6955870806
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
124 lines
3.9 KiB
C++
124 lines
3.9 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/include/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 size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
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class AudioDeviceObserver;
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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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virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
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virtual 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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virtual int32_t SetRecordingChannels(size_t channels);
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virtual int32_t SetPlayoutChannels(size_t channels);
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size_t RecordingChannels() const;
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size_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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virtual int32_t SetRecordedBuffer(const void* audioBuffer,
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size_t nSamples);
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int32_t SetCurrentMicLevel(uint32_t level);
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virtual void SetVQEData(int playDelayMS,
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int recDelayMS,
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int clockDrift);
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virtual int32_t DeliverRecordedData();
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uint32_t NewMicLevel() const;
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virtual int32_t RequestPlayoutData(size_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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size_t _recChannels;
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size_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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size_t _recBytesPerSample;
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size_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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size_t _recSamples;
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size_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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size_t _playSamples;
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size_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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