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Merge the preambles of the ProcessStream implementations

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The two functions have a lot of shared logic and locking. This CL consolidates that into a single function.

Bug: webrtc:111235
Change-Id: Ib1c32165dbf0e212c7d4b0753bcbb5ffd05eb6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163022
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30144}
This commit is contained in:
Sam Zackrisson
2020-01-03 14:54:20 +01:00
committed by Commit Bot
parent 0f14db22de
commit 12e319aafe
4 changed files with 51 additions and 93 deletions
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106
modules/audio_processing/audio_processing_impl.cc
View File

@ -746,8 +746,7 @@ void AudioProcessingImpl::ApplyAgc1Config(
}
// TODO(webrtc:5298): Remove.
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
}
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {}
int AudioProcessingImpl::proc_sample_rate_hz() const {
// Used as callback from submodules, hence locking is not allowed.
@ -844,28 +843,16 @@ void AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
int AudioProcessingImpl::MaybeInitializeCapture(
const StreamConfig& input_config,
const StreamConfig& output_config) {
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Acquire the capture lock in order to:
// - Safely call the function that retrieves the render side data. This
// function accesses APM getters that need the capture lock held when
// being called.
// - Access api_format. The lock is released immediately due to the
// conditional reinitialization.
// 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_);
EmptyQueuedRenderAudio();
if (!src || !dest) {
return kNullPointerError;
}
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
}
@ -881,15 +868,25 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
}
if (reinitialization_required) {
// Reinitialize.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
return kNoError;
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
if (!src || !dest) {
return kNullPointerError;
}
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
RTC_DCHECK_EQ(processing_config.input_stream().num_frames(),
formats_.api_format.input_stream().num_frames());
if (aec_dump_) {
RecordUnprocessedCaptureStream(src);
@ -1114,64 +1111,18 @@ void AudioProcessingImpl::EmptyQueuedRenderAudio() {
int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
ProcessingConfig processing_config;
bool reinitialization_required = false;
{
// Acquire the capture lock in order to:
// - Safely call the function that retrieves the render side data. This
// function accesses APM getters that need the capture lock held when
// being called.
// - Access api_format. The lock is released immediately due to the
// conditional reinitialization.
rtc::CritScope cs_capture(&crit_capture_);
EmptyQueuedRenderAudio();
if (!frame) {
return kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != kSampleRate8kHz &&
frame->sample_rate_hz_ != kSampleRate16kHz &&
frame->sample_rate_hz_ != kSampleRate32kHz &&
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
// TODO(ajm): The input and output rates and channels are currently
// constrained to be identical in the int16 interface.
processing_config = formats_.api_format;
reinitialization_required = UpdateActiveSubmoduleStates();
if (!frame) {
return kNullPointerError;
}
reinitialization_required =
reinitialization_required ||
processing_config.input_stream().sample_rate_hz() !=
frame->sample_rate_hz_ ||
processing_config.input_stream().num_channels() != frame->num_channels_ ||
processing_config.output_stream().sample_rate_hz() !=
frame->sample_rate_hz_ ||
processing_config.output_stream().num_channels() != frame->num_channels_;
if (reinitialization_required) {
processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.input_stream().set_num_channels(frame->num_channels_);
processing_config.output_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.output_stream().set_num_channels(frame->num_channels_);
// Reinitialize.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
RETURN_ON_ERR(InitializeLocked(processing_config));
}
StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
RETURN_ON_ERR(MaybeInitializeCapture(input_config, output_config));
rtc::CritScope cs_capture(&crit_capture_);
if (frame->samples_per_channel_ !=
formats_.api_format.input_stream().num_frames()) {
return kBadDataLengthError;
}
if (aec_dump_) {
RecordUnprocessedCaptureStream(*frame);
@ -1204,6 +1155,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
}
int AudioProcessingImpl::ProcessCaptureStreamLocked() {
EmptyQueuedRenderAudio();
HandleCaptureRuntimeSettings();
// Ensure that not both the AEC and AECM are active at the same time.

19
modules/audio_processing/audio_processing_impl.h
View File

@ -211,14 +211,18 @@ class AudioProcessingImpl : public AudioProcessing {
bool first_update_ = true;
};
// Method for modifying the formats struct that are called from both
// the render and capture threads. The check for whether modifications
// are needed is done while holding the render lock only, thereby avoiding
// that the capture thread blocks the render thread.
// The struct is modified in a single-threaded manner by holding both the
// render and capture locks.
// Methods for modifying the formats struct that is used by both
// the render and capture threads. The check for whether modifications are
// needed is done while holding a single lock only, thereby avoiding that the
// capture thread blocks the render thread.
// Called by render: Holds the render lock when reading the format struct and
// acquires both locks if reinitialization is required.
int MaybeInitializeRender(const ProcessingConfig& processing_config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
// Called by capture: Holds the capture lock when reading the format struct
// and acquires both locks if reinitialization is needed.
int MaybeInitializeCapture(const StreamConfig& input_config,
const StreamConfig& output_config);
// Method for updating the state keeping track of the active submodules.
// Returns a bool indicating whether the state has changed.
@ -473,9 +477,6 @@ class AudioProcessingImpl : public AudioProcessing {
SwapQueue<AudioProcessingStats> stats_message_queue_;
} stats_reporter_;
std::vector<float> aec_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<float> aec_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);
std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<int16_t> aecm_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);

9
modules/audio_processing/audio_processing_impl_unittest.cc
View File

@ -378,6 +378,15 @@ TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
constexpr int16_t kAudioLevel = 1000;
constexpr int kSampleRateHz = 16000;
constexpr size_t kNumChannels = 1;
// Explicitly initialize APM to ensure no render frames are discarded.
const ProcessingConfig processing_config = {{
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
}};
apm->Initialize(processing_config);
AudioFrame frame;
InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame);

10
modules/audio_processing/audio_processing_unittest.cc
View File

@ -832,13 +832,9 @@ TEST_F(ApmTest, Channels) {
}
TEST_F(ApmTest, SampleRatesInt) {
// Testing invalid sample rates
SetContainerFormat(10000, 2, &frame_, &float_cb_);
EXPECT_EQ(apm_->kBadSampleRateError, ProcessStreamChooser(kIntFormat));
// Testing valid sample rates
int fs[] = {8000, 16000, 32000, 48000};
for (size_t i = 0; i < arraysize(fs); i++) {
SetContainerFormat(fs[i], 2, &frame_, &float_cb_);
// Testing some valid sample rates.
for (int sample_rate : {8000, 12000, 16000, 32000, 44100, 48000, 96000}) {
SetContainerFormat(sample_rate, 2, &frame_, &float_cb_);
EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
}
}