| /* | |
| * 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/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 scoped_refptr<AudioMixerImpl> Create(); | |
| static 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. | |
| 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_ |