vdr-plugin-softhddevice-drm-gles 1.6.4-d0291bb
Public Member Functions | Protected Member Functions | Private Member Functions | Private Attributes | Static Private Attributes | List of all members
cSoftHdAudio Class Reference
Audio Module

Audio Interface. More...

#include <audio.h>

Public Member Functions

 cSoftHdAudio (cSoftHdDevice *)
 Create a new audio context.
 
void LazyInit (void)
 Initialize audio output module (alsa)
 
void Exit (void)
 Cleanup audio output module (alsa)
 
int Setup (AVCodecContext *, int, int, int)
 Alsa setup wrapper.
 
void SetPaused (bool)
 Set audio playback pause state.
 
bool IsPaused (void)
 
void Filter (AVFrame *, AVCodecContext *)
 Send audio frame to filter and enqueue it.
 
void EnqueueSpdif (uint16_t *, int, AVFrame *)
 Enqueue prepared spdif bursts in audio output queue.
 
bool IsBufferFull (void)
 
void FlushBuffers (void)
 Flush audio buffers.
 
int GetUsedBytes (void)
 Get used bytes in audio ringbuffer.
 
int64_t GetHardwareOutputPtsMs (void)
 Get the hardware output PTS in milliseconds.
 
int64_t GetHardwareOutputDelayMs (void)
 Get the hardware delay in milliseconds.
 
int64_t GetHardwareOutputPtsTimebaseUnits (void)
 Get the hardware output PTS in timebase units.
 
int GetPassthrough (void) const
 
bool HasInputPts (void)
 
int64_t GetInputPtsMs (void)
 
int64_t GetOutputPtsMs (void)
 Get the output PTS of the ringbuffer.
 
int GetAvResyncBorderMs (void)
 
void SetEq (int[18], int)
 Set equalizer bands.
 
void SetVolume (int)
 Set mixer volume (0-1000)
 
void SetDownmix (int downMix)
 
void SetSoftvol (bool softVolume)
 
void SetNormalize (bool, int)
 Set normalize volume parameters.
 
void SetCompression (bool, int)
 Set volume compression parameters.
 
void SetStereoDescent (int)
 Set stereo loudness descent.
 
void SetPassthroughMask (int)
 Set audio passthrough mask.
 
void SetAutoAES (bool appendAes)
 
void SetTimebase (AVRational *timebase)
 
void DropSamplesOlderThanPtsMs (int64_t)
 Drop samples older than the given PTS.
 
void ClockDriftCompensation (void)
 Calculate clock drift compensation.
 
void ResetHwDelayBaseline (void)
 Reset the hw delay baseline.
 
void SetHwDelayBaseline (void)
 Set the hw delay baseline.
 

Protected Member Functions

virtual void Action (void)
 Audio thread loop, started with Start().
 

Private Member Functions

void Enqueue (uint16_t *, int, AVFrame *)
 Send audio data to ringbuffer.
 
void EnqueueFrame (AVFrame *)
 Place samples in audio output queue.
 
bool SendAudio (int)
 Write regular audio data from the ringbuffer to the hardware.
 
bool SendPause (void)
 Write pause to passthrough device.
 
void BuildPauseBurst (void)
 Build a pause spdif burst with the size of the last recognized normal spdif audio.
 
void Stop (void)
 Stop the thread.
 
void Normalize (uint16_t *, int)
 Normalize audio samples.
 
void Compress (uint16_t *, int)
 Compress audio samples.
 
void SoftAmplify (int16_t *, int)
 Amplify the samples in software.
 
int InitFilter (AVCodecContext *)
 Init audio filters.
 
AVFrame * FilterGetFrame (void)
 Get frame from filter sink.
 
int CheckForFilterReady (AVCodecContext *)
 Check if the filter has changed and is ready, init the filter if needed.
 
int AlsaSetup (int, int, int)
 Setup alsa audio for requested format.
 
char * OpenAlsaDevice (const char *, int)
 Open an alsa device.
 
char * FindAlsaDevice (const char *, const char *, int)
 Find alsa device giving some search hints.
 
void AlsaInitPCMDevice (void)
 Search for an alsa pcm device and open it.
 
void AlsaInitMixer (void)
 Initialize alsa mixer.
 
void AlsaSetVolume (int)
 Set alsa mixer volume (0-1000)
 
void AlsaInit (void)
 Initialize the alsa audio output module.
 
void AlsaExit (void)
 Cleanup the alsa audio output module.
 
void FlushAlsaBuffers (void)
 Flush alsa buffers.
 
void DropAlsaBuffers (void)
 Drop alsa buffers.
 
void FlushAlsaBuffersInternal (bool)
 Flush alsa buffers internally.
 
bool CyclicCall (void)
 Cyclic audio playback call.
 
void ProcessEvents (void)
 Process queued events and forward them to event receiver.
 
void HandleError (int)
 Handle an alsa error.
 
int64_t GetOutputPtsMsInternal (void)
 
int64_t PtsToMs (int64_t pts)
 
int64_t MsToPts (int64_t ptsMs)
 
int MsToFrames (int milliseconds)
 
int FramesToMs (int frames)
 
double FramesToMsDouble (int frames)
 

Private Attributes

cSoftHdDevice * m_pDevice
 pointer to device
 
cSoftHdConfig * m_pConfig
 pointer to config
 
IEventReceiver * m_pEventReceiver
 pointer to event receiver
 
cBufferFillLevelLowPassFilter m_fillLevel
 low pass filter for the buffer fill level
 
cPidController m_pidController {3, 0.005, 0, 1000}
 PID controller for clock drift compensation with tuning values coming from educated guesses.
 
std::chrono::steady_clock::time_point m_lastPidInvocation
 last time the PID controller was invoked
 
int m_alsaBufferSizeFrames = 0
 alsa buffer size in frames
 
int m_packetCounter = 0
 packet counter for logging
 
bool m_initialized = false
 class initialized
 
const int m_bytesPerSample = 2
 number of bytes per sample
 
unsigned int m_hwSampleRate = 0
 hardware sample rate in Hz
 
unsigned int m_hwNumChannels = 0
 number of hardware channels
 
AVRational * m_pTimebase
 pointer to AVCodecContext pkts_timebase
 
std::mutex m_mutex
 mutex for thread safety
 
std::mutex m_pauseMutex
 mutex for a safe thread pausing
 
std::vector< Event > m_eventQueue
 event queue for incoming events
 
std::atomic< double > m_pitchPpm = 0
 pitch adjustment in ppm. Positive values are faster
 
int m_pitchAdjustFrameCounter = 0
 counter for pitch adjustment frames
 
int m_downmix
 set stereo downmix
 
int64_t m_inputPts = AV_NOPTS_VALUE
 pts clock (last pts in ringbuffer)
 
std::atomic< bool > m_paused = true
 audio is paused
 
bool m_softVolume
 flag to use soft volume
 
int m_passthrough
 passthrough mask
 
const char * m_pPCMDevice
 PCM device name.
 
const char * m_pPassthroughDevice
 passthrough device name
 
bool m_appendAES
 flag ato utomatic append AES
 
int m_spdifBurstSize = 0
 size of the current spdif burst
 
std::vector< uint16_t > m_pauseBurst
 holds the burst data itself
 
snd_pcm_sframes_t m_hwBaseline = 0
 saves the hw delay (pause bursts) once a real audio frame to correctly do the AV-Sync
 
bool m_firstRealAudioReceived = false
 false, as long as no real audio was sent - used to trigger the baseline set
 
bool m_normalize
 flag to use volume normalize
 
const int m_normalizeSamples = 4096
 number of normalize samples
 
int m_normalizeCounter
 normalize sample counter
 
uint32_t m_normalizeAverage [NORMALIZE_MAX_INDEX]
 average of n last normalize sample blocks
 
int m_normalizeIndex
 index into normalize average table
 
int m_normalizeReady
 index normalize counter
 
int m_normalizeFactor
 current normalize factor
 
const int m_normalizeMinFactor = 100
 min. normalize factor
 
int m_normalizeMaxFactor
 max. normalize factor
 
bool m_compression
 flag to use compress volume
 
int m_compressionFactor = 0
 current compression factor
 
int m_compressionMaxFactor
 max. compression factor
 
int m_amplifier
 software volume amplify factor
 
int m_stereoDescent
 volume descent for stereo
 
int m_volume = 0
 current volume (0 .. 1000)
 
int m_useEqualizer
 flag to use equalizer
 
float m_equalizerBand [18]
 equalizer band
 
const char * m_pMixerDevice = nullptr
 mixer device name (not used)
 
const char * m_pMixerChannel
 mixer channel name
 
int m_filterChanged = 0
 filter has changed
 
int m_filterReady = 0
 filter is ready
 
AVFilterGraph * m_pFilterGraph = nullptr
 
AVFilterContext * m_pBuffersrcCtx
 
AVFilterContext * m_pBuffersinkCtx
 
cSoftHdRingbuffer m_pRingbuffer {RINGBUFFER_SIZE}
 sample ring buffer
 
snd_pcm_t * m_pAlsaPCMHandle
 alsa pcm handle
 
snd_mixer_t * m_pAlsaMixer = nullptr
 alsa mixer handle
 
snd_mixer_elem_t * m_pAlsaMixerElem = nullptr
 alsa mixer element
 
int m_alsaRatio
 internal -> mixer ratio * 1000
 
bool m_alsaUseMmap
 use mmap
 

Static Private Attributes

static constexpr int AUDIO_MIN_BUFFER_FREE = 3072 * 8 * 8
 Minimum free space in audio buffer 8 packets for 8 channels.
 
static constexpr int NORMALIZE_MAX_INDEX = 128
 number of normalize average samples
 
static constexpr int AV_SYNC_BORDER_MS = 5000
 absolute max a/v difference in ms which should trigger a resync
 
static constexpr unsigned RINGBUFFER_SIZE = 3 * 5 * 7 * 8 * 2 * 1000
 default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
 

Detailed Description

Audio Interface.

Definition at line 50 of file audio.h.

Constructor & Destructor Documentation

◆ cSoftHdAudio()

cSoftHdAudio::cSoftHdAudio ( cSoftHdDevice *  device)

Create a new audio context.

Definition at line 52 of file audio.cpp.

References cSoftHdConfig::ConfigAudioCompression, cSoftHdConfig::ConfigAudioEq, cSoftHdConfig::ConfigAudioEqBand, cSoftHdConfig::ConfigAudioMaxCompression, cSoftHdConfig::ConfigAudioMaxNormalize, cSoftHdConfig::ConfigAudioNormalize, cSoftHdConfig::ConfigAudioStereoDescent, m_pConfig, SetCompression(), SetEq(), SetNormalize(), and SetStereoDescent().

Member Function Documentation

◆ Action()

void cSoftHdAudio::Action ( void  )
protectedvirtual

Audio thread loop, started with Start().

Tries to periodically send frames to the hardware and checks for events (underruns)

Definition at line 1319 of file audio.cpp.

References CyclicCall(), LOGDEBUG, and ProcessEvents().

◆ AlsaExit()

void cSoftHdAudio::AlsaExit ( void  )
private

Cleanup the alsa audio output module.

Definition at line 1944 of file audio.cpp.

References m_pAlsaMixer, m_pAlsaMixerElem, and m_pAlsaPCMHandle.

Referenced by Exit().

◆ AlsaInit()

void cSoftHdAudio::AlsaInit ( void  )
private

Initialize the alsa audio output module.

Definition at line 1928 of file audio.cpp.

References AlsaInitMixer(), AlsaInitPCMDevice(), and AlsaNoopCallback().

Referenced by LazyInit().

◆ AlsaInitMixer()

void cSoftHdAudio::AlsaInitMixer ( void  )
private

Initialize alsa mixer.

Definition at line 1727 of file audio.cpp.

References L_SOUND, LOGDEBUG2, LOGERROR, m_alsaRatio, m_pAlsaMixer, m_pAlsaMixerElem, m_pMixerChannel, and m_pMixerDevice.

Referenced by AlsaInit().

◆ AlsaInitPCMDevice()

void cSoftHdAudio::AlsaInitPCMDevice ( void  )
private

Search for an alsa pcm device and open it.

Definition at line 1648 of file audio.cpp.

References FindAlsaDevice(), L_SOUND, LOGDEBUG2, LOGERROR, LOGFATAL, LOGINFO, LOGWARNING, m_pAlsaPCMHandle, m_passthrough, m_pPassthroughDevice, m_pPCMDevice, and OpenAlsaDevice().

Referenced by AlsaInit().

◆ AlsaSetup()

int cSoftHdAudio::AlsaSetup ( int  channels,
int  sample_rate,
int  passthrough 
)
private

Setup alsa audio for requested format.

Parameters
channelsChannels requested
sample_rateSampleRate requested
passthroughuse pass-through (AC-3, ...) device
Return values
0everything ok
-1something gone wrong
Todo:
FIXME: remove pointer for freq + channels

Definition at line 1803 of file audio.cpp.

References DropAlsaBuffers(), GetAlsaChannelLayoutAsArray(), L_SOUND, LOGDEBUG2, LOGERROR, LOGINFO, LOGWARNING, m_alsaBufferSizeFrames, m_alsaUseMmap, m_downmix, m_hwNumChannels, m_hwSampleRate, m_pAlsaPCMHandle, MsToFrames(), and Stop().

Referenced by InitFilter(), and Setup().

◆ AlsaSetVolume()

void cSoftHdAudio::AlsaSetVolume ( int  volume)
private

Set alsa mixer volume (0-1000)

Parameters
volumevolume (0 .. 1000)

Definition at line 1781 of file audio.cpp.

References m_alsaRatio, m_pAlsaMixer, and m_pAlsaMixerElem.

Referenced by SetVolume().

◆ BuildPauseBurst()

void cSoftHdAudio::BuildPauseBurst ( void  )
private

Build a pause spdif burst with the size of the last recognized normal spdif audio.

Definition at line 754 of file audio.cpp.

References IEC61937_NULL, IEC61937_PREAMBLE1, IEC61937_PREAMBLE2, m_pauseBurst, and m_spdifBurstSize.

Referenced by EnqueueSpdif().

◆ CheckForFilterReady()

int cSoftHdAudio::CheckForFilterReady ( AVCodecContext *  ctx)
private

Check if the filter has changed and is ready, init the filter if needed.

Parameters
ctxAVCodec audio decoding context
Return values
1error, init failed
0filter initiated

Definition at line 905 of file audio.cpp.

References InitFilter(), L_SOUND, LOGDEBUG2, m_filterChanged, m_filterReady, and m_pFilterGraph.

Referenced by Filter().

◆ ClockDriftCompensation()

void cSoftHdAudio::ClockDriftCompensation ( void  )

Calculate clock drift compensation.

Uses a PID controller to adjust the playback pitch based on the audio buffer fill level. This keeps the buffer level constant and compensates for clock drift between the sender and the audio hardware.

Also updates the low-pass filter for the buffer fill level.

Definition at line 1966 of file audio.cpp.

References FramesToMsDouble(), cBufferFillLevelLowPassFilter::GetBufferFillLevelFramesAvg(), cPidController::GetDTerm(), cPidController::GetITerm(), cPidController::GetPTerm(), cPidController::GetTargetValue(), cBufferFillLevelLowPassFilter::IsSettled(), L_SOUND, LOGDEBUG2, LOGWARNING, m_alsaBufferSizeFrames, m_fillLevel, m_lastPidInvocation, m_packetCounter, m_pAlsaPCMHandle, m_passthrough, m_pidController, m_pitchPpm, cPidController::SetTargetValue(), cPidController::Update(), and cBufferFillLevelLowPassFilter::UpdateAvgBufferFillLevel().

Referenced by cSoftHdDevice::PlayAudio().

◆ Compress()

void cSoftHdAudio::Compress ( uint16_t *  samples,
int  count 
)
private

Compress audio samples.

Parameters
samplessample buffer
countnumber of bytes in sample buffer

Definition at line 162 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_bytesPerSample, m_compressionFactor, and m_compressionMaxFactor.

Referenced by EnqueueFrame().

◆ CyclicCall()

bool cSoftHdAudio::CyclicCall ( void  )
private

Cyclic audio playback call.

Handles audio output to ALSA, writing samples from the ring buffer to the hardware when space is available.

If passthrough is enabled, the thread continues sending data (pause bursts) even if audio playback is paused. This prevents, that the AV-Receiver looses the lock and may switch to PCM instead.

Returns
true if data was written or the next write should be scheduled immediately

Definition at line 1357 of file audio.cpp.

References HandleError(), L_SOUND, LOGDEBUG2, LOGERROR, m_mutex, m_pAlsaPCMHandle, m_passthrough, m_paused, m_pauseMutex, m_spdifBurstSize, SendAudio(), and SendPause().

Referenced by Action().

◆ DropAlsaBuffers()

void cSoftHdAudio::DropAlsaBuffers ( void  )
private

Drop alsa buffers.

Definition at line 1254 of file audio.cpp.

References FlushAlsaBuffersInternal().

Referenced by AlsaSetup().

◆ DropSamplesOlderThanPtsMs()

void cSoftHdAudio::DropSamplesOlderThanPtsMs ( int64_t  ptsMs)

Drop samples older than the given PTS.

Removes audio samples from the ringbuffer that have a presentation timestamp older than the specified ptsMs.

Parameters
ptsMspresentation timestamp in milliseconds - samples older than this will be dropped

Definition at line 698 of file audio.cpp.

References GetOutputPtsMsInternal(), HasInputPts(), L_AV_SYNC, LOGDEBUG2, m_fillLevel, m_mutex, m_pAlsaPCMHandle, m_pidController, m_pRingbuffer, MsToFrames(), cSoftHdRingbuffer::ReadAdvance(), cBufferFillLevelLowPassFilter::Reset(), cPidController::Reset(), Timestamp2String(), cSoftHdRingbuffer::UsedBytes(), and cBufferFillLevelLowPassFilter::WroteFrames().

Referenced by cVideoRender::DisplayFrame(), and cSoftHdDevice::OnEventReceived().

◆ Enqueue()

void cSoftHdAudio::Enqueue ( uint16_t *  buffer,
int  count,
AVFrame *  frame 
)
private

Send audio data to ringbuffer.

Parameters
bufferdata buffer
countnumber of bytes in data buffer
framedecoded frame (used to get frame parameters)

Definition at line 797 of file audio.cpp.

References AV_NOPTS_VALUE, AV_SYNC_BORDER_MS, L_AV_SYNC, LOGDEBUG2, LOGERROR, m_eventQueue, m_fillLevel, m_inputPts, m_mutex, m_pAlsaPCMHandle, m_pitchAdjustFrameCounter, m_pitchPpm, m_pRingbuffer, PtsToMs(), cBufferFillLevelLowPassFilter::ReceivedFrames(), Timestamp2String(), and cSoftHdRingbuffer::Write().

Referenced by EnqueueFrame(), and EnqueueSpdif().

◆ EnqueueFrame()

void cSoftHdAudio::EnqueueFrame ( AVFrame *  frame)
private

Place samples in audio output queue.

Parameters
frameaudio frame

Definition at line 730 of file audio.cpp.

References Compress(), Enqueue(), m_bytesPerSample, m_compression, m_normalize, and Normalize().

Referenced by Filter().

◆ EnqueueSpdif()

void cSoftHdAudio::EnqueueSpdif ( uint16_t *  buffer,
int  count,
AVFrame *  frame 
)

Enqueue prepared spdif bursts in audio output queue.

Wrapper for Enqueue(), but builds a new pause burst if necessary

Parameters
bufferdata buffer
countnumber of bytes in data buffer
framedecoded frame (used to get frame parameters)

Definition at line 775 of file audio.cpp.

References BuildPauseBurst(), Enqueue(), L_SOUND, LOGDEBUG2, m_pauseBurst, m_pauseMutex, and m_spdifBurstSize.

Referenced by cAudioDecoder::DecodePassthrough().

◆ Exit()

void cSoftHdAudio::Exit ( void  )

Cleanup audio output module (alsa)

This currently also stops the audio thread.

Todo:
Move stopping the thread to AlsaExit()

Definition at line 1210 of file audio.cpp.

References AlsaExit(), L_SOUND, LOGDEBUG2, m_initialized, m_pFilterGraph, and Stop().

Referenced by cSoftHdDevice::OnEnteringState().

◆ Filter()

void cSoftHdAudio::Filter ( AVFrame *  inframe,
AVCodecContext *  ctx 
)

Send audio frame to filter and enqueue it.

Parameters
inframeincoming audio frame to be filtered
ctxAVCodec audio decoding context
Return values
1error, send again
0running

Definition at line 938 of file audio.cpp.

References CheckForFilterReady(), EnqueueFrame(), FilterGetFrame(), L_SOUND, LOGDEBUG2, LOGERROR, m_filterChanged, and m_pBuffersrcCtx.

Referenced by cAudioDecoder::Decode().

◆ FilterGetFrame()

AVFrame * cSoftHdAudio::FilterGetFrame ( void  )
private

Get frame from filter sink.

Returns
pointer to AVFrame if success, NULL otherwise

Definition at line 872 of file audio.cpp.

References LOGERROR, and m_pBuffersinkCtx.

Referenced by Filter().

◆ FindAlsaDevice()

char * cSoftHdAudio::FindAlsaDevice ( const char *  devname,
const char *  hint,
int  passthrough 
)
private

Find alsa device giving some search hints.

Parameters
devnameinterface identification (e.g. "pcm")
hintstring to compare with device name hints
passthroughset, if we want a passthrough device
Returns
an opened alsa device name if successful, NULL otherwise NOTE: Returned string is allocated and must be freed by caller

Definition at line 1612 of file audio.cpp.

References LOGWARNING, and OpenAlsaDevice().

Referenced by AlsaInitPCMDevice().

◆ FlushAlsaBuffers()

void cSoftHdAudio::FlushAlsaBuffers ( void  )
private

Flush alsa buffers.

Definition at line 1246 of file audio.cpp.

References FlushAlsaBuffersInternal().

Referenced by FlushBuffers().

◆ FlushAlsaBuffersInternal()

void cSoftHdAudio::FlushAlsaBuffersInternal ( bool  drop)
private

Flush alsa buffers internally.

Parameters
dropforce a snd_pcm_drop of the audio frames already in the kernel

Definition at line 1264 of file audio.cpp.

References L_SOUND, LOGDEBUG2, LOGERROR, m_compressionFactor, m_compressionMaxFactor, m_normalizeAverage, m_normalizeCounter, m_normalizeFactor, m_normalizeReady, m_pAlsaPCMHandle, m_passthrough, and NORMALIZE_MAX_INDEX.

Referenced by DropAlsaBuffers(), and FlushAlsaBuffers().

◆ FlushBuffers()

void cSoftHdAudio::FlushBuffers ( void  )

Flush audio buffers.

Flush the alsa buffers and reset audio: pts, ringbuffer, pidController, fillLevel

Definition at line 981 of file audio.cpp.

References AV_NOPTS_VALUE, FlushAlsaBuffers(), L_SOUND, LOGDEBUG2, m_fillLevel, m_filterChanged, m_initialized, m_inputPts, m_mutex, m_pidController, m_pRingbuffer, cBufferFillLevelLowPassFilter::Reset(), cPidController::Reset(), cSoftHdRingbuffer::Reset(), and cBufferFillLevelLowPassFilter::ResetFramesCounters().

Referenced by cSoftHdDevice::FlushAudio().

◆ FramesToMs()

int cSoftHdAudio::FramesToMs ( int  frames)
inlineprivate

Definition at line 212 of file audio.h.

References m_hwSampleRate.

Referenced by GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), GetOutputPtsMsInternal(), and SetHwDelayBaseline().

◆ FramesToMsDouble()

double cSoftHdAudio::FramesToMsDouble ( int  frames)
inlineprivate

Definition at line 213 of file audio.h.

References m_hwSampleRate.

Referenced by ClockDriftCompensation().

◆ GetAvResyncBorderMs()

int cSoftHdAudio::GetAvResyncBorderMs ( void  )
inline

Definition at line 72 of file audio.h.

References AV_SYNC_BORDER_MS.

Referenced by cVideoRender::DisplayFrame().

◆ GetHardwareOutputDelayMs()

int64_t cSoftHdAudio::GetHardwareOutputDelayMs ( void  )

Get the hardware delay in milliseconds.

Returns
delay in milliseconds, or AV_NOPTS_VALUE if not available

Definition at line 1065 of file audio.cpp.

References AV_NOPTS_VALUE, FramesToMs(), m_hwSampleRate, m_inputPts, m_mutex, and m_pAlsaPCMHandle.

Referenced by cVideoRender::LogDroppedDuped().

◆ GetHardwareOutputPtsMs()

int64_t cSoftHdAudio::GetHardwareOutputPtsMs ( void  )

Get the hardware output PTS in milliseconds.

Calculates the presentation timestamp of audio currently being output by the hardware by accounting for ALSA/kernel buffer delays. This represents the PTS of the audio that is actually being played right now.

Returns
PTS in milliseconds, or AV_NOPTS_VALUE if not available

Definition at line 1043 of file audio.cpp.

References AV_NOPTS_VALUE, FramesToMs(), GetOutputPtsMsInternal(), m_hwBaseline, m_hwSampleRate, m_inputPts, m_mutex, and m_pAlsaPCMHandle.

Referenced by cSoftHdPlayer::Action(), cVideoRender::DisplayFrame(), GetHardwareOutputPtsTimebaseUnits(), and cSoftHdPlayer::Play().

◆ GetHardwareOutputPtsTimebaseUnits()

int64_t cSoftHdAudio::GetHardwareOutputPtsTimebaseUnits ( void  )

Get the hardware output PTS in timebase units.

Returns
presentation timestamp in timebase units

Definition at line 1084 of file audio.cpp.

References AV_NOPTS_VALUE, GetHardwareOutputPtsMs(), and MsToPts().

Referenced by cSoftHdDevice::GetSTC().

◆ GetInputPtsMs()

int64_t cSoftHdAudio::GetInputPtsMs ( void  )
inline

Definition at line 70 of file audio.h.

References m_inputPts, and PtsToMs().

Referenced by cSoftHdDevice::IsBufferingThresholdReached().

◆ GetOutputPtsMs()

int64_t cSoftHdAudio::GetOutputPtsMs ( void  )

Get the output PTS of the ringbuffer.

Calculates the presentation timestamp of the next audio sample that will be output from the ringbuffer. This is the input PTS minus the duration of audio currently buffered in the ringbuffer.

Note: This does not account for ALSA/kernel buffer delays. For the actual hardware output PTS, use GetHardwareOutputPtsMs() instead.

Returns
PTS in milliseconds

Definition at line 1022 of file audio.cpp.

References GetOutputPtsMsInternal(), and m_mutex.

Referenced by cSoftHdDevice::GetFirstAudioPtsMsToPlay(), cSoftHdDevice::GetFirstVideoPtsMsToPlay(), cSoftHdDevice::IsBufferingThresholdReached(), and cSoftHdDevice::OnEventReceived().

◆ GetOutputPtsMsInternal()

int64_t cSoftHdAudio::GetOutputPtsMsInternal ( void  )
private

Definition at line 1029 of file audio.cpp.

References FramesToMs(), m_inputPts, m_pAlsaPCMHandle, m_pRingbuffer, PtsToMs(), and cSoftHdRingbuffer::UsedBytes().

Referenced by DropSamplesOlderThanPtsMs(), GetHardwareOutputPtsMs(), and GetOutputPtsMs().

◆ GetPassthrough()

int cSoftHdAudio::GetPassthrough ( void  ) const
inline

Definition at line 68 of file audio.h.

References m_passthrough.

◆ GetUsedBytes()

int cSoftHdAudio::GetUsedBytes ( void  )

Get used bytes in audio ringbuffer.

Definition at line 1004 of file audio.cpp.

References m_pRingbuffer, and cSoftHdRingbuffer::UsedBytes().

Referenced by cVideoRender::LogDroppedDuped().

◆ HandleError()

void cSoftHdAudio::HandleError ( int  error)
private

Handle an alsa error.

Definition at line 1231 of file audio.cpp.

References AUDIO, LOGERROR, m_eventQueue, m_pAlsaPCMHandle, and m_passthrough.

Referenced by CyclicCall().

◆ HasInputPts()

bool cSoftHdAudio::HasInputPts ( void  )
inline

Definition at line 69 of file audio.h.

References AV_NOPTS_VALUE, and m_inputPts.

Referenced by DropSamplesOlderThanPtsMs(), cSoftHdDevice::IsBufferingThresholdReached(), and cSoftHdDevice::OnEventReceived().

◆ InitFilter()

int cSoftHdAudio::InitFilter ( AVCodecContext *  audioCtx)
private

Init audio filters.

The following alsa filters are set:

  • abuffer
  • channelmap
  • superequalizer
  • aformat
  • abuffersink
Return values
0everything ok
1didn't support channels, downmix set -> scrap this frame, test next
-1something gone wrong

Definition at line 473 of file audio.cpp.

References AlsaSetup(), BuildChannelMapFilter(), L_SOUND, LOGDEBUG2, LOGERROR, LOGWARNING, m_downmix, m_equalizerBand, m_filterChanged, m_filterReady, m_hwNumChannels, m_hwSampleRate, m_pAlsaPCMHandle, m_pBuffersinkCtx, m_pBuffersrcCtx, m_pFilterGraph, m_pTimebase, and m_useEqualizer.

Referenced by CheckForFilterReady().

◆ IsBufferFull()

bool cSoftHdAudio::IsBufferFull ( void  )
inline

Definition at line 61 of file audio.h.

References AUDIO_MIN_BUFFER_FREE, cSoftHdRingbuffer::FreeBytes(), and m_pRingbuffer.

Referenced by cSoftHdDevice::PlayAudio(), cSoftHdDevice::PlayAudioPkts(), and cSoftHdDevice::Poll().

◆ IsPaused()

bool cSoftHdAudio::IsPaused ( void  )
inline

Definition at line 58 of file audio.h.

References m_paused.

Referenced by cVideoRender::DisplayFrame().

◆ LazyInit()

void cSoftHdAudio::LazyInit ( void  )

Initialize audio output module (alsa)

The init is done lazily as soon as there is a STOP->PLAY state change or the mediaplayer wants to play video or audio.

This function can safely be called anytime, because it does nothing, if the init has already be done.

Definition at line 1195 of file audio.cpp.

References AlsaInit(), and m_initialized.

Referenced by cSoftHdDevice::OnEventReceived(), cSoftHdDevice::PlayAudioPkts(), and cSoftHdDevice::PlayVideoPkts().

◆ MsToFrames()

int cSoftHdAudio::MsToFrames ( int  milliseconds)
inlineprivate

Definition at line 211 of file audio.h.

References m_hwSampleRate.

Referenced by AlsaSetup(), and DropSamplesOlderThanPtsMs().

◆ MsToPts()

int64_t cSoftHdAudio::MsToPts ( int64_t  ptsMs)
inlineprivate

Definition at line 210 of file audio.h.

References m_pTimebase.

Referenced by GetHardwareOutputPtsTimebaseUnits().

◆ Normalize()

void cSoftHdAudio::Normalize ( uint16_t *  samples,
int  count 
)
private

Normalize audio samples.

Parameters
samplessample buffer
countnumber of bytes in sample buffer

Definition at line 81 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_bytesPerSample, m_normalizeAverage, m_normalizeCounter, m_normalizeFactor, m_normalizeIndex, m_normalizeMaxFactor, m_normalizeMinFactor, m_normalizeReady, m_normalizeSamples, and NORMALIZE_MAX_INDEX.

Referenced by EnqueueFrame().

◆ OpenAlsaDevice()

char * cSoftHdAudio::OpenAlsaDevice ( const char *  device,
int  passthrough 
)
private

Open an alsa device.

Parameters
devicealsa device to be opened
passthroughset, if this is a passthrough device
Returns
the alsa device if successful, NULL otherwise

Definition at line 1560 of file audio.cpp.

References L_SOUND, LOGDEBUG2, LOGWARNING, m_appendAES, and m_pAlsaPCMHandle.

Referenced by AlsaInitPCMDevice(), and FindAlsaDevice().

◆ ProcessEvents()

void cSoftHdAudio::ProcessEvents ( void  )
private

Process queued events and forward them to event receiver.

Definition at line 1544 of file audio.cpp.

References m_eventQueue, m_pEventReceiver, and IEventReceiver::OnEventReceived().

Referenced by Action().

◆ PtsToMs()

int64_t cSoftHdAudio::PtsToMs ( int64_t  pts)
inlineprivate

Definition at line 209 of file audio.h.

References m_pTimebase.

Referenced by Enqueue(), GetInputPtsMs(), and GetOutputPtsMsInternal().

◆ ResetHwDelayBaseline()

void cSoftHdAudio::ResetHwDelayBaseline ( void  )

Reset the hw delay baseline.

Definition at line 1532 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_firstRealAudioReceived, m_hwBaseline, and m_mutex.

Referenced by cSoftHdDevice::OnEnteringState(), cSoftHdDevice::OnEventReceived(), and cSoftHdDevice::OnLeavingState().

◆ SendAudio()

bool cSoftHdAudio::SendAudio ( int  freeAlsaBufferFrames)
private

Write regular audio data from the ringbuffer to the hardware.

Parameters
freeAlsaBufferFramesnumber of frames that can be written to the hardware
Return values
trueif data was written or the write should be scheduled again immediately
falseif no data was written

Definition at line 1418 of file audio.cpp.

References cSoftHdRingbuffer::GetReadPointer(), L_SOUND, LOGDEBUG2, LOGERROR, LOGWARNING, m_alsaUseMmap, m_fillLevel, m_pAlsaPCMHandle, m_passthrough, m_pRingbuffer, m_softVolume, m_volume, cSoftHdRingbuffer::ReadAdvance(), SoftAmplify(), and cBufferFillLevelLowPassFilter::WroteFrames().

Referenced by CyclicCall().

◆ SendPause()

bool cSoftHdAudio::SendPause ( void  )
private

Write pause to passthrough device.

Returns
true if a complete burst was written, false otherwise

Definition at line 1478 of file audio.cpp.

References LOGERROR, LOGWARNING, m_alsaUseMmap, m_pAlsaPCMHandle, m_pauseBurst, and m_spdifBurstSize.

Referenced by CyclicCall().

◆ SetAutoAES()

void cSoftHdAudio::SetAutoAES ( bool  appendAes)
inline

Definition at line 82 of file audio.h.

References m_appendAES.

Referenced by cMenuSetupSoft::Store().

◆ SetCompression()

void cSoftHdAudio::SetCompression ( bool  enable,
int  maxfac 
)

Set volume compression parameters.

Parameters
enabletrue, turn on compression
maxfacmax. factor of compression / 1000

Definition at line 1147 of file audio.cpp.

References m_compression, m_compressionFactor, and m_compressionMaxFactor.

Referenced by cSoftHdAudio(), and cMenuSetupSoft::Store().

◆ SetDownmix()

void cSoftHdAudio::SetDownmix ( int  downMix)
inline

Definition at line 76 of file audio.h.

References m_downmix.

Referenced by cMenuSetupSoft::Store().

◆ SetEq()

void cSoftHdAudio::SetEq ( int  band[18],
int  onoff 
)

Set equalizer bands.

Parameters
bandsetting frequenz bands
onoffset using equalizer

Definition at line 248 of file audio.cpp.

References m_equalizerBand, m_filterChanged, and m_useEqualizer.

Referenced by cSoftHdAudio(), and cMenuSetupSoft::Store().

◆ SetHwDelayBaseline()

void cSoftHdAudio::SetHwDelayBaseline ( void  )

Set the hw delay baseline.

Definition at line 1512 of file audio.cpp.

References FramesToMs(), L_SOUND, LOGDEBUG2, m_firstRealAudioReceived, m_hwBaseline, m_pAlsaPCMHandle, and m_passthrough.

Referenced by cSoftHdDevice::OnEventReceived(), and cSoftHdDevice::OnLeavingState().

◆ SetNormalize()

void cSoftHdAudio::SetNormalize ( bool  enable,
int  maxfac 
)

Set normalize volume parameters.

Parameters
enabletrue, turn on normalize
maxfacmax. factor of normalize / 1000

Definition at line 1135 of file audio.cpp.

References m_normalize, and m_normalizeMaxFactor.

Referenced by cSoftHdAudio(), and cMenuSetupSoft::Store().

◆ SetPassthroughMask()

void cSoftHdAudio::SetPassthroughMask ( int  mask)

Set audio passthrough mask.

Parameters
maskpassthrough mask (as a bitmask)

Definition at line 1176 of file audio.cpp.

References m_passthrough, and m_pitchPpm.

Referenced by cSoftHdDevice::SetPassthroughMask().

◆ SetPaused()

void cSoftHdAudio::SetPaused ( bool  pause)

Set audio playback pause state.

Parameters
pausetrue to pause, false to resume

Definition at line 1121 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_paused, and m_pauseMutex.

Referenced by cSoftHdDevice::Clear(), cVideoRender::DisplayFrame(), cSoftHdDevice::OnEnteringState(), cSoftHdDevice::OnEventReceived(), and cSoftHdDevice::OnLeavingState().

◆ SetSoftvol()

void cSoftHdAudio::SetSoftvol ( bool  softVolume)
inline

Definition at line 77 of file audio.h.

References m_softVolume.

Referenced by cMenuSetupSoft::Store().

◆ SetStereoDescent()

void cSoftHdAudio::SetStereoDescent ( int  delta)

Set stereo loudness descent.

Parameters
deltavalue (/1000) to reduce stereo volume

Definition at line 1165 of file audio.cpp.

References m_stereoDescent, m_volume, and SetVolume().

Referenced by cSoftHdAudio(), and cMenuSetupSoft::Store().

◆ SetTimebase()

void cSoftHdAudio::SetTimebase ( AVRational *  timebase)
inline

Definition at line 83 of file audio.h.

References m_pTimebase.

Referenced by cAudioDecoder::DecodePassthrough().

◆ Setup()

int cSoftHdAudio::Setup ( AVCodecContext *  ctx,
int  samplerate,
int  channels,
int  passthrough 
)

Alsa setup wrapper.

only used for passthrough atm, setting up PCM goes via Filter()

Parameters
AudioCtxAVCodec audio decoding context
sampleratestream samplerate
channelsstream nb of channels
passthroughpassthrough enabled
Return values
0everything ok
-1something gone wrong in AlsaSetup
1no parameter change, no setup needed

Definition at line 849 of file audio.cpp.

References AlsaSetup(), LOGERROR, m_downmix, m_hwNumChannels, m_hwSampleRate, and m_pTimebase.

Referenced by cAudioDecoder::UpdateFormat().

◆ SetVolume()

void cSoftHdAudio::SetVolume ( int  volume)

Set mixer volume (0-1000)

Parameters
volumevolume (0 .. 1000)

Definition at line 1098 of file audio.cpp.

References AlsaSetVolume(), m_amplifier, m_hwNumChannels, m_passthrough, m_softVolume, m_stereoDescent, and m_volume.

Referenced by cSoftHdDevice::OnEventReceived(), SetStereoDescent(), and cSoftHdDevice::SetVolumeDevice().

◆ SoftAmplify()

void cSoftHdAudio::SoftAmplify ( int16_t *  samples,
int  count 
)
private

Amplify the samples in software.

Parameters
samplessample buffer
countnumber of bytes in sample buffer
Todo:
FIXME: this does hard clipping

Definition at line 219 of file audio.cpp.

References m_amplifier, m_bytesPerSample, and m_volume.

Referenced by SendAudio().

◆ Stop()

void cSoftHdAudio::Stop ( void  )
private

Stop the thread.

Definition at line 1337 of file audio.cpp.

References LOGDEBUG.

Referenced by AlsaSetup(), and Exit().

Member Data Documentation

◆ AUDIO_MIN_BUFFER_FREE

constexpr int cSoftHdAudio::AUDIO_MIN_BUFFER_FREE = 3072 * 8 * 8
staticconstexprprivate

Minimum free space in audio buffer 8 packets for 8 channels.

Definition at line 94 of file audio.h.

Referenced by IsBufferFull().

◆ AV_SYNC_BORDER_MS

constexpr int cSoftHdAudio::AV_SYNC_BORDER_MS = 5000
staticconstexprprivate

absolute max a/v difference in ms which should trigger a resync

Definition at line 96 of file audio.h.

Referenced by Enqueue(), and GetAvResyncBorderMs().

◆ m_alsaBufferSizeFrames

int cSoftHdAudio::m_alsaBufferSizeFrames = 0
private

alsa buffer size in frames

Definition at line 103 of file audio.h.

Referenced by AlsaSetup(), and ClockDriftCompensation().

◆ m_alsaRatio

int cSoftHdAudio::m_alsaRatio
private

internal -> mixer ratio * 1000

Definition at line 190 of file audio.h.

Referenced by AlsaInitMixer(), and AlsaSetVolume().

◆ m_alsaUseMmap

bool cSoftHdAudio::m_alsaUseMmap
private

use mmap

Definition at line 191 of file audio.h.

Referenced by AlsaSetup(), SendAudio(), and SendPause().

◆ m_amplifier

int cSoftHdAudio::m_amplifier
private

software volume amplify factor

Definition at line 159 of file audio.h.

Referenced by SetVolume(), and SoftAmplify().

◆ m_appendAES

bool cSoftHdAudio::m_appendAES
private

flag ato utomatic append AES

Definition at line 127 of file audio.h.

Referenced by OpenAlsaDevice(), and SetAutoAES().

◆ m_bytesPerSample

const int cSoftHdAudio::m_bytesPerSample = 2
private

number of bytes per sample

Definition at line 108 of file audio.h.

Referenced by Compress(), EnqueueFrame(), Normalize(), and SoftAmplify().

◆ m_compression

bool cSoftHdAudio::m_compression
private

flag to use compress volume

Definition at line 153 of file audio.h.

Referenced by EnqueueFrame(), and SetCompression().

◆ m_compressionFactor

int cSoftHdAudio::m_compressionFactor = 0
private

current compression factor

Definition at line 154 of file audio.h.

Referenced by Compress(), FlushAlsaBuffersInternal(), and SetCompression().

◆ m_compressionMaxFactor

int cSoftHdAudio::m_compressionMaxFactor
private

max. compression factor

Definition at line 155 of file audio.h.

Referenced by Compress(), FlushAlsaBuffersInternal(), and SetCompression().

◆ m_downmix

int cSoftHdAudio::m_downmix
private

set stereo downmix

Definition at line 118 of file audio.h.

Referenced by AlsaSetup(), InitFilter(), SetDownmix(), and Setup().

◆ m_equalizerBand

float cSoftHdAudio::m_equalizerBand[18]
private

equalizer band

Definition at line 166 of file audio.h.

Referenced by InitFilter(), and SetEq().

◆ m_eventQueue

std::vector<Event> cSoftHdAudio::m_eventQueue
private

event queue for incoming events

Definition at line 114 of file audio.h.

Referenced by Enqueue(), HandleError(), and ProcessEvents().

◆ m_fillLevel

cBufferFillLevelLowPassFilter cSoftHdAudio::m_fillLevel
private

low pass filter for the buffer fill level

Definition at line 100 of file audio.h.

Referenced by ClockDriftCompensation(), DropSamplesOlderThanPtsMs(), Enqueue(), FlushBuffers(), and SendAudio().

◆ m_filterChanged

int cSoftHdAudio::m_filterChanged = 0
private

filter has changed

Definition at line 173 of file audio.h.

Referenced by CheckForFilterReady(), Filter(), FlushBuffers(), InitFilter(), and SetEq().

◆ m_filterReady

int cSoftHdAudio::m_filterReady = 0
private

filter is ready

Definition at line 174 of file audio.h.

Referenced by CheckForFilterReady(), and InitFilter().

◆ m_firstRealAudioReceived

bool cSoftHdAudio::m_firstRealAudioReceived = false
private

false, as long as no real audio was sent - used to trigger the baseline set

Definition at line 131 of file audio.h.

Referenced by ResetHwDelayBaseline(), and SetHwDelayBaseline().

◆ m_hwBaseline

snd_pcm_sframes_t cSoftHdAudio::m_hwBaseline = 0
private

saves the hw delay (pause bursts) once a real audio frame to correctly do the AV-Sync

Definition at line 130 of file audio.h.

Referenced by GetHardwareOutputPtsMs(), ResetHwDelayBaseline(), and SetHwDelayBaseline().

◆ m_hwNumChannels

unsigned int cSoftHdAudio::m_hwNumChannels = 0
private

number of hardware channels

Definition at line 110 of file audio.h.

Referenced by AlsaSetup(), InitFilter(), Setup(), and SetVolume().

◆ m_hwSampleRate

unsigned int cSoftHdAudio::m_hwSampleRate = 0
private

hardware sample rate in Hz

Definition at line 109 of file audio.h.

Referenced by AlsaSetup(), FramesToMs(), FramesToMsDouble(), GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), InitFilter(), MsToFrames(), and Setup().

◆ m_initialized

bool cSoftHdAudio::m_initialized = false
private

class initialized

Definition at line 107 of file audio.h.

Referenced by Exit(), FlushBuffers(), and LazyInit().

◆ m_inputPts

int64_t cSoftHdAudio::m_inputPts = AV_NOPTS_VALUE
private

pts clock (last pts in ringbuffer)

Definition at line 120 of file audio.h.

Referenced by Enqueue(), FlushBuffers(), GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), GetInputPtsMs(), GetOutputPtsMsInternal(), and HasInputPts().

◆ m_lastPidInvocation

std::chrono::steady_clock::time_point cSoftHdAudio::m_lastPidInvocation
private

last time the PID controller was invoked

Definition at line 102 of file audio.h.

Referenced by ClockDriftCompensation().

◆ m_mutex

std::mutex cSoftHdAudio::m_mutex
private

mutex for thread safety

Definition at line 112 of file audio.h.

Referenced by CyclicCall(), DropSamplesOlderThanPtsMs(), Enqueue(), FlushBuffers(), GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), GetOutputPtsMs(), and ResetHwDelayBaseline().

◆ m_normalize

bool cSoftHdAudio::m_normalize
private

flag to use volume normalize

Definition at line 141 of file audio.h.

Referenced by EnqueueFrame(), and SetNormalize().

◆ m_normalizeAverage

uint32_t cSoftHdAudio::m_normalizeAverage[NORMALIZE_MAX_INDEX]
private

average of n last normalize sample blocks

Definition at line 144 of file audio.h.

Referenced by FlushAlsaBuffersInternal(), and Normalize().

◆ m_normalizeCounter

int cSoftHdAudio::m_normalizeCounter
private

normalize sample counter

Definition at line 143 of file audio.h.

Referenced by FlushAlsaBuffersInternal(), and Normalize().

◆ m_normalizeFactor

int cSoftHdAudio::m_normalizeFactor
private

current normalize factor

Definition at line 147 of file audio.h.

Referenced by FlushAlsaBuffersInternal(), and Normalize().

◆ m_normalizeIndex

int cSoftHdAudio::m_normalizeIndex
private

index into normalize average table

Definition at line 145 of file audio.h.

Referenced by Normalize().

◆ m_normalizeMaxFactor

int cSoftHdAudio::m_normalizeMaxFactor
private

max. normalize factor

Definition at line 149 of file audio.h.

Referenced by Normalize(), and SetNormalize().

◆ m_normalizeMinFactor

const int cSoftHdAudio::m_normalizeMinFactor = 100
private

min. normalize factor

Definition at line 148 of file audio.h.

Referenced by Normalize().

◆ m_normalizeReady

int cSoftHdAudio::m_normalizeReady
private

index normalize counter

Definition at line 146 of file audio.h.

Referenced by FlushAlsaBuffersInternal(), and Normalize().

◆ m_normalizeSamples

const int cSoftHdAudio::m_normalizeSamples = 4096
private

number of normalize samples

Definition at line 142 of file audio.h.

Referenced by Normalize().

◆ m_packetCounter

int cSoftHdAudio::m_packetCounter = 0
private

packet counter for logging

Definition at line 104 of file audio.h.

Referenced by ClockDriftCompensation().

◆ m_pAlsaMixer

snd_mixer_t* cSoftHdAudio::m_pAlsaMixer = nullptr
private

alsa mixer handle

Definition at line 188 of file audio.h.

Referenced by AlsaExit(), AlsaInitMixer(), and AlsaSetVolume().

◆ m_pAlsaMixerElem

snd_mixer_elem_t* cSoftHdAudio::m_pAlsaMixerElem = nullptr
private

alsa mixer element

Definition at line 189 of file audio.h.

Referenced by AlsaExit(), AlsaInitMixer(), and AlsaSetVolume().

◆ m_pAlsaPCMHandle

snd_pcm_t* cSoftHdAudio::m_pAlsaPCMHandle
private

alsa pcm handle

Definition at line 187 of file audio.h.

Referenced by AlsaExit(), AlsaInitPCMDevice(), AlsaSetup(), ClockDriftCompensation(), CyclicCall(), DropSamplesOlderThanPtsMs(), Enqueue(), FlushAlsaBuffersInternal(), GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), GetOutputPtsMsInternal(), HandleError(), InitFilter(), OpenAlsaDevice(), SendAudio(), SendPause(), and SetHwDelayBaseline().

◆ m_passthrough

int cSoftHdAudio::m_passthrough
private

passthrough mask

Definition at line 124 of file audio.h.

Referenced by AlsaInitPCMDevice(), ClockDriftCompensation(), CyclicCall(), FlushAlsaBuffersInternal(), GetPassthrough(), HandleError(), SendAudio(), SetHwDelayBaseline(), SetPassthroughMask(), and SetVolume().

◆ m_pauseBurst

std::vector<uint16_t> cSoftHdAudio::m_pauseBurst
private

holds the burst data itself

Definition at line 129 of file audio.h.

Referenced by BuildPauseBurst(), EnqueueSpdif(), and SendPause().

◆ m_paused

std::atomic<bool> cSoftHdAudio::m_paused = true
private

audio is paused

Definition at line 121 of file audio.h.

Referenced by CyclicCall(), IsPaused(), and SetPaused().

◆ m_pauseMutex

std::mutex cSoftHdAudio::m_pauseMutex
private

mutex for a safe thread pausing

Definition at line 113 of file audio.h.

Referenced by CyclicCall(), EnqueueSpdif(), and SetPaused().

◆ m_pBuffersinkCtx

AVFilterContext* cSoftHdAudio::m_pBuffersinkCtx
private

Definition at line 177 of file audio.h.

Referenced by FilterGetFrame(), and InitFilter().

◆ m_pBuffersrcCtx

AVFilterContext* cSoftHdAudio::m_pBuffersrcCtx
private

Definition at line 176 of file audio.h.

Referenced by Filter(), and InitFilter().

◆ m_pConfig

cSoftHdConfig* cSoftHdAudio::m_pConfig
private

pointer to config

Definition at line 98 of file audio.h.

Referenced by cSoftHdAudio().

◆ m_pDevice

cSoftHdDevice* cSoftHdAudio::m_pDevice
private

pointer to device

Definition at line 97 of file audio.h.

◆ m_pEventReceiver

IEventReceiver* cSoftHdAudio::m_pEventReceiver
private

pointer to event receiver

Definition at line 99 of file audio.h.

Referenced by ProcessEvents().

◆ m_pFilterGraph

AVFilterGraph* cSoftHdAudio::m_pFilterGraph = nullptr
private

Definition at line 175 of file audio.h.

Referenced by CheckForFilterReady(), Exit(), and InitFilter().

◆ m_pidController

cPidController cSoftHdAudio::m_pidController {3, 0.005, 0, 1000}
private

PID controller for clock drift compensation with tuning values coming from educated guesses.

Definition at line 101 of file audio.h.

Referenced by ClockDriftCompensation(), DropSamplesOlderThanPtsMs(), and FlushBuffers().

◆ m_pitchAdjustFrameCounter

int cSoftHdAudio::m_pitchAdjustFrameCounter = 0
private

counter for pitch adjustment frames

Definition at line 116 of file audio.h.

Referenced by Enqueue().

◆ m_pitchPpm

std::atomic<double> cSoftHdAudio::m_pitchPpm = 0
private

pitch adjustment in ppm. Positive values are faster

Definition at line 115 of file audio.h.

Referenced by ClockDriftCompensation(), Enqueue(), and SetPassthroughMask().

◆ m_pMixerChannel

const char* cSoftHdAudio::m_pMixerChannel
private

mixer channel name

Definition at line 170 of file audio.h.

Referenced by AlsaInitMixer().

◆ m_pMixerDevice

const char* cSoftHdAudio::m_pMixerDevice = nullptr
private

mixer device name (not used)

Definition at line 169 of file audio.h.

Referenced by AlsaInitMixer().

◆ m_pPassthroughDevice

const char* cSoftHdAudio::m_pPassthroughDevice
private

passthrough device name

Definition at line 126 of file audio.h.

Referenced by AlsaInitPCMDevice().

◆ m_pPCMDevice

const char* cSoftHdAudio::m_pPCMDevice
private

PCM device name.

Definition at line 125 of file audio.h.

Referenced by AlsaInitPCMDevice().

◆ m_pRingbuffer

cSoftHdRingbuffer cSoftHdAudio::m_pRingbuffer {RINGBUFFER_SIZE}
private

sample ring buffer

Definition at line 184 of file audio.h.

Referenced by DropSamplesOlderThanPtsMs(), Enqueue(), FlushBuffers(), GetOutputPtsMsInternal(), GetUsedBytes(), IsBufferFull(), and SendAudio().

◆ m_pTimebase

AVRational* cSoftHdAudio::m_pTimebase
private

pointer to AVCodecContext pkts_timebase

Definition at line 111 of file audio.h.

Referenced by InitFilter(), MsToPts(), PtsToMs(), SetTimebase(), and Setup().

◆ m_softVolume

bool cSoftHdAudio::m_softVolume
private

flag to use soft volume

Definition at line 123 of file audio.h.

Referenced by SendAudio(), SetSoftvol(), and SetVolume().

◆ m_spdifBurstSize

int cSoftHdAudio::m_spdifBurstSize = 0
private

size of the current spdif burst

Definition at line 128 of file audio.h.

Referenced by BuildPauseBurst(), CyclicCall(), EnqueueSpdif(), and SendPause().

◆ m_stereoDescent

int cSoftHdAudio::m_stereoDescent
private

volume descent for stereo

Definition at line 160 of file audio.h.

Referenced by SetStereoDescent(), and SetVolume().

◆ m_useEqualizer

int cSoftHdAudio::m_useEqualizer
private

flag to use equalizer

Definition at line 165 of file audio.h.

Referenced by InitFilter(), and SetEq().

◆ m_volume

int cSoftHdAudio::m_volume = 0
private

current volume (0 .. 1000)

Definition at line 161 of file audio.h.

Referenced by SendAudio(), SetStereoDescent(), SetVolume(), and SoftAmplify().

◆ NORMALIZE_MAX_INDEX

constexpr int cSoftHdAudio::NORMALIZE_MAX_INDEX = 128
staticconstexprprivate

number of normalize average samples

Definition at line 95 of file audio.h.

Referenced by FlushAlsaBuffersInternal(), and Normalize().

◆ RINGBUFFER_SIZE

constexpr unsigned cSoftHdAudio::RINGBUFFER_SIZE = 3 * 5 * 7 * 8 * 2 * 1000
staticconstexprprivate

default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)

Definition at line 183 of file audio.h.

The documentation for this class was generated from the following files: