ac09501381
Reason for revert: Looks like things are still breaking upstream... :( Original issue's description: > Reland of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2141413002/ ) > > Reason for revert: > Will make one more try since we have now confirmed that our TaskQueue tests works on Android. Let's hope for the best... > > Original issue's description: > > Revert of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2138403003/ ) > > > > Reason for revert: > > Reverting again since it might have caused this issue: > > > > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/13622/steps/content_browsertests/logs/stdio > > > > Original issue's description: > > > Reland of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2139233002/ ) > > > > > > Reason for revert: > > > My original patch broke things that are now fixed by https://codereview.webrtc.org/2141193002/. > > > > > > Hence I am relanding my original change. > > > > > > Original issue's description: > > > > Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ ) > > > > > > > > Reason for revert: > > > > Seems to break things upstream. > > > > > > > > Original issue's description: > > > > > Adds data logging in native AudioDeviceBuffer class. > > > > > > > > > > Goal is to provide periodic logging of most essential audio parameters > > > > > for playout and recording sides. It will allow us to track if the native audio layer is working as intended. > > > > > > > > > > BUG=NONE > > > > > > > > > > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae > > > > > Cr-Commit-Position: refs/heads/master@{#13440} > > > > > > > > TBR=stefan@webrtc.org,henrika@webrtc.org > > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > > NOPRESUBMIT=true > > > > NOTREECHECKS=true > > > > NOTRY=true > > > > BUG=NONE > > > > > > > > Committed: https://crrev.com/025aa94ccb85e4c6fe20a3fecdac5d27ec9ba3da > > > > Cr-Commit-Position: refs/heads/master@{#13441} > > > > > > TBR=stefan@webrtc.org,sprang@webrtc.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=NONE > > > > > > Committed: https://crrev.com/dd2fdecc78c50377d10ec98b41179acde9218ee7 > > > Cr-Commit-Position: refs/heads/master@{#13455} > > > > TBR=stefan@webrtc.org,sprang@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > BUG=NONE > > > > Committed: https://crrev.com/5dd941e5a5ccde541d9b40a1df379ed59c5fab5c > > Cr-Commit-Position: refs/heads/master@{#13457} > > TBR=stefan@webrtc.org,sprang@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=NONE > > Committed: https://crrev.com/b201da3fab5efc048a4341f39293d2dcf27b2eec > Cr-Commit-Position: refs/heads/master@{#13462} TBR=stefan@webrtc.org,henrika@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=NONE Review-Url: https://codereview.webrtc.org/2148623004 Cr-Commit-Position: refs/heads/master@{#13464}
115 lines
3.2 KiB
C++
115 lines
3.2 KiB
C++
/*
|
|
* Copyright (c) 2012 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_DEVICE_AUDIO_DEVICE_BUFFER_H
|
|
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|
|
|
|
#include "webrtc/modules/audio_device/include/audio_device.h"
|
|
#include "webrtc/system_wrappers/include/file_wrapper.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
class CriticalSectionWrapper;
|
|
|
|
const uint32_t kPulsePeriodMs = 1000;
|
|
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
|
|
|
|
class AudioDeviceObserver;
|
|
|
|
class AudioDeviceBuffer {
|
|
public:
|
|
AudioDeviceBuffer();
|
|
virtual ~AudioDeviceBuffer();
|
|
|
|
void SetId(uint32_t id) {};
|
|
int32_t RegisterAudioCallback(AudioTransport* audioCallback);
|
|
|
|
int32_t InitPlayout();
|
|
int32_t InitRecording();
|
|
|
|
virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
|
|
virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
|
|
int32_t RecordingSampleRate() const;
|
|
int32_t PlayoutSampleRate() const;
|
|
|
|
virtual int32_t SetRecordingChannels(size_t channels);
|
|
virtual int32_t SetPlayoutChannels(size_t channels);
|
|
size_t RecordingChannels() const;
|
|
size_t PlayoutChannels() const;
|
|
int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
|
|
int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
|
|
|
|
virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples);
|
|
int32_t SetCurrentMicLevel(uint32_t level);
|
|
virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift);
|
|
virtual int32_t DeliverRecordedData();
|
|
uint32_t NewMicLevel() const;
|
|
|
|
virtual int32_t RequestPlayoutData(size_t nSamples);
|
|
virtual int32_t GetPlayoutData(void* audioBuffer);
|
|
|
|
int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
|
|
int32_t StopInputFileRecording();
|
|
int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
|
|
int32_t StopOutputFileRecording();
|
|
|
|
int32_t SetTypingStatus(bool typingStatus);
|
|
|
|
private:
|
|
CriticalSectionWrapper& _critSect;
|
|
CriticalSectionWrapper& _critSectCb;
|
|
|
|
AudioTransport* _ptrCbAudioTransport;
|
|
|
|
uint32_t _recSampleRate;
|
|
uint32_t _playSampleRate;
|
|
|
|
size_t _recChannels;
|
|
size_t _playChannels;
|
|
|
|
// selected recording channel (left/right/both)
|
|
AudioDeviceModule::ChannelType _recChannel;
|
|
|
|
// 2 or 4 depending on mono or stereo
|
|
size_t _recBytesPerSample;
|
|
size_t _playBytesPerSample;
|
|
|
|
// 10ms in stereo @ 96kHz
|
|
int8_t _recBuffer[kMaxBufferSizeBytes];
|
|
|
|
// one sample <=> 2 or 4 bytes
|
|
size_t _recSamples;
|
|
size_t _recSize; // in bytes
|
|
|
|
// 10ms in stereo @ 96kHz
|
|
int8_t _playBuffer[kMaxBufferSizeBytes];
|
|
|
|
// one sample <=> 2 or 4 bytes
|
|
size_t _playSamples;
|
|
size_t _playSize; // in bytes
|
|
|
|
FileWrapper& _recFile;
|
|
FileWrapper& _playFile;
|
|
|
|
uint32_t _currentMicLevel;
|
|
uint32_t _newMicLevel;
|
|
|
|
bool _typingStatus;
|
|
|
|
int _playDelayMS;
|
|
int _recDelayMS;
|
|
int _clockDrift;
|
|
int high_delay_counter_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
|