2553450959
The addition of logging.h in r4729 was causing the win trybot to fail with "#pragma deprecated" errors in standard library headers. This turned out to be due to including strsafe.h (via audio_device_config.h) before sstream (via logging.h). strsafe.h was only being included for the unused DEBUG_PRINT macro. I removed all references to it. This incidentally removes a bunch of other unneeded headers discovered while trying to track the problem down. This didn't show up in the commitbots; my guess is that the trybots are using the VC10 toolchain and the commitbots the VC11 toolchain. TBR=pbos Review URL: https://webrtc-codereview.appspot.com/2204004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4738 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
3.8 KiB
C++
124 lines
3.8 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 "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/system_wrappers/interface/file_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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const uint32_t kPulsePeriodMs = 1000;
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const uint32_t 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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AudioDeviceBuffer();
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virtual ~AudioDeviceBuffer();
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void SetId(uint32_t id);
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int32_t RegisterAudioCallback(AudioTransport* audioCallback);
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int32_t InitPlayout();
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int32_t InitRecording();
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int32_t SetRecordingSampleRate(uint32_t fsHz);
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int32_t SetPlayoutSampleRate(uint32_t fsHz);
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int32_t RecordingSampleRate() const;
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int32_t PlayoutSampleRate() const;
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int32_t SetRecordingChannels(uint8_t channels);
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int32_t SetPlayoutChannels(uint8_t channels);
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uint8_t RecordingChannels() const;
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uint8_t PlayoutChannels() const;
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int32_t SetRecordingChannel(
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const AudioDeviceModule::ChannelType channel);
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int32_t RecordingChannel(
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AudioDeviceModule::ChannelType& channel) const;
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int32_t SetRecordedBuffer(const void* audioBuffer, uint32_t nSamples);
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int32_t SetCurrentMicLevel(uint32_t level);
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void SetVQEData(int playDelayMS,
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int recDelayMS,
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int clockDrift);
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int32_t DeliverRecordedData();
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uint32_t NewMicLevel() const;
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virtual int32_t RequestPlayoutData(uint32_t nSamples);
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virtual int32_t GetPlayoutData(void* audioBuffer);
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int32_t StartInputFileRecording(
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const char fileName[kAdmMaxFileNameSize]);
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int32_t StopInputFileRecording();
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int32_t StartOutputFileRecording(
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const char fileName[kAdmMaxFileNameSize]);
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int32_t StopOutputFileRecording();
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int32_t SetTypingStatus(bool typingStatus);
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private:
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int32_t _id;
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CriticalSectionWrapper& _critSect;
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CriticalSectionWrapper& _critSectCb;
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AudioTransport* _ptrCbAudioTransport;
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uint32_t _recSampleRate;
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uint32_t _playSampleRate;
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uint8_t _recChannels;
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uint8_t _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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uint8_t _recBytesPerSample;
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uint8_t _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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uint32_t _recSamples;
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uint32_t _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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uint32_t _playSamples;
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uint32_t _playSize; // in bytes
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FileWrapper& _recFile;
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FileWrapper& _playFile;
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uint32_t _currentMicLevel;
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uint32_t _newMicLevel;
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bool _typingStatus;
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int _playDelayMS;
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int _recDelayMS;
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int _clockDrift;
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int high_delay_counter_;
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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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