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 (AVRational, int, int, bool)
	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 (const uint16_t *, int, int64_t pts)
	Enqueue prepared spdif bursts in audio output queue.

bool	IsBufferFull (void)

void	FlushBuffers (void)
	Flush audio buffers.

int	GetUsedRingbufferBytes (void)
	Get used bytes in audio ringbuffer.

int	GetUsedRingbufferMs (void)
	Get used ms 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.

bool	HasInputPts (void)

int64_t	GetInputPtsMs (void)

int64_t	GetOutputPtsMs (void)
	Get the output PTS of the ringbuffer.

int	GetAvResyncBorderMs (void)

void	SetVolume (int)
	Set mixer volume (0-1000)

void	SetSoftvol (bool softVolume)

void	SetNormalize (bool, int)
	Set normalize volume parameters.

void	SetCompression (bool, int)
	Set volume compression parameters.

void	SetEqualizer (bool, int[18])
	Set equalizer bands.

void	SetStereoDescent (int)
	Set stereo loudness descent.

void	SetPassthroughMask (int mask)

void	SetAutoAES (bool appendAes)

void	SetTimebase (AVRational timebase)

void	SetDownmix (int downMix)

int	GetPassthroughMask (void) const

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 (const uint16_t *, int, int64_t)
	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	RebuildPauseBurst (int)
	Rebuild the pause spdif burst with the size of the last recognized normal spdif audio if size changed.

void	Stop (void)
	Stop the thread.

void	FlushAlsaBuffers (void)
	Flush alsa buffers.

void	DropAlsaBuffers (void)
	Drop alsa buffers.

bool	CyclicCall (void)
	Cyclic audio playback call.

void	ProcessEvents (void)
	Process queued events and forward them to event receiver.

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.

std::string	BuildChannelMapFilter (const AVChannelLayout &)
	Build the "\|"-separated mappings list for the channelmap filter.

int64_t	GetOutputPtsMsInternal (void)

Private Attributes
cSoftHdDevice *	m_pDevice
	pointer to device

cSoftHdConfig *	m_pConfig
	pointer to config

cAlsaDevice	m_alsa
	alsa device

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_packetCounter = 0
	packet counter for logging

bool	m_initialized = false
	class initialized

std::mutex	m_mutex
	mutex for thread safety

std::mutex	m_pauseMutex
	mutex for a safe thread pausing

std::mutex	m_queueMutex
	mutex for queue safety

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_volume = 0
	current volume (0 .. 1000)

int	m_stereoDescent
	volume descent for stereo

AVRational	m_timebase
	AVCodecContext pkts_timebase.

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_spdifBurstSize = 0
	size of the current spdif burst

std::vector< uint16_t >	m_pauseBurst
	holds the burst data itself

int	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

cAudioProcessor	m_audioProcessor

bool	m_useNormalizer
	flag to use volume normalize

bool	m_useCompressor
	flag to use compress volume

bool	m_useEqualizer
	flag to use equalizer

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

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	AV_SYNC_BORDER_MS = 5000
	absolute max a/v difference in ms which should trigger a resync

static constexpr int	BYTES_PER_SAMPLE = 2
	number of bytes per sample

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 51 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(), SetEqualizer(), 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 993 of file audio.cpp.

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

◆ CheckForFilterReady()

int cSoftHdAudio::CheckForFilterReady ( AVCodecContext * ctx )

private

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

Parameters

ctx	AVCodec audio decoding context

Return values

1	error, init failed
0	filter initiated

Definition at line 646 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 1179 of file audio.cpp.

References cAlsaDevice::FramesToMsDouble(), cAlsaDevice::GetAvailableBufferFrames(), cBufferFillLevelLowPassFilter::GetBufferFillLevelFramesAvg(), cAlsaDevice::GetBufferSizeFrames(), cPidController::GetDTerm(), cPidController::GetITerm(), cPidController::GetPTerm(), cPidController::GetTargetValue(), cAlsaDevice::IsPassthroughActive(), cBufferFillLevelLowPassFilter::IsSettled(), L_SOUND, LOGDEBUG2, m_alsa, m_fillLevel, m_lastPidInvocation, m_packetCounter, m_pidController, m_pitchPpm, cPidController::SetTargetValue(), cPidController::Update(), and cBufferFillLevelLowPassFilter::UpdateAvgBufferFillLevel().

Referenced by cSoftHdDevice::PlayAudio().

◆ 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 1031 of file audio.cpp.

References AUDIO, cAlsaDevice::FramesToBytes(), cAlsaDevice::GetAvailableBufferFrames(), cAlsaDevice::HandleError(), cAlsaDevice::IsPassthroughActive(), m_alsa, m_eventQueue, m_mutex, m_paused, m_pauseMutex, m_queueMutex, m_spdifBurstSize, SendAudio(), SendPause(), and cAlsaDevice::WaitUntilReady().

Referenced by Action().

◆ DropAlsaBuffers()

void cSoftHdAudio::DropAlsaBuffers ( void )

private

Drop alsa buffers.

Definition at line 977 of file audio.cpp.

References cAlsaDevice::FlushBuffers(), m_alsa, m_audioProcessor, cAudioProcessor::ResetCompressor(), and cAudioProcessor::ResetNormalizer().

Referenced by Setup().

◆ 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

ptsMs presentation timestamp in milliseconds - samples older than this will be dropped

Definition at line 421 of file audio.cpp.

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

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

◆ Enqueue()

void cSoftHdAudio::Enqueue	(	const uint16_t *	buffer,
		int	count,
		int64_t	pts
	)

private

Send audio data to ringbuffer.

Parameters

buffer	data buffer
count	number of bytes in data buffer
pts	pts to set

Definition at line 527 of file audio.cpp.

References AV_NOPTS_VALUE, AV_SYNC_BORDER_MS, cAlsaDevice::BytesToFrames(), cAlsaDevice::FramesToBytes(), cAlsaDevice::FramesToPts(), cAlsaDevice::IsPassthroughActive(), L_AV_SYNC, LOGDEBUG2, LOGERROR, m_alsa, m_eventQueue, m_fillLevel, m_inputPts, m_mutex, m_pitchAdjustFrameCounter, m_pitchPpm, m_pRingbuffer, m_queueMutex, m_timebase, cAlsaDevice::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

frame audio frame

Definition at line 453 of file audio.cpp.

References BYTES_PER_SAMPLE, cAudioProcessor::Compress(), Enqueue(), m_audioProcessor, m_useCompressor, m_useNormalizer, and cAudioProcessor::Normalize().

Referenced by Filter().

◆ EnqueueSpdif()

void cSoftHdAudio::EnqueueSpdif	(	const uint16_t *	buffer,
		int	count,
		int64_t	pts
	)

Enqueue prepared spdif bursts in audio output queue.

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

Parameters

buffer	data buffer
count	number of bytes in data buffer
pts	pts of the buffer

Definition at line 511 of file audio.cpp.

References Enqueue(), m_pauseMutex, and RebuildPauseBurst().

Referenced by cAudioDecoder::Passthrough().

◆ 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 949 of file audio.cpp.

References cAlsaDevice::Exit(), L_SOUND, LOGDEBUG2, m_alsa, 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

inframe	incoming audio frame to be filtered
ctx	AVCodec audio decoding context

Return values

1	error, send again
0	running

Definition at line 679 of file audio.cpp.

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

Referenced by cAudioDecoder::DecodePCM().

◆ FilterGetFrame()

AVFrame * cSoftHdAudio::FilterGetFrame ( void )

private

Get frame from filter sink.

Returns: pointer to AVFrame if success, NULL otherwise

Definition at line 613 of file audio.cpp.

References LOGERROR, and m_pBuffersinkCtx.

Referenced by Filter().

◆ FlushAlsaBuffers()

void cSoftHdAudio::FlushAlsaBuffers ( void )

private

Flush alsa buffers.

Definition at line 966 of file audio.cpp.

References cAlsaDevice::FlushBuffers(), m_alsa, m_audioProcessor, cAudioProcessor::ResetCompressor(), and cAudioProcessor::ResetNormalizer().

Referenced by FlushBuffers().

◆ FlushBuffers()

void cSoftHdAudio::FlushBuffers ( void )

Flush audio buffers.

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

Definition at line 722 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().

◆ GetAvResyncBorderMs()

int cSoftHdAudio::GetAvResyncBorderMs ( void )

inline

Definition at line 73 of file audio.h.

References AV_SYNC_BORDER_MS.

Referenced by cVideoRender::FrameDropNecessary().

◆ 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 815 of file audio.cpp.

References AV_NOPTS_VALUE, cAlsaDevice::FramesToMs(), cAlsaDevice::GetHwDelayFrames(), cAlsaDevice::IsRunning(), m_alsa, m_inputPts, and m_mutex.

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 795 of file audio.cpp.

References AV_NOPTS_VALUE, cAlsaDevice::FramesToMs(), cAlsaDevice::GetHwDelayFrames(), GetOutputPtsMsInternal(), cAlsaDevice::IsRunning(), m_alsa, m_hwBaseline, m_inputPts, and m_mutex.

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 832 of file audio.cpp.

References AV_NOPTS_VALUE, GetHardwareOutputPtsMs(), m_alsa, m_timebase, and cAlsaDevice::MsToPts().

Referenced by cSoftHdDevice::GetSTC().

◆ GetInputPtsMs()

int64_t cSoftHdAudio::GetInputPtsMs ( void )

inline

Definition at line 71 of file audio.h.

References m_alsa, m_inputPts, m_timebase, and cAlsaDevice::PtsToMs().

Referenced by cSoftHdDevice::IsBufferingThresholdReached(), and cVideoRender::LogDroppedDuped().

◆ 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 774 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 781 of file audio.cpp.

References cAlsaDevice::BytesToFrames(), cAlsaDevice::FramesToMs(), m_alsa, m_inputPts, m_pRingbuffer, m_timebase, cAlsaDevice::PtsToMs(), and cSoftHdRingbuffer::UsedBytes().

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

◆ GetPassthroughMask()

int cSoftHdAudio::GetPassthroughMask ( void ) const

inline

Definition at line 87 of file audio.h.

References cAlsaDevice::GetPassthroughMask(), and m_alsa.

◆ GetUsedRingbufferBytes()

int cSoftHdAudio::GetUsedRingbufferBytes ( void )

Get used bytes in audio ringbuffer.

Definition at line 745 of file audio.cpp.

References m_mutex, m_pRingbuffer, and cSoftHdRingbuffer::UsedBytes().

Referenced by cVideoRender::LogDroppedDuped().

◆ GetUsedRingbufferMs()

int cSoftHdAudio::GetUsedRingbufferMs ( void )

Get used ms in audio ringbuffer.

Definition at line 755 of file audio.cpp.

References cAlsaDevice::BytesToFrames(), cAlsaDevice::FramesToMs(), m_alsa, m_mutex, m_pRingbuffer, and cSoftHdRingbuffer::UsedBytes().

Referenced by cVideoRender::LogDroppedDuped().

◆ HasInputPts()

bool cSoftHdAudio::HasInputPts ( void )

inline

Definition at line 70 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

0	everything ok
1	didn't support channels, downmix set -> scrap this frame, test next
-1	something gone wrong

Definition at line 204 of file audio.cpp.

References BuildChannelMapFilter(), cAlsaDevice::GetDownmix(), cAudioProcessor::GetEqualizerOptions(), cAlsaDevice::GetHwNumChannels(), cAlsaDevice::GetHwSampleRate(), L_SOUND, LOGDEBUG2, LOGERROR, LOGWARNING, m_alsa, m_audioProcessor, m_filterChanged, m_filterReady, m_pBuffersinkCtx, m_pBuffersrcCtx, m_pFilterGraph, m_useEqualizer, and Setup().

Referenced by CheckForFilterReady().

◆ IsBufferFull()

bool cSoftHdAudio::IsBufferFull ( void )

inline

Definition at line 62 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 59 of file audio.h.

References m_paused.

Referenced by cVideoRender::FrameDropNecessary().

◆ 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 933 of file audio.cpp.

References cAlsaDevice::Init(), LOGFATAL, m_alsa, and m_initialized.

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

◆ ProcessEvents()

void cSoftHdAudio::ProcessEvents ( void )

private

Process queued events and forward them to event receiver.

Definition at line 1161 of file audio.cpp.

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

Referenced by Action().

◆ RebuildPauseBurst()

void cSoftHdAudio::RebuildPauseBurst ( int size )

private

Rebuild the pause spdif burst with the size of the last recognized normal spdif audio if size changed.

Parameters

size	spdif burst size in bytes

Definition at line 479 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_pauseBurst, and m_spdifBurstSize.

Referenced by EnqueueSpdif().

◆ ResetHwDelayBaseline()

void cSoftHdAudio::ResetHwDelayBaseline ( void )

Reset the hw delay baseline.

Definition at line 1149 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

freeAlsaBufferFrames number of frames that can be written to the hardware

Return values

true	if data was written or the write should be scheduled again immediately
false	if no data was written

Definition at line 1085 of file audio.cpp.

References cAudioProcessor::Amplify(), cAlsaDevice::BytesToFrames(), cAlsaDevice::CheckWrittenFrames(), cAlsaDevice::FramesToBytes(), cSoftHdRingbuffer::GetReadPointer(), cAlsaDevice::IsPassthroughActive(), m_alsa, m_audioProcessor, m_fillLevel, m_pRingbuffer, m_softVolume, m_volume, cSoftHdRingbuffer::ReadAdvance(), cAlsaDevice::Write(), 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 1119 of file audio.cpp.

References cAlsaDevice::BytesToFrames(), cAlsaDevice::CheckWrittenFrames(), m_alsa, m_pauseBurst, m_spdifBurstSize, and cAlsaDevice::Write().

Referenced by CyclicCall().

◆ SetAutoAES()

void cSoftHdAudio::SetAutoAES ( bool appendAes )

inline

Definition at line 84 of file audio.h.

References m_alsa, and cAlsaDevice::SetAutoAES().

Referenced by cMenuSetupSoft::Store().

◆ SetCompression()

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

Set volume compression parameters.

Parameters

enable	true, turn on compression
maxfac	max. factor of compression / 1000

Definition at line 894 of file audio.cpp.

References m_audioProcessor, m_useCompressor, and cAudioProcessor::SetCompressor().

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

◆ SetDownmix()

void cSoftHdAudio::SetDownmix ( int downMix )

inline

Definition at line 86 of file audio.h.

References m_alsa, and cAlsaDevice::SetDownmix().

Referenced by cMenuSetupSoft::Store().

◆ SetEqualizer()

void cSoftHdAudio::SetEqualizer	(	bool	enable,
		int	band[18]
	)

Set equalizer bands.

Parameters

enable	set using equalizer
band	setting frequenz bands

Definition at line 906 of file audio.cpp.

References m_audioProcessor, m_filterChanged, m_useEqualizer, and cAudioProcessor::SetEqualizer().

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

◆ SetHwDelayBaseline()

void cSoftHdAudio::SetHwDelayBaseline ( void )

Set the hw delay baseline.

Definition at line 1131 of file audio.cpp.

References cAlsaDevice::FramesToMs(), cAlsaDevice::GetHwDelayFrames(), cAlsaDevice::IsPassthroughActive(), L_SOUND, LOGDEBUG2, m_alsa, m_firstRealAudioReceived, and m_hwBaseline.

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

◆ SetNormalize()

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

Set normalize volume parameters.

Parameters

enable	true, turn on normalize
maxfac	max. factor of normalize / 1000

Definition at line 882 of file audio.cpp.

References m_audioProcessor, m_useNormalizer, and cAudioProcessor::SetNormalizer().

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

◆ SetPassthroughMask()

void cSoftHdAudio::SetPassthroughMask ( int mask )

inline

Definition at line 83 of file audio.h.

References m_alsa, and cAlsaDevice::SetPassthroughMask().

Referenced by cSoftHdDevice::SetPassthroughMask().

◆ SetPaused()

void cSoftHdAudio::SetPaused ( bool pause )

Set audio playback pause state.

Parameters

pause true to pause, false to resume

Definition at line 868 of file audio.cpp.

References L_SOUND, LOGDEBUG2, m_paused, and m_pauseMutex.

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

◆ SetSoftvol()

void cSoftHdAudio::SetSoftvol ( bool softVolume )

inline

Definition at line 76 of file audio.h.

References m_softVolume.

Referenced by cMenuSetupSoft::Store().

◆ SetStereoDescent()

void cSoftHdAudio::SetStereoDescent ( int delta )

Set stereo loudness descent.

Parameters

delta value (/1000) to reduce stereo volume

Definition at line 918 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 85 of file audio.h.

References m_timebase.

Referenced by cAudioDecoder::Passthrough().

◆ Setup()

int cSoftHdAudio::Setup	(	AVRational	timebase,
		int	samplerate,
		int	channels,
		bool	passthrough
	)

Alsa setup wrapper.

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

Parameters

timebase	codec timebase
samplerate	stream samplerate
channels	stream nb of channels
passthrough	passthrough enabled

Return values

0	everything ok
-1	something gone wrong in AlsaSetup
1	no parameter change, no setup needed

Definition at line 582 of file audio.cpp.

References cSoftHdConfig::ConfigAudioDownmix, DropAlsaBuffers(), cAlsaDevice::GetDownmix(), cAlsaDevice::GetHwNumChannels(), cAlsaDevice::GetHwSampleRate(), cAlsaDevice::IsPassthroughActive(), LOGERROR, m_alsa, m_pConfig, m_timebase, cAlsaDevice::Setup(), and Stop().

Referenced by cAudioDecoder::CheckUpdateFormat(), and InitFilter().

◆ SetVolume()

void cSoftHdAudio::SetVolume ( int volume )

Set mixer volume (0-1000)

Parameters

volume volume (0 .. 1000)

Definition at line 846 of file audio.cpp.

References cAlsaDevice::GetHwNumChannels(), cAlsaDevice::IsPassthroughActive(), m_alsa, m_audioProcessor, m_softVolume, m_stereoDescent, m_volume, cAudioProcessor::SetAmplifier(), and cAlsaDevice::SetVolume().

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

◆ Stop()

void cSoftHdAudio::Stop ( void )

private

Stop the thread.

Definition at line 1011 of file audio.cpp.

References LOGDEBUG.

Referenced by Exit(), and Setup().

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 98 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 99 of file audio.h.

Referenced by Enqueue(), and GetAvResyncBorderMs().

◆ BYTES_PER_SAMPLE

constexpr int cSoftHdAudio::BYTES_PER_SAMPLE = 2

staticconstexprprivate

number of bytes per sample

Definition at line 100 of file audio.h.

Referenced by EnqueueFrame().

◆ m_alsa

cAlsaDevice cSoftHdAudio::m_alsa

private

alsa device

Definition at line 104 of file audio.h.

Referenced by BuildChannelMapFilter(), ClockDriftCompensation(), CyclicCall(), DropAlsaBuffers(), DropSamplesOlderThanPtsMs(), Enqueue(), Exit(), FlushAlsaBuffers(), GetHardwareOutputDelayMs(), GetHardwareOutputPtsMs(), GetHardwareOutputPtsTimebaseUnits(), GetInputPtsMs(), GetOutputPtsMsInternal(), GetPassthroughMask(), GetUsedRingbufferMs(), InitFilter(), LazyInit(), SendAudio(), SendPause(), SetAutoAES(), SetDownmix(), SetHwDelayBaseline(), SetPassthroughMask(), Setup(), and SetVolume().

◆ m_audioProcessor

cAudioProcessor cSoftHdAudio::m_audioProcessor

private

Definition at line 145 of file audio.h.

Referenced by DropAlsaBuffers(), EnqueueFrame(), FlushAlsaBuffers(), InitFilter(), SendAudio(), SetCompression(), SetEqualizer(), SetNormalize(), and SetVolume().

◆ m_eventQueue

std::vector<Event> cSoftHdAudio::m_eventQueue

private

event queue for incoming events

Definition at line 116 of file audio.h.

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

◆ m_fillLevel

cBufferFillLevelLowPassFilter cSoftHdAudio::m_fillLevel

private

low pass filter for the buffer fill level

Definition at line 106 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 155 of file audio.h.

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

◆ m_filterReady

int cSoftHdAudio::m_filterReady = 0

private

filter is ready

Definition at line 156 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

int 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_initialized

bool cSoftHdAudio::m_initialized = false

private

class initialized

Definition at line 112 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 124 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 108 of file audio.h.

Referenced by ClockDriftCompensation().

◆ m_mutex

std::mutex cSoftHdAudio::m_mutex

private

mutex for thread safety

Definition at line 113 of file audio.h.

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

◆ m_packetCounter

int cSoftHdAudio::m_packetCounter = 0

private

packet counter for logging

Definition at line 109 of file audio.h.

Referenced by ClockDriftCompensation().

◆ 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 RebuildPauseBurst(), and SendPause().

◆ m_paused

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

private

audio is paused

Definition at line 125 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 114 of file audio.h.

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

◆ m_pBuffersinkCtx

AVFilterContext* cSoftHdAudio::m_pBuffersinkCtx

private

Definition at line 159 of file audio.h.

Referenced by FilterGetFrame(), and InitFilter().

◆ m_pBuffersrcCtx

AVFilterContext* cSoftHdAudio::m_pBuffersrcCtx

private

Definition at line 158 of file audio.h.

Referenced by Filter(), and InitFilter().

◆ m_pConfig

cSoftHdConfig* cSoftHdAudio::m_pConfig

private

pointer to config

Definition at line 103 of file audio.h.

Referenced by cSoftHdAudio(), and Setup().

◆ m_pDevice

cSoftHdDevice* cSoftHdAudio::m_pDevice

private

pointer to device

Definition at line 102 of file audio.h.

◆ m_pEventReceiver

IEventReceiver* cSoftHdAudio::m_pEventReceiver

private

pointer to event receiver

Definition at line 105 of file audio.h.

Referenced by ProcessEvents().

◆ m_pFilterGraph

AVFilterGraph* cSoftHdAudio::m_pFilterGraph = nullptr

private

Definition at line 157 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 107 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 118 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 117 of file audio.h.

Referenced by ClockDriftCompensation(), and Enqueue().

◆ m_pMixerChannel

const char* cSoftHdAudio::m_pMixerChannel

private

mixer channel name

Definition at line 152 of file audio.h.

◆ m_pMixerDevice

const char* cSoftHdAudio::m_pMixerDevice = nullptr

private

mixer device name (not used)

Definition at line 151 of file audio.h.

◆ m_pRingbuffer

cSoftHdRingbuffer cSoftHdAudio::m_pRingbuffer {RINGBUFFER_SIZE}

private

sample ring buffer

Definition at line 167 of file audio.h.

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

◆ m_queueMutex

std::mutex cSoftHdAudio::m_queueMutex

private

mutex for queue safety

Definition at line 115 of file audio.h.

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

◆ m_softVolume

bool cSoftHdAudio::m_softVolume

private

flag to use soft volume

Definition at line 127 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 CyclicCall(), RebuildPauseBurst(), and SendPause().

◆ m_stereoDescent

int cSoftHdAudio::m_stereoDescent

private

volume descent for stereo

Definition at line 121 of file audio.h.

Referenced by SetStereoDescent(), and SetVolume().

◆ m_timebase

AVRational cSoftHdAudio::m_timebase

private

AVCodecContext pkts_timebase.

Definition at line 122 of file audio.h.

Referenced by Enqueue(), GetHardwareOutputPtsTimebaseUnits(), GetInputPtsMs(), GetOutputPtsMsInternal(), SetTimebase(), and Setup().

◆ m_useCompressor

bool cSoftHdAudio::m_useCompressor

private

flag to use compress volume

Definition at line 147 of file audio.h.

Referenced by EnqueueFrame(), and SetCompression().

◆ m_useEqualizer

bool cSoftHdAudio::m_useEqualizer

private

flag to use equalizer

Definition at line 148 of file audio.h.

Referenced by InitFilter(), and SetEqualizer().

◆ m_useNormalizer

bool cSoftHdAudio::m_useNormalizer

private

flag to use volume normalize

Definition at line 146 of file audio.h.

Referenced by EnqueueFrame(), and SetNormalize().

◆ m_volume

int cSoftHdAudio::m_volume = 0

private

current volume (0 .. 1000)

Definition at line 120 of file audio.h.

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

◆ 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 166 of file audio.h.

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

Public Member Functions

Protected Member Functions

Private Member Functions

Private Attributes

Static Private Attributes

Detailed Description

Constructor & Destructor Documentation

◆ cSoftHdAudio()

Member Function Documentation

◆ Action()

◆ CheckForFilterReady()

◆ ClockDriftCompensation()

◆ CyclicCall()

◆ DropAlsaBuffers()

◆ DropSamplesOlderThanPtsMs()

◆ Enqueue()

◆ EnqueueFrame()

◆ EnqueueSpdif()

◆ Exit()

◆ Filter()

◆ FilterGetFrame()

◆ FlushAlsaBuffers()

◆ FlushBuffers()

◆ GetAvResyncBorderMs()

◆ GetHardwareOutputDelayMs()

◆ GetHardwareOutputPtsMs()

◆ GetHardwareOutputPtsTimebaseUnits()

◆ GetInputPtsMs()

◆ GetOutputPtsMs()

◆ GetOutputPtsMsInternal()

◆ GetPassthroughMask()

◆ GetUsedRingbufferBytes()

◆ GetUsedRingbufferMs()

◆ HasInputPts()

◆ InitFilter()

◆ IsBufferFull()

◆ IsPaused()

◆ LazyInit()

◆ ProcessEvents()

◆ RebuildPauseBurst()

◆ ResetHwDelayBaseline()

◆ SendAudio()

◆ SendPause()

◆ SetAutoAES()

◆ SetCompression()

◆ SetDownmix()

◆ SetEqualizer()

◆ SetHwDelayBaseline()

◆ SetNormalize()

◆ SetPassthroughMask()

◆ SetPaused()

◆ SetSoftvol()

◆ SetStereoDescent()

◆ SetTimebase()

◆ Setup()

◆ SetVolume()

◆ Stop()

Member Data Documentation

◆ AUDIO_MIN_BUFFER_FREE

◆ AV_SYNC_BORDER_MS

◆ BYTES_PER_SAMPLE

◆ m_alsa

◆ m_audioProcessor

◆ m_eventQueue

◆ m_fillLevel

◆ m_filterChanged

◆ m_filterReady

◆ m_firstRealAudioReceived

◆ m_hwBaseline

◆ m_initialized

◆ m_inputPts

◆ m_lastPidInvocation

◆ m_mutex

◆ m_packetCounter

◆ m_pauseBurst

◆ m_paused

◆ m_pauseMutex

◆ m_pBuffersinkCtx

◆ m_pBuffersrcCtx

◆ m_pConfig