blob: 831948701800d6aa0e8f47208f685363dcd67b28 [file] [log] [blame] [edit]
/*
* 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/array_view.h"
#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/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class AudioMixerImpl : public AudioMixer {
public:
struct SourceStatus;
// AudioProcessing only accepts 10 ms frames.
static const int kFrameDurationInMs = 10;
static rtc::scoped_refptr<AudioMixerImpl> Create();
static rtc::scoped_refptr<AudioMixerImpl> Create(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter);
~AudioMixerImpl() override;
AudioMixerImpl(const AudioMixerImpl&) = delete;
AudioMixerImpl& operator=(const AudioMixerImpl&) = delete;
// 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_);
protected:
AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter);
private:
struct HelperContainers;
void UpdateSourceCountStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Fetches audio frames to mix from sources.
rtc::ArrayView<AudioFrame* const> GetAudioFromSources(int output_frequency)
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_;
std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
// List of all audio sources.
std::vector<std::unique_ptr<SourceStatus>> audio_source_list_
RTC_GUARDED_BY(mutex_);
const std::unique_ptr<HelperContainers> helper_containers_
RTC_GUARDED_BY(mutex_);
// Component that handles actual adding of audio frames.
FrameCombiner frame_combiner_;
// The highest source count this mixer has ever had. Used for UMA stats.
size_t max_source_count_ever_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_