14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
130 lines
4.3 KiB
C++
130 lines
4.3 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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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#include "typedefs.h"
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#include "common_audio/resampler/include/resampler.h"
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#include "file_wrapper.h"
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#include "audio_device.h"
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#include "list_wrapper.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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const WebRtc_UWord32 kPulsePeriodMs = 1000;
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const WebRtc_UWord32 kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
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class AudioDeviceObserver;
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class MediaFile;
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class AudioDeviceBuffer
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{
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public:
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void SetId(WebRtc_UWord32 id);
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WebRtc_Word32 RegisterAudioCallback(AudioTransport* audioCallback);
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WebRtc_Word32 InitPlayout();
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WebRtc_Word32 InitRecording();
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WebRtc_Word32 SetRecordingSampleRate(WebRtc_UWord32 fsHz);
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WebRtc_Word32 SetPlayoutSampleRate(WebRtc_UWord32 fsHz);
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WebRtc_Word32 RecordingSampleRate() const;
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WebRtc_Word32 PlayoutSampleRate() const;
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WebRtc_Word32 SetRecordingChannels(WebRtc_UWord8 channels);
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WebRtc_Word32 SetPlayoutChannels(WebRtc_UWord8 channels);
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WebRtc_UWord8 RecordingChannels() const;
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WebRtc_UWord8 PlayoutChannels() const;
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WebRtc_Word32 SetRecordingChannel(
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const AudioDeviceModule::ChannelType channel);
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WebRtc_Word32 RecordingChannel(
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AudioDeviceModule::ChannelType& channel) const;
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WebRtc_Word32 SetRecordedBuffer(const void* audioBuffer,
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WebRtc_UWord32 nSamples);
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WebRtc_Word32 SetCurrentMicLevel(WebRtc_UWord32 level);
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WebRtc_Word32 SetVQEData(WebRtc_UWord32 playDelayMS,
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WebRtc_UWord32 recDelayMS,
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WebRtc_Word32 clockDrift);
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WebRtc_Word32 DeliverRecordedData();
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WebRtc_UWord32 NewMicLevel() const;
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WebRtc_Word32 RequestPlayoutData(WebRtc_UWord32 nSamples);
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WebRtc_Word32 GetPlayoutData(void* audioBuffer);
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WebRtc_Word32 StartInputFileRecording(
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const char fileName[kAdmMaxFileNameSize]);
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WebRtc_Word32 StopInputFileRecording();
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WebRtc_Word32 StartOutputFileRecording(
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const char fileName[kAdmMaxFileNameSize]);
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WebRtc_Word32 StopOutputFileRecording();
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AudioDeviceBuffer();
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~AudioDeviceBuffer();
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private:
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void _EmptyList();
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private:
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WebRtc_Word32 _id;
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CriticalSectionWrapper& _critSect;
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CriticalSectionWrapper& _critSectCb;
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AudioTransport* _ptrCbAudioTransport;
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WebRtc_UWord32 _recSampleRate;
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WebRtc_UWord32 _playSampleRate;
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WebRtc_UWord8 _recChannels;
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WebRtc_UWord8 _playChannels;
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// selected recording channel (left/right/both)
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AudioDeviceModule::ChannelType _recChannel;
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// 2 or 4 depending on mono or stereo
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WebRtc_UWord8 _recBytesPerSample;
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WebRtc_UWord8 _playBytesPerSample;
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// 10ms in stereo @ 96kHz
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int8_t _recBuffer[kMaxBufferSizeBytes];
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// one sample <=> 2 or 4 bytes
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WebRtc_UWord32 _recSamples;
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WebRtc_UWord32 _recSize; // in bytes
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// 10ms in stereo @ 96kHz
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int8_t _playBuffer[kMaxBufferSizeBytes];
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// one sample <=> 2 or 4 bytes
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WebRtc_UWord32 _playSamples;
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WebRtc_UWord32 _playSize; // in bytes
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FileWrapper& _recFile;
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FileWrapper& _playFile;
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WebRtc_UWord32 _currentMicLevel;
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WebRtc_UWord32 _newMicLevel;
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WebRtc_UWord32 _playDelayMS;
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WebRtc_UWord32 _recDelayMS;
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WebRtc_Word32 _clockDrift;
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bool _measureDelay;
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ListWrapper _pulseList;
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WebRtc_UWord32 _lastPulseTime;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
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