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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.
*/
#ifndef MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
#define MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
#include <stddef.h>
#include <memory>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/audio/audio_mixer.h"
#include "api/scoped_refptr.h"
#include "modules/audio_mixer/frame_combiner.h"
#include "modules/audio_mixer/output_rate_calculator.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
typedef std::vector<AudioFrame*> AudioFrameList;
class AudioMixerImpl : public AudioMixer {
public:
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;
enum : 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
RTC_LOCKS_EXCLUDED(mutex_);
// 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;
protected:
AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter);
private:
// 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() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// 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.
mutable Mutex mutex_;
rtc::RaceChecker race_checker_;
std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
// The current sample frequency and sample size when mixing.
int output_frequency_ RTC_GUARDED_BY(race_checker_);
size_t sample_size_ RTC_GUARDED_BY(race_checker_);
// List of all audio sources. Note all lists are disjunct
SourceStatusList audio_source_list_ RTC_GUARDED_BY(mutex_); // May be mixed.
// Component that handles actual adding of audio frames.
FrameCombiner frame_combiner_ RTC_GUARDED_BY(race_checker_);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl);
};
} // namespace webrtc
#endif // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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