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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
78 lines
3 KiB
C++
78 lines
3 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 AUDIO_AUDIO_TRANSPORT_PROXY_H_
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#define AUDIO_AUDIO_TRANSPORT_PROXY_H_
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#include "api/audio/audio_mixer.h"
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#include "common_audio/resampler/include/push_resampler.h"
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/scoped_ref_ptr.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* audio_processing,
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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* audio_processing_;
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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 // AUDIO_AUDIO_TRANSPORT_PROXY_H_
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