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