diff --git a/modules/audio_mixer/frame_combiner.cc b/modules/audio_mixer/frame_combiner.cc index abfc5e3ad9..96c62f6b0d 100644 --- a/modules/audio_mixer/frame_combiner.cc +++ b/modules/audio_mixer/frame_combiner.cc @@ -164,8 +164,6 @@ void FrameCombiner::Combine(rtc::ArrayView mix_list, AudioFrame* 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, number_of_streams, audio_frame_for_mixing); @@ -212,32 +210,4 @@ void FrameCombiner::Combine(rtc::ArrayView mix_list, InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing); } -void FrameCombiner::LogMixingStats( - rtc::ArrayView 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(number_of_streams)); - RTC_HISTOGRAM_COUNTS_LINEAR( - "WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2", - rtc::dchecked_cast(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 diff --git a/modules/audio_mixer/frame_combiner.h b/modules/audio_mixer/frame_combiner.h index 9ddf81e41e..4c858e1d99 100644 --- a/modules/audio_mixer/frame_combiner.h +++ b/modules/audio_mixer/frame_combiner.h @@ -47,15 +47,10 @@ class FrameCombiner { kMaximumNumberOfChannels>; private: - void LogMixingStats(rtc::ArrayView mix_list, - int sample_rate, - size_t number_of_streams) const; - std::unique_ptr data_dumper_; std::unique_ptr mixing_buffer_; Limiter limiter_; const bool use_limiter_; - mutable int uma_logging_counter_ = 0; }; } // namespace webrtc