From 6aa755c2014510fb9afd5bcf664e9a7aaa51de04 Mon Sep 17 00:00:00 2001
From: Alessio Bazzica <alessiob@webrtc.org>
Date: Wed, 16 Nov 2022 14:23:41 +0100
Subject: [PATCH] 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}
---
 modules/audio_mixer/frame_combiner.cc | 30 ---------------------------
 modules/audio_mixer/frame_combiner.h  |  5 -----
 2 files changed, 35 deletions(-)

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<AudioFrame* const> 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<AudioFrame* const> mix_list,
   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
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<const AudioFrame* const> mix_list,
-                      int sample_rate,
-                      size_t number_of_streams) const;
-
   std::unique_ptr<ApmDataDumper> data_dumper_;
   std::unique_ptr<MixingBuffer> mixing_buffer_;
   Limiter limiter_;
   const bool use_limiter_;
-  mutable int uma_logging_counter_ = 0;
 };
 }  // namespace webrtc