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This adds the histogram WebRTC.Audio.AudioMixer.NewHighestSourceCount which logs the highest number of sources an AudioMixer has had. The statistic is logged whenever the highest number of sources increases. This allows us to differentiate the statistic to see how many times the mixer has had a certain maximum number of sources. Chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4414896 Bug: chromium:1430806 Change-Id: Iab92e201a0b667741cc8f3bbbed92fa989d7fcda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300860 Reviewed-by: Olga Sharonova <olka@webrtc.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39833}
278 lines
9.5 KiB
C++
278 lines
9.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_mixer/audio_mixer_impl.h"
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#include <stdint.h>
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#include <algorithm>
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#include <iterator>
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#include <type_traits>
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#include <utility>
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#include "modules/audio_mixer/audio_frame_manipulator.h"
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#include "modules/audio_mixer/default_output_rate_calculator.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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struct AudioMixerImpl::SourceStatus {
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SourceStatus(Source* audio_source, bool is_mixed, float gain)
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: audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
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Source* audio_source = nullptr;
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bool is_mixed = false;
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float gain = 0.0f;
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// A frame that will be passed to audio_source->GetAudioFrameWithInfo.
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AudioFrame audio_frame;
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};
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namespace {
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struct SourceFrame {
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SourceFrame() = default;
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SourceFrame(AudioMixerImpl::SourceStatus* source_status,
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AudioFrame* audio_frame,
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bool muted)
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: source_status(source_status), audio_frame(audio_frame), muted(muted) {
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RTC_DCHECK(source_status);
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RTC_DCHECK(audio_frame);
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if (!muted) {
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energy = AudioMixerCalculateEnergy(*audio_frame);
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}
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}
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SourceFrame(AudioMixerImpl::SourceStatus* source_status,
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AudioFrame* audio_frame,
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bool muted,
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uint32_t energy)
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: source_status(source_status),
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audio_frame(audio_frame),
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muted(muted),
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energy(energy) {
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RTC_DCHECK(source_status);
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RTC_DCHECK(audio_frame);
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}
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AudioMixerImpl::SourceStatus* source_status = nullptr;
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AudioFrame* audio_frame = nullptr;
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bool muted = true;
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uint32_t energy = 0;
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};
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// ShouldMixBefore(a, b) is used to select mixer sources.
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// Returns true if `a` is preferred over `b` as a source to be mixed.
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bool ShouldMixBefore(const SourceFrame& a, const SourceFrame& b) {
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if (a.muted != b.muted) {
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return b.muted;
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}
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const auto a_activity = a.audio_frame->vad_activity_;
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const auto b_activity = b.audio_frame->vad_activity_;
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if (a_activity != b_activity) {
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return a_activity == AudioFrame::kVadActive;
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}
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return a.energy > b.energy;
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}
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void RampAndUpdateGain(
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rtc::ArrayView<const SourceFrame> mixed_sources_and_frames) {
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for (const auto& source_frame : mixed_sources_and_frames) {
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float target_gain = source_frame.source_status->is_mixed ? 1.0f : 0.0f;
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Ramp(source_frame.source_status->gain, target_gain,
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source_frame.audio_frame);
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source_frame.source_status->gain = target_gain;
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}
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}
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std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>>::const_iterator
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FindSourceInList(
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AudioMixerImpl::Source const* audio_source,
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std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>> const*
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audio_source_list) {
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return std::find_if(
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audio_source_list->begin(), audio_source_list->end(),
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[audio_source](const std::unique_ptr<AudioMixerImpl::SourceStatus>& p) {
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return p->audio_source == audio_source;
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});
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}
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} // namespace
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struct AudioMixerImpl::HelperContainers {
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void resize(size_t size) {
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audio_to_mix.resize(size);
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audio_source_mixing_data_list.resize(size);
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ramp_list.resize(size);
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preferred_rates.resize(size);
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}
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std::vector<AudioFrame*> audio_to_mix;
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std::vector<SourceFrame> audio_source_mixing_data_list;
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std::vector<SourceFrame> ramp_list;
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std::vector<int> preferred_rates;
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};
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AudioMixerImpl::AudioMixerImpl(
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std::unique_ptr<OutputRateCalculator> output_rate_calculator,
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bool use_limiter,
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int max_sources_to_mix)
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: max_sources_to_mix_(max_sources_to_mix),
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output_rate_calculator_(std::move(output_rate_calculator)),
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audio_source_list_(),
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helper_containers_(std::make_unique<HelperContainers>()),
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frame_combiner_(use_limiter) {
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RTC_CHECK_GE(max_sources_to_mix, 1) << "At least one source must be mixed";
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audio_source_list_.reserve(max_sources_to_mix);
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helper_containers_->resize(max_sources_to_mix);
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}
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AudioMixerImpl::~AudioMixerImpl() {}
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rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
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int max_sources_to_mix) {
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return Create(std::unique_ptr<DefaultOutputRateCalculator>(
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new DefaultOutputRateCalculator()),
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/*use_limiter=*/true, max_sources_to_mix);
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}
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rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
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std::unique_ptr<OutputRateCalculator> output_rate_calculator,
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bool use_limiter,
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int max_sources_to_mix) {
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return rtc::make_ref_counted<AudioMixerImpl>(
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std::move(output_rate_calculator), use_limiter, max_sources_to_mix);
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}
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void AudioMixerImpl::Mix(size_t number_of_channels,
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AudioFrame* audio_frame_for_mixing) {
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TRACE_EVENT0("webrtc", "AudioMixerImpl::Mix");
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RTC_DCHECK(number_of_channels >= 1);
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MutexLock lock(&mutex_);
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size_t number_of_streams = audio_source_list_.size();
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std::transform(audio_source_list_.begin(), audio_source_list_.end(),
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helper_containers_->preferred_rates.begin(),
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[&](std::unique_ptr<SourceStatus>& a) {
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return a->audio_source->PreferredSampleRate();
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});
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int output_frequency = output_rate_calculator_->CalculateOutputRateFromRange(
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rtc::ArrayView<const int>(helper_containers_->preferred_rates.data(),
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number_of_streams));
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frame_combiner_.Combine(GetAudioFromSources(output_frequency),
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number_of_channels, output_frequency,
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number_of_streams, audio_frame_for_mixing);
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}
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bool AudioMixerImpl::AddSource(Source* audio_source) {
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RTC_DCHECK(audio_source);
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MutexLock lock(&mutex_);
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RTC_DCHECK(FindSourceInList(audio_source, &audio_source_list_) ==
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audio_source_list_.end())
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<< "Source already added to mixer";
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audio_source_list_.emplace_back(new SourceStatus(audio_source, false, 0));
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helper_containers_->resize(audio_source_list_.size());
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UpdateSourceCountStats();
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return true;
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}
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void AudioMixerImpl::RemoveSource(Source* audio_source) {
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RTC_DCHECK(audio_source);
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MutexLock lock(&mutex_);
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const auto iter = FindSourceInList(audio_source, &audio_source_list_);
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RTC_DCHECK(iter != audio_source_list_.end()) << "Source not present in mixer";
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audio_source_list_.erase(iter);
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}
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rtc::ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
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int output_frequency) {
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// Get audio from the audio sources and put it in the SourceFrame vector.
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int audio_source_mixing_data_count = 0;
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for (auto& source_and_status : audio_source_list_) {
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const auto audio_frame_info =
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source_and_status->audio_source->GetAudioFrameWithInfo(
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output_frequency, &source_and_status->audio_frame);
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if (audio_frame_info == Source::AudioFrameInfo::kError) {
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RTC_LOG_F(LS_WARNING) << "failed to GetAudioFrameWithInfo() from source";
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continue;
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}
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helper_containers_
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->audio_source_mixing_data_list[audio_source_mixing_data_count++] =
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SourceFrame(source_and_status.get(), &source_and_status->audio_frame,
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audio_frame_info == Source::AudioFrameInfo::kMuted);
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}
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rtc::ArrayView<SourceFrame> audio_source_mixing_data_view(
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helper_containers_->audio_source_mixing_data_list.data(),
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audio_source_mixing_data_count);
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// Sort frames by sorting function.
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std::sort(audio_source_mixing_data_view.begin(),
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audio_source_mixing_data_view.end(), ShouldMixBefore);
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int max_audio_frame_counter = max_sources_to_mix_;
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int ramp_list_lengh = 0;
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int audio_to_mix_count = 0;
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// Go through list in order and put unmuted frames in result list.
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for (const auto& p : audio_source_mixing_data_view) {
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// Filter muted.
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if (p.muted) {
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p.source_status->is_mixed = false;
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continue;
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}
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// Add frame to result vector for mixing.
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bool is_mixed = false;
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if (max_audio_frame_counter > 0) {
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--max_audio_frame_counter;
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helper_containers_->audio_to_mix[audio_to_mix_count++] = p.audio_frame;
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helper_containers_->ramp_list[ramp_list_lengh++] =
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SourceFrame(p.source_status, p.audio_frame, false, -1);
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is_mixed = true;
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}
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p.source_status->is_mixed = is_mixed;
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}
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RampAndUpdateGain(rtc::ArrayView<SourceFrame>(
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helper_containers_->ramp_list.data(), ramp_list_lengh));
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return rtc::ArrayView<AudioFrame* const>(
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helper_containers_->audio_to_mix.data(), audio_to_mix_count);
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}
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bool AudioMixerImpl::GetAudioSourceMixabilityStatusForTest(
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AudioMixerImpl::Source* audio_source) const {
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MutexLock lock(&mutex_);
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const auto iter = FindSourceInList(audio_source, &audio_source_list_);
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if (iter != audio_source_list_.end()) {
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return (*iter)->is_mixed;
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}
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RTC_LOG(LS_ERROR) << "Audio source unknown";
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return false;
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}
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void AudioMixerImpl::UpdateSourceCountStats() {
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size_t current_source_count = audio_source_list_.size();
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// Log to the histogram whenever the maximum number of sources increases.
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if (current_source_count > max_source_count_ever_) {
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RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AudioMixer.NewHighestSourceCount",
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current_source_count, 1, 20, 20);
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max_source_count_ever_ = current_source_count;
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}
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}
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} // namespace webrtc
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