| /* |
| * 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/constructor_magic.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; |
| 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: |
| struct HelperContainers; |
| |
| // Compute what audio sources to mix from audio_source_list_. Ramp |
| // in and out. Update mixed status. Mixes up to |
| // kMaximumAmountOfMixedAudioSources audio 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_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl); |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_ |