04c0722cf1
This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
79 lines
3.0 KiB
C++
79 lines
3.0 KiB
C++
/*
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* Copyright (c) 2016 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_AUDIO_AUDIO_TRANSPORT_PROXY_H_
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#define WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
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#include "webrtc/api/audio/audio_mixer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/modules/audio_device/include/audio_device_defines.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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class AudioTransportProxy : public AudioTransport {
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public:
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AudioTransportProxy(AudioTransport* voe_audio_transport,
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AudioProcessing* apm,
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AudioMixer* mixer);
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~AudioTransportProxy() override;
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int32_t RecordedDataIsAvailable(const void* audioSamples,
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const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void PushCaptureData(int voe_channel,
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const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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void PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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private:
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AudioTransport* voe_audio_transport_;
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AudioProcessing* apm_;
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rtc::scoped_refptr<AudioMixer> mixer_;
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AudioFrame mixed_frame_;
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// Converts mixed audio to the audio device output rate.
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PushResampler<int16_t> resampler_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy);
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
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