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269a3d415e
Removes the top 3 filtering based on frame energy. This behaviour is unexpected for many application developers and the platform should not have such arbitrary limitations. Developers can still implement top-N filtering using WebAudio or an SFU (recommended to increase scalability). Performance is not really a concern in this case since decoders on all receive streams are called regardless if they are mixed or not (assuming packets are received). This also fixes glitches caused by the current implementation since sources are not ramped out. Bug: chromium:1446655,webrtc:13818 Change-Id: I179a6d68d2517b94ff2d99ec269031a54e5099e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310180 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40349}
91 lines
2.8 KiB
C++
91 lines
2.8 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 MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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#define MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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#include <stddef.h>
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio/audio_frame.h"
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#include "api/audio/audio_mixer.h"
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#include "api/scoped_refptr.h"
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#include "modules/audio_mixer/frame_combiner.h"
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#include "modules/audio_mixer/output_rate_calculator.h"
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#include "rtc_base/race_checker.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class AudioMixerImpl : public AudioMixer {
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public:
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struct 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 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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AudioMixerImpl(const AudioMixerImpl&) = delete;
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AudioMixerImpl& operator=(const AudioMixerImpl&) = delete;
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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
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RTC_LOCKS_EXCLUDED(mutex_);
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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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struct HelperContainers;
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void UpdateSourceCountStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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// Fetches audio frames to mix from sources.
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rtc::ArrayView<AudioFrame* const> GetAudioFromSources(int output_frequency)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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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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mutable Mutex mutex_;
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std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
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// List of all audio sources.
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std::vector<std::unique_ptr<SourceStatus>> audio_source_list_
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RTC_GUARDED_BY(mutex_);
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const std::unique_ptr<HelperContainers> helper_containers_
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RTC_GUARDED_BY(mutex_);
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// Component that handles actual adding of audio frames.
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FrameCombiner frame_combiner_;
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// The highest source count this mixer has ever had. Used for UMA stats.
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size_t max_source_count_ever_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
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