transient_suppressor_.get()) {
transient_suppressor_.reset(new TransientSuppressor());
@@ -1115,6 +1162,7 @@ void AudioProcessingImpl::InitializeTransient() {
}
void AudioProcessingImpl::InitializeBeamformer() {
+ // Called from several threads, thread check not possible.
if (beamformer_enabled_) {
if (!beamformer_) {
beamformer_.reset(
@@ -1125,6 +1173,7 @@ void AudioProcessingImpl::InitializeBeamformer() {
}
void AudioProcessingImpl::InitializeIntelligibility() {
+ // Called from several threads, thread check not possible.
if (intelligibility_enabled_) {
IntelligibilityEnhancer::Config config;
config.sample_rate_hz = split_rate_;
@@ -1135,6 +1184,7 @@ void AudioProcessingImpl::InitializeIntelligibility() {
}
void AudioProcessingImpl::MaybeUpdateHistograms() {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
static const int kMinDiffDelayMs = 60;
if (echo_cancellation()->is_enabled()) {
@@ -1181,6 +1231,7 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
}
void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
CriticalSectionScoped crit_scoped(crit_);
if (stream_delay_jumps_ > -1) {
RTC_HISTOGRAM_ENUMERATION(
@@ -1200,6 +1251,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
int AudioProcessingImpl::WriteMessageToDebugFile() {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
int32_t size = event_msg_->ByteSize();
if (size <= 0) {
return kUnspecifiedError;
@@ -1227,6 +1279,7 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
}
int AudioProcessingImpl::WriteInitMessage() {
+ // Called from both render and capture threads, not threadchecker possible.
event_msg_->set_type(audioproc::Event::INIT);
audioproc::Init* msg = event_msg_->mutable_init();
msg->set_sample_rate(
@@ -1248,6 +1301,7 @@ int AudioProcessingImpl::WriteInitMessage() {
}
int AudioProcessingImpl::WriteConfigMessage(bool forced) {
+ RTC_DCHECK(capture_thread_checker_.CalledOnValidThread());
audioproc::Config config;
config.set_aec_enabled(echo_cancellation_->is_enabled());
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