blob: 93eea9816803932903697eabc61ae117253bbf0c [file] [log] [blame]
* 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 <memory>
#include <vector>
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/modules/audio_mixer/frame_combiner.h"
#include "webrtc/modules/audio_mixer/output_rate_calculator.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/race_checker.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread_annotations.h"
#include "webrtc/typedefs.h"
namespace webrtc {
typedef std::vector<AudioFrame*> AudioFrameList;
class AudioMixerImpl : public AudioMixer {
struct SourceStatus {
SourceStatus(Source* audio_source, bool is_mixed, float gain)
: audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
Source* audio_source = nullptr;
bool is_mixed = false;
float gain = 0.0f;
// A frame that will be passed to audio_source->GetAudioFrameWithInfo.
AudioFrame audio_frame;
using SourceStatusList = std::vector<std::unique_ptr<SourceStatus>>;
// AudioProcessing only accepts 10 ms frames.
static const int kFrameDurationInMs = 10;
static const int kMaximumAmountOfMixedAudioSources = 3;
static rtc::scoped_refptr<AudioMixerImpl> Create();
static rtc::scoped_refptr<AudioMixerImpl> Create(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter);
~AudioMixerImpl() override;
// AudioMixer functions
bool AddSource(Source* audio_source) override;
void RemoveSource(Source* audio_source) override;
void Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) override LOCKS_EXCLUDED(crit_);
// Returns true if the source was mixed last round. Returns
// false and logs an error if the source was never added to the
// mixer.
bool GetAudioSourceMixabilityStatusForTest(Source* audio_source) const;
AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter);
// Set mixing frequency through OutputFrequencyCalculator.
void CalculateOutputFrequency();
// Get mixing frequency.
int OutputFrequency() const;
// Compute what audio sources to mix from audio_source_list_. Ramp
// in and out. Update mixed status. Mixes up to
// kMaximumAmountOfMixedAudioSources audio sources.
AudioFrameList GetAudioFromSources() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Add/remove the MixerAudioSource to the specified
// MixerAudioSource list.
bool AddAudioSourceToList(Source* audio_source,
SourceStatusList* audio_source_list) const;
bool RemoveAudioSourceFromList(Source* remove_audio_source,
SourceStatusList* audio_source_list) const;
// The critical section lock guards audio source insertion and
// removal, which can be done from any thread. The race checker
// checks that mixing is done sequentially.
rtc::CriticalSection crit_;
rtc::RaceChecker race_checker_;
std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
// The current sample frequency and sample size when mixing.
int output_frequency_ GUARDED_BY(race_checker_);
size_t sample_size_ GUARDED_BY(race_checker_);
// List of all audio sources. Note all lists are disjunct
SourceStatusList audio_source_list_ GUARDED_BY(crit_); // May be mixed.
// Component that handles actual adding of audio frames.
FrameCombiner frame_combiner_ GUARDED_BY(race_checker_);
} // namespace webrtc