Remove FrameCombiner stats
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}
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WebRTC LUCI CQ

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@ -164,8 +164,6 @@ void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
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AudioFrame* audio_frame_for_mixing) {
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RTC_DCHECK(audio_frame_for_mixing);
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LogMixingStats(mix_list, sample_rate, number_of_streams);
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SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
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number_of_streams, audio_frame_for_mixing);
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@ -212,32 +210,4 @@ void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
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InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
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}
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void FrameCombiner::LogMixingStats(
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rtc::ArrayView<const AudioFrame* const> mix_list,
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int sample_rate,
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size_t number_of_streams) const {
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// Log every second.
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uma_logging_counter_++;
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if (uma_logging_counter_ > 1000 / AudioMixerImpl::kFrameDurationInMs) {
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uma_logging_counter_ = 0;
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RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.AudioMixer.NumIncomingStreams",
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static_cast<int>(number_of_streams));
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RTC_HISTOGRAM_COUNTS_LINEAR(
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"WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2",
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rtc::dchecked_cast<int>(mix_list.size()), /*min=*/1, /*max=*/16,
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/*bucket_count=*/16);
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using NativeRate = AudioProcessing::NativeRate;
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static constexpr NativeRate native_rates[] = {
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NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
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NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
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const auto* rate_position = std::lower_bound(
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std::begin(native_rates), std::end(native_rates), sample_rate);
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RTC_HISTOGRAM_ENUMERATION(
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"WebRTC.Audio.AudioMixer.MixingRate",
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std::distance(std::begin(native_rates), rate_position),
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arraysize(native_rates));
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}
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}
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} // namespace webrtc
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@ -47,15 +47,10 @@ class FrameCombiner {
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kMaximumNumberOfChannels>;
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private:
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void LogMixingStats(rtc::ArrayView<const AudioFrame* const> mix_list,
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int sample_rate,
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size_t number_of_streams) const;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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std::unique_ptr<MixingBuffer> mixing_buffer_;
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Limiter limiter_;
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const bool use_limiter_;
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mutable int uma_logging_counter_ = 0;
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};
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} // namespace webrtc
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