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Use ProcessReverseStream in VoiceEngines OutputMixer

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Review URL: https://codereview.webrtc.org/1776363002

Cr-Commit-Position: refs/heads/master@{#12044}
This commit is contained in:
aluebs 2016-03-17 16:43:29 -07:00 committed by Commit bot
parent 56d4d059ac
commit da116c4c37
5 changed files with 3 additions and 23 deletions
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1
webrtc/media/engine/fakewebrtcvoiceengine.h
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@ -57,7 +57,6 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
}
WEBRTC_STUB_CONST(input_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
size_t num_input_channels() const override { return 0; }

8
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -419,12 +419,6 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
#endif // WEBRTC_ANDROID_PLATFORM_BUILD
}
int AudioProcessingImpl::input_sample_rate_hz() const {
// Accessed from outside APM, hence a lock is needed.
rtc::CritScope cs(&crit_capture_);
return formats_.api_format.input_stream().sample_rate_hz();
}
int AudioProcessingImpl::proc_sample_rate_hz() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
@ -842,8 +836,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
RETURN_ON_ERR(AnalyzeReverseStream(frame));
rtc::CritScope cs(&crit_render_);
RETURN_ON_ERR(AnalyzeReverseStream(frame));
if (is_rev_processed()) {
render_.render_audio->InterleaveTo(frame, true);
}

1
webrtc/modules/audio_processing/audio_processing_impl.h
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@ -83,7 +83,6 @@ class AudioProcessingImpl : public AudioProcessing {
void set_delay_offset_ms(int offset) override;
int delay_offset_ms() const override;
void set_stream_key_pressed(bool key_pressed) override;
int input_sample_rate_hz() const override;
// Render-side exclusive methods possibly running APM in a
// multi-threaded manner. Acquire the render lock.

7
webrtc/modules/audio_processing/include/audio_processing.h
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@ -296,10 +296,6 @@ class AudioProcessing {
// ensures the options are applied immediately.
virtual void SetExtraOptions(const Config& config) = 0;
// TODO(peah): Remove after voice engine no longer requires it to resample
// the reverse stream to the forward rate.
virtual int input_sample_rate_hz() const = 0;
// TODO(ajm): Only intended for internal use. Make private and friend the
// necessary classes?
virtual int proc_sample_rate_hz() const = 0;
@ -367,8 +363,7 @@ class AudioProcessing {
// chances are you don't need to use it.
//
// The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_|
// members of |frame| must be valid. |sample_rate_hz_| must correspond to
// |input_sample_rate_hz()|
// members of |frame| must be valid.
//
// TODO(ajm): add const to input; requires an implementation fix.
// DEPRECATED: Use |ProcessReverseStream| instead.

9
webrtc/voice_engine/output_mixer.cc
View File

@ -492,14 +492,7 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
// --- Far-end Voice Quality Enhancement (AudioProcessing Module)
if (feed_data_to_apm) {
// Convert from mixing to AudioProcessing sample rate, similarly to how it
// is done on the send side. Downmix to mono.
AudioFrame frame;
frame.num_channels_ = 1;
frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz();
RemixAndResample(_audioFrame, &audioproc_resampler_, &frame);
if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) != 0) {
if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"AudioProcessingModule::AnalyzeReverseStream() => error");
RTC_DCHECK(false);