1066b1379d
AudioMixerImpl::CreateWithOutputRateCalculator has become deprecated. Instead, either Create() or Create(OutputRateCalculator, bool use_limiter) should be used. The first uses sensible default values for missing arguments. The second takes all arguments. The old CreateWithOutputRateCalculator is deprecated so that we don't have different Create:s with all possible combinations of parameters. Note that the factory methods may change in the future. The reason for adding 'use_limiter' was that the limiter that was used had questionable benefit and was very computationally expensive. Now work is going on to replace it with a much cheaper version. After the change, the factory method may change again to not allow for disabling the limiter. Bug: webrtc:7167 Change-Id: I0f9005e27e726fa552ee38dcbe965274e5006544 Reviewed-on: https://chromium-review.googlesource.com/528074 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18496}
114 lines
4.0 KiB
C++
114 lines
4.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio/audio_mixer.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/base/race_checker.h"
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#include "webrtc/modules/audio_mixer/frame_combiner.h"
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#include "webrtc/modules/audio_mixer/output_rate_calculator.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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typedef std::vector<AudioFrame*> AudioFrameList;
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class AudioMixerImpl : public AudioMixer {
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public:
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struct SourceStatus {
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SourceStatus(Source* audio_source, bool is_mixed, float gain)
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: audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
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Source* audio_source = nullptr;
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bool is_mixed = false;
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float gain = 0.0f;
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// A frame that will be passed to audio_source->GetAudioFrameWithInfo.
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AudioFrame audio_frame;
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};
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using SourceStatusList = std::vector<std::unique_ptr<SourceStatus>>;
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// AudioProcessing only accepts 10 ms frames.
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static const int kFrameDurationInMs = 10;
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static const int kMaximumAmountOfMixedAudioSources = 3;
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static rtc::scoped_refptr<AudioMixerImpl> Create();
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static rtc::scoped_refptr<AudioMixerImpl> Create(
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std::unique_ptr<OutputRateCalculator> output_rate_calculator,
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bool use_limiter);
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~AudioMixerImpl() override;
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// AudioMixer functions
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bool AddSource(Source* audio_source) override;
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void RemoveSource(Source* audio_source) override;
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void Mix(size_t number_of_channels,
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AudioFrame* audio_frame_for_mixing) override LOCKS_EXCLUDED(crit_);
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// Returns true if the source was mixed last round. Returns
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// false and logs an error if the source was never added to the
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// mixer.
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bool GetAudioSourceMixabilityStatusForTest(Source* audio_source) const;
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protected:
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AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,
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bool use_limiter);
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private:
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// Set mixing frequency through OutputFrequencyCalculator.
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void CalculateOutputFrequency();
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// Get mixing frequency.
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int OutputFrequency() const;
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// Compute what audio sources to mix from audio_source_list_. Ramp
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// in and out. Update mixed status. Mixes up to
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// kMaximumAmountOfMixedAudioSources audio sources.
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AudioFrameList GetAudioFromSources() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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// Add/remove the MixerAudioSource to the specified
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// MixerAudioSource list.
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bool AddAudioSourceToList(Source* audio_source,
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SourceStatusList* audio_source_list) const;
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bool RemoveAudioSourceFromList(Source* remove_audio_source,
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SourceStatusList* audio_source_list) const;
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// The critical section lock guards audio source insertion and
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// removal, which can be done from any thread. The race checker
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// checks that mixing is done sequentially.
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rtc::CriticalSection crit_;
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rtc::RaceChecker race_checker_;
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std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
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// The current sample frequency and sample size when mixing.
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int output_frequency_ GUARDED_BY(race_checker_);
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size_t sample_size_ GUARDED_BY(race_checker_);
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// List of all audio sources. Note all lists are disjunct
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SourceStatusList audio_source_list_ GUARDED_BY(crit_); // May be mixed.
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// Component that handles actual adding of audio frames.
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FrameCombiner frame_combiner_ GUARDED_BY(race_checker_);
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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