/* * 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 #include #include "api/audio/audio_mixer.h" #include "modules/audio_mixer/frame_combiner.h" #include "modules/audio_mixer/output_rate_calculator.h" #include "modules/audio_processing/include/audio_processing.h" #include "modules/include/module_common_types.h" #include "rtc_base/race_checker.h" #include "rtc_base/scoped_ref_ptr.h" #include "rtc_base/thread_annotations.h" #include "typedefs.h" // NOLINT(build/include) namespace webrtc { typedef std::vector 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>; // AudioProcessing only accepts 10 ms frames. static const int kFrameDurationInMs = 10; static const int kMaximumAmountOfMixedAudioSources = 3; static rtc::scoped_refptr Create(); static rtc::scoped_refptr Create( std::unique_ptr 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(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; protected: AudioMixerImpl(std::unique_ptr 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(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 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(crit_); // 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_