9561183708
This change changes mixing to be done at the lowest possible APM-native rate that does not lead to quality loss. An Audio Processing-native rate is one of 8, 16, 32, or 48 kHz. Mixing at a lower sampling rate and avoiding resampling can in many cases lead to big efficiency improvements, as reported by experiments. This CL also fixes a design issue with the AudioMixer: audio at non-native rates is no longer fed to the APM instance which is the limiter. NOTRY=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2458703002 Cr-Commit-Position: refs/heads/master@{#14980}
80 lines
3.1 KiB
C++
80 lines
3.1 KiB
C++
/*
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* Copyright (c) 2011 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_API_AUDIO_AUDIO_MIXER_H_
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#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
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#include <memory>
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#include "webrtc/base/refcount.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle.
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class AudioMixer : public rtc::RefCountInterface {
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public:
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// A callback class that all mixer participants must inherit from/implement.
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class Source {
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public:
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enum class AudioFrameInfo {
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kNormal, // The samples in audio_frame are valid and should be used.
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kMuted, // The samples in audio_frame should not be used, but
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// should be implicitly interpreted as zero. Other
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// fields in audio_frame may be read and should
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// contain meaningful values.
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kError, // The audio_frame will not be used.
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};
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// Overwrites |audio_frame|. The data_ field is overwritten with
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// 10 ms of new audio (either 1 or 2 interleaved channels) at
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// |sample_rate_hz|. All fields in |audio_frame| must be updated.
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virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
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AudioFrame* audio_frame) = 0;
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// A way for a mixer implementation to distinguish participants.
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virtual int Ssrc() const = 0;
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// A way for this source to say that GetAudioFrameWithInfo called
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// with this sample rate or higher will not cause quality loss.
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virtual int PreferredSampleRate() const = 0;
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virtual ~Source() {}
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};
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// Returns true if adding/removing was successful. A source is never
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// added twice and removal is never attempted if a source has not
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// been successfully added to the mixer. Addition and removal can
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// happen on different threads.
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virtual bool AddSource(Source* audio_source) = 0;
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virtual bool RemoveSource(Source* audio_source) = 0;
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// Performs mixing by asking registered audio sources for audio. The
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// mixed result is placed in the provided AudioFrame. This method
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// will only be called from a single thread. The channels argument
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// specifies the number of channels of the mix result. The mixer
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// should mix at a rate that doesn't cause quality loss of the
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// sources' audio. The mixing rate is one of the rates listed in
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// AudioProcessing::NativeRate. All fields in
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// |audio_frame_for_mixing| must be updated.
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virtual void Mix(size_t number_of_channels,
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AudioFrame* audio_frame_for_mixing) = 0;
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protected:
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// Since the mixer is reference counted, the destructor may be
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// called from any thread.
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~AudioMixer() override {}
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};
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} // namespace webrtc
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#endif // WEBRTC_API_AUDIO_AUDIO_MIXER_H_
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