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To verify the upcoming code changes it is required

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that the level of the output in the audio processing
module is monitored. This CL adds that.

BUG=webrtc:6181, webrtc:6183, webrtc:6220

Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}
This commit is contained in:
peah 2016-12-20 13:45:58 -08:00 committed by Commit bot
parent 0838327ec9
commit 1b08dc33eb
2 changed files with 19 additions and 6 deletions
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20
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -1132,12 +1132,13 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity.
rms_.Analyze(rtc::ArrayView<const int16_t>(
capture_input_rms_.Analyze(rtc::ArrayView<const int16_t>(
capture_buffer->channels_const()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
if (++rms_interval_counter_ >= 1000) {
rms_interval_counter_ = 0;
RmsLevel::Levels levels = rms_.AverageAndPeak();
const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
if (log_rms) {
capture_rms_interval_counter_ = 0;
RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
levels.average, 1, RmsLevel::kMinLevelDb, 64);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
@ -1275,6 +1276,17 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
// The level estimator operates on the recombined data.
public_submodules_->level_estimator->ProcessStream(capture_buffer);
capture_output_rms_.Analyze(rtc::ArrayView<const int16_t>(
capture_buffer->channels_const()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
if (log_rms) {
RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelAverageRms",
levels.average, 1, RmsLevel::kMinLevelDb, 64);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelPeakRms",
levels.peak, 1, RmsLevel::kMinLevelDb, 64);
}
capture_.was_stream_delay_set = false;
return kNoError;
}

5
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -415,8 +415,9 @@ class AudioProcessingImpl : public AudioProcessing {
std::vector<float> red_render_queue_buffer_ GUARDED_BY(crit_render_);
std::vector<float> red_capture_queue_buffer_ GUARDED_BY(crit_capture_);
RmsLevel rms_ GUARDED_BY(crit_capture_);
int rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
RmsLevel capture_input_rms_ GUARDED_BY(crit_capture_);
RmsLevel capture_output_rms_ GUARDED_BY(crit_capture_);
int capture_rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
// Lock protection not needed.
std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>