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Porting gtalk alsa impl from depot to webrtc

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Review URL: http://webrtc-codereview.appspot.com/123002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@484 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@google.com 2011-08-30 07:41:55 +00:00
parent 46171cf546
commit 932096c84f
3 changed files with 595 additions and 1864 deletions
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3
src/modules/audio_device/main/source/linux/alsasymboltable_linux.h
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@ -51,8 +51,11 @@ namespace webrtc_adm_linux_alsa {
X(snd_pcm_reset) \
X(snd_pcm_state) \
X(snd_pcm_set_params) \
X(snd_pcm_get_params) \
X(snd_pcm_start) \
X(snd_pcm_stream) \
X(snd_pcm_frames_to_bytes) \
X(snd_pcm_bytes_to_frames) \
X(snd_pcm_wait) \
X(snd_pcm_writei) \
X(snd_pcm_info_get_class) \

2387
src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc
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File diff suppressed because it is too large Load Diff

69
src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h
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@ -25,23 +25,6 @@ namespace webrtc
class EventWrapper;
class ThreadWrapper;
// Number of continuous buffer check errors before going 0->1
const WebRtc_UWord16 THR_OLD_BUFFER_CHECK_METHOD = 30;
// Number of buffer check errors before going 1->2
const WebRtc_UWord16 THR_IGNORE_BUFFER_CHECK = 30;
// 2.7 seconds (decimal 131071)
const WebRtc_UWord32 ALSA_SNDCARD_BUFF_SIZE_REC = 0x1ffff;
// ~170 ms (decimal 8191) - enough since we only write to buffer if it contains
// less than 50 ms
const WebRtc_UWord32 ALSA_SNDCARD_BUFF_SIZE_PLAY = 0x1fff;
const WebRtc_UWord32 REC_TIMER_PERIOD_MS = 2;
const WebRtc_UWord32 PLAY_TIMER_PERIOD_MS = 5;
const WebRtc_UWord16 PLAYBACK_THRESHOLD = 50;
const WebRtc_UWord32 REC_SAMPLES_PER_MS = 48;
const WebRtc_UWord32 PLAY_SAMPLES_PER_MS = 48;
class AudioDeviceLinuxALSA : public AudioDeviceGeneric
{
public:
@ -184,7 +167,6 @@ private:
char* enumDeviceName = NULL,
const WebRtc_Word32 ednLen = 0) const;
WebRtc_Word32 ErrorRecovery(WebRtc_Word32 error, snd_pcm_t* deviceHandle);
void FillPlayoutBuffer();
private:
void Lock() { _critSect.Enter(); };
@ -193,10 +175,6 @@ private:
inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
WebRtc_Word32 PrepareStartRecording();
WebRtc_Word32 GetPlayoutBufferDelay();
WebRtc_Word32 GetRecordingBufferDelay(bool preRead);
private:
static bool RecThreadFunc(void*);
static bool PlayThreadFunc(void*);
@ -207,10 +185,6 @@ private:
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
EventWrapper& _timeEventRec;
EventWrapper& _timeEventPlay;
EventWrapper& _recStartEvent;
EventWrapper& _playStartEvent;
ThreadWrapper* _ptrThreadRec;
ThreadWrapper* _ptrThreadPlay;
@ -229,17 +203,28 @@ private:
snd_pcm_t* _handleRecord;
snd_pcm_t* _handlePlayout;
snd_pcm_uframes_t _recSndcardBuffsize;
snd_pcm_uframes_t _playSndcardBuffsize;
snd_pcm_uframes_t _recordingBuffersizeInFrame;
snd_pcm_uframes_t _recordingPeriodSizeInFrame;
snd_pcm_uframes_t _playoutBufferSizeInFrame;
snd_pcm_uframes_t _playoutPeriodSizeInFrame;
WebRtc_UWord32 _samplingFreqRec;
WebRtc_UWord32 _samplingFreqPlay;
ssize_t _recordingBufferSizeIn10MS;
ssize_t _playoutBufferSizeIn10MS;
WebRtc_UWord32 _recordingFramesIn10MS;
WebRtc_UWord32 _playoutFramesIn10MS;
WebRtc_UWord32 _recordingFreq;
WebRtc_UWord32 _playoutFreq;
WebRtc_UWord8 _recChannels;
WebRtc_UWord8 _playChannels;
WebRtc_Word8* _recordingBuffer; // in byte
WebRtc_Word8* _playoutBuffer; // in byte
WebRtc_UWord32 _recordingFramesLeft;
WebRtc_UWord32 _playoutFramesLeft;
WebRtc_UWord32 _playbackBufferSize;
WebRtc_UWord32 _recordBufferSize;
WebRtc_Word16* _recBuffer;
AudioDeviceModule::BufferType _playBufType;
private:
@ -248,28 +233,12 @@ private:
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
bool _startRec;
bool _stopRec;
bool _startPlay;
bool _stopPlay;
bool _AGC;
bool _buffersizeFromZeroAvail;
bool _buffersizeFromZeroDelay;
WebRtc_UWord32 _sndCardPlayDelay; // Just to store last value
WebRtc_UWord32 _previousSndCardPlayDelay; // Stores previous _sndCardPlayDelay value
WebRtc_UWord8 _delayMonitorStatePlay; // 0 normal, 1 monitor delay change (after error)
WebRtc_Word16 _largeDelayCountPlay; // Used when monitoring delay change
WebRtc_UWord32 _sndCardRecDelay;
WebRtc_UWord32 _numReadyRecSamples;
snd_pcm_sframes_t _recordingDelay;
snd_pcm_sframes_t _playoutDelay;
WebRtc_UWord8 _bufferCheckMethodPlay;
WebRtc_UWord8 _bufferCheckMethodRec;
WebRtc_UWord32 _bufferCheckErrorsPlay;
WebRtc_UWord32 _bufferCheckErrorsRec;
WebRtc_Word32 _lastBufferCheckValuePlay;
WebRtc_Word32 _writeErrors;
WebRtc_UWord16 _playWarning;
WebRtc_UWord16 _playError;
WebRtc_UWord16 _recWarning;