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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_mixer/audio_mixer_impl.h"
#include <stdint.h>
#include <algorithm>
#include <iterator>
#include <type_traits>
#include <utility>
#include "modules/audio_mixer/audio_frame_manipulator.h"
#include "modules/audio_mixer/default_output_rate_calculator.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
struct AudioMixerImpl::SourceStatus {
explicit SourceStatus(Source* audio_source) : audio_source(audio_source) {}
Source* audio_source = nullptr;
// A frame that will be passed to audio_source->GetAudioFrameWithInfo.
AudioFrame audio_frame;
};
namespace {
std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>>::const_iterator
FindSourceInList(
AudioMixerImpl::Source const* audio_source,
std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>> const*
audio_source_list) {
return std::find_if(
audio_source_list->begin(), audio_source_list->end(),
[audio_source](const std::unique_ptr<AudioMixerImpl::SourceStatus>& p) {
return p->audio_source == audio_source;
});
}
} // namespace
struct AudioMixerImpl::HelperContainers {
void resize(size_t size) {
audio_to_mix.resize(size);
preferred_rates.resize(size);
}
std::vector<AudioFrame*> audio_to_mix;
std::vector<int> preferred_rates;
};
AudioMixerImpl::AudioMixerImpl(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter)
: output_rate_calculator_(std::move(output_rate_calculator)),
audio_source_list_(),
helper_containers_(std::make_unique<HelperContainers>()),
frame_combiner_(use_limiter) {}
AudioMixerImpl::~AudioMixerImpl() {}
scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create() {
return Create(std::unique_ptr<DefaultOutputRateCalculator>(
new DefaultOutputRateCalculator()),
/*use_limiter=*/true);
}
scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter) {
return make_ref_counted<AudioMixerImpl>(std::move(output_rate_calculator),
use_limiter);
}
void AudioMixerImpl::Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) {
TRACE_EVENT0("webrtc", "AudioMixerImpl::Mix");
RTC_DCHECK(number_of_channels >= 1);
MutexLock lock(&mutex_);
size_t number_of_streams = audio_source_list_.size();
std::transform(audio_source_list_.begin(), audio_source_list_.end(),
helper_containers_->preferred_rates.begin(),
[&](std::unique_ptr<SourceStatus>& a) {
return a->audio_source->PreferredSampleRate();
});
int output_frequency = output_rate_calculator_->CalculateOutputRateFromRange(
ArrayView<const int>(helper_containers_->preferred_rates.data(),
number_of_streams));
frame_combiner_.Combine(GetAudioFromSources(output_frequency),
number_of_channels, output_frequency,
number_of_streams, audio_frame_for_mixing);
}
bool AudioMixerImpl::AddSource(Source* audio_source) {
RTC_DCHECK(audio_source);
MutexLock lock(&mutex_);
RTC_DCHECK(FindSourceInList(audio_source, &audio_source_list_) ==
audio_source_list_.end())
<< "Source already added to mixer";
audio_source_list_.emplace_back(new SourceStatus(audio_source));
helper_containers_->resize(audio_source_list_.size());
UpdateSourceCountStats();
return true;
}
void AudioMixerImpl::RemoveSource(Source* audio_source) {
RTC_DCHECK(audio_source);
MutexLock lock(&mutex_);
const auto iter = FindSourceInList(audio_source, &audio_source_list_);
RTC_DCHECK(iter != audio_source_list_.end()) << "Source not present in mixer";
audio_source_list_.erase(iter);
}
ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
int output_frequency) {
int audio_to_mix_count = 0;
for (auto& source_and_status : audio_source_list_) {
const auto audio_frame_info =
source_and_status->audio_source->GetAudioFrameWithInfo(
output_frequency, &source_and_status->audio_frame);
switch (audio_frame_info) {
case Source::AudioFrameInfo::kError:
RTC_LOG_F(LS_WARNING)
<< "failed to GetAudioFrameWithInfo() from source";
break;
case Source::AudioFrameInfo::kMuted:
break;
case Source::AudioFrameInfo::kNormal:
helper_containers_->audio_to_mix[audio_to_mix_count++] =
&source_and_status->audio_frame;
}
}
return ArrayView<AudioFrame* const>(helper_containers_->audio_to_mix.data(),
audio_to_mix_count);
}
void AudioMixerImpl::UpdateSourceCountStats() {
size_t current_source_count = audio_source_list_.size();
// Log to the histogram whenever the maximum number of sources increases.
if (current_source_count > max_source_count_ever_) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AudioMixer.NewHighestSourceCount",
current_source_count, 1, 20, 20);
max_source_count_ever_ = current_source_count;
}
}
} // namespace webrtc
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