zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove FrameCombiner stats

Browse Source 
Stop logging WebRTC.Audio.AudioMixer.* histograms.

Bug: chromium:1308711, chromium:1328289
Change-Id: Iba1c89a112842c532d99900cd54aee7f38f759fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38651}
This commit is contained in:
Alessio Bazzica
2022-11-16 14:23:41 +01:00
committed by WebRTC LUCI CQ
parent b301b58b3f
commit 6aa755c201
2 changed files with 0 additions and 35 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
modules/audio_mixer/frame_combiner.cc
View File

@ -164,8 +164,6 @@ void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
AudioFrame* audio_frame_for_mixing) { AudioFrame* audio_frame_for_mixing) {
RTC_DCHECK(audio_frame_for_mixing); RTC_DCHECK(audio_frame_for_mixing);
LogMixingStats(mix_list, sample_rate, number_of_streams);
SetAudioFrameFields(mix_list, number_of_channels, sample_rate, SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
number_of_streams, audio_frame_for_mixing); number_of_streams, audio_frame_for_mixing);
@ -212,32 +210,4 @@ void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing); InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
} }
void FrameCombiner::LogMixingStats(
rtc::ArrayView<const AudioFrame* const> mix_list,
int sample_rate,
size_t number_of_streams) const {
// Log every second.
uma_logging_counter_++;
if (uma_logging_counter_ > 1000 / AudioMixerImpl::kFrameDurationInMs) {
uma_logging_counter_ = 0;
RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.AudioMixer.NumIncomingStreams",
static_cast<int>(number_of_streams));
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2",
rtc::dchecked_cast<int>(mix_list.size()), /*min=*/1, /*max=*/16,
/*bucket_count=*/16);
using NativeRate = AudioProcessing::NativeRate;
static constexpr NativeRate native_rates[] = {
NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
const auto* rate_position = std::lower_bound(
std::begin(native_rates), std::end(native_rates), sample_rate);
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.AudioMixer.MixingRate",
std::distance(std::begin(native_rates), rate_position),
arraysize(native_rates));
}
}
} // namespace webrtc } // namespace webrtc

5
modules/audio_mixer/frame_combiner.h
View File

@ -47,15 +47,10 @@ class FrameCombiner {
kMaximumNumberOfChannels>; kMaximumNumberOfChannels>;
private: private:
void LogMixingStats(rtc::ArrayView<const AudioFrame* const> mix_list,
int sample_rate,
size_t number_of_streams) const;
std::unique_ptr<ApmDataDumper> data_dumper_; std::unique_ptr<ApmDataDumper> data_dumper_;
std::unique_ptr<MixingBuffer> mixing_buffer_; std::unique_ptr<MixingBuffer> mixing_buffer_;
Limiter limiter_; Limiter limiter_;
const bool use_limiter_; const bool use_limiter_;
mutable int uma_logging_counter_ = 0;
}; };
} // namespace webrtc } // namespace webrtc