From 1b08dc33ebcdb48f951258cf75d97dff0e60b4f1 Mon Sep 17 00:00:00 2001 From: peah Date: Tue, 20 Dec 2016 13:45:58 -0800 Subject: [PATCH] To verify the upcoming code changes it is required 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} --- .../audio_processing/audio_processing_impl.cc | 20 +++++++++++++++---- .../audio_processing/audio_processing_impl.h | 5 +++-- 2 files changed, 19 insertions(+), 6 deletions(-) diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index ce71461b2e..3a8f7f5463 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -1132,12 +1132,13 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() { AudioBuffer* capture_buffer = capture_.capture_audio.get(); // For brevity. - rms_.Analyze(rtc::ArrayView( + capture_input_rms_.Analyze(rtc::ArrayView( 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( + 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; } diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h index ae4d89b131..01b640fcfc 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.h +++ b/webrtc/modules/audio_processing/audio_processing_impl.h @@ -415,8 +415,9 @@ class AudioProcessingImpl : public AudioProcessing { std::vector red_render_queue_buffer_ GUARDED_BY(crit_render_); std::vector 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, RenderQueueItemVerifier>>