Chromium Code Reviews| Index: webrtc/audio/audio_transport_proxy.h |
| diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..730f50186e2324943d5090c43d7ca0f1c704baea |
| --- /dev/null |
| +++ b/webrtc/audio/audio_transport_proxy.h |
| @@ -0,0 +1,85 @@ |
| +/* |
| + * Copyright (c) 2016 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 WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |
| +#define WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |
| + |
| +#include "webrtc/api/audio/audio_mixer.h" |
| +#include "webrtc/base/constructormagic.h" |
| +#include "webrtc/modules/audio_device/include/audio_device_defines.h" |
| +#include "webrtc/modules/audio_processing/include/audio_processing.h" |
| + |
| +namespace webrtc { |
| + |
| +class AudioTransportProxy : public AudioTransport { |
| + public: |
| + AudioTransportProxy(AudioTransport* voe_audio_transport, |
| + AudioProcessing* apm, |
| + AudioMixer* mixer) |
| + : voe_audio_transport_(voe_audio_transport) { |
|
the sun
2016/11/14 13:50:09
Move to .cc
aleloi
2016/11/14 14:24:43
For clarity to have all defs in the same place? (D
|
| + RTC_DCHECK(voe_audio_transport); |
| + RTC_DCHECK(apm); |
| + } |
| + |
| + ~AudioTransportProxy() override {} |
| + |
| + int32_t RecordedDataIsAvailable(const void* audioSamples, |
| + const size_t nSamples, |
| + const size_t nBytesPerSample, |
| + const size_t nChannels, |
| + const uint32_t samplesPerSec, |
| + const uint32_t totalDelayMS, |
| + const int32_t clockDrift, |
| + const uint32_t currentMicLevel, |
| + const bool keyPressed, |
| + uint32_t& newMicLevel) override { |
| + // Pass call through to original audio transport instance. |
| + return voe_audio_transport_->RecordedDataIsAvailable( |
|
the sun
2016/11/14 13:50:09
Move to .cc
aleloi
2016/11/14 14:24:42
Done.
|
| + audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec, |
| + totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel); |
| + } |
| + |
| + int32_t NeedMorePlayData(const size_t nSamples, |
| + const size_t nBytesPerSample, |
| + const size_t nChannels, |
| + const uint32_t samplesPerSec, |
| + void* audioSamples, |
| + size_t& nSamplesOut, |
| + int64_t* elapsed_time_ms, |
| + int64_t* ntp_time_ms) override; |
| + |
| + void PushCaptureData(int voe_channel, |
| + const void* audio_data, |
| + int bits_per_sample, |
| + int sample_rate, |
| + size_t number_of_channels, |
| + size_t number_of_frames) override { |
| + // This is part of deprecated VoE interface operating on specific |
| + // VoE channels. It should not be used. |
| + RTC_NOTREACHED(); |
|
the sun
2016/11/14 13:50:08
Here as well
aleloi
2016/11/14 14:24:43
Done.
|
| + } |
| + |
| + void PullRenderData(int bits_per_sample, |
| + int sample_rate, |
| + size_t number_of_channels, |
| + size_t number_of_frames, |
| + void* audio_data, |
| + int64_t* elapsed_time_ms, |
| + int64_t* ntp_time_ms) override; |
| + |
| + private: |
| + AudioTransport* voe_audio_transport_; |
| + AudioFrame frame_for_mixing_; |
| + |
| + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy); |
| +}; |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_ |