cSoftHdAudio - Audio class More...

#include <audio.h>

Public Member Functions
	cSoftHdAudio (cSoftHdDevice *)
	cSoftHdAudio constructor

void	LazyInit (void)
	Initialize audio output module.

void	Exit (void)
	Cleanup audio output module.

int	Setup (AVCodecContext *, int, int, int)
	Setup alsa.

void	SetPaused (bool)
	Set audio playback paused state.

bool	IsPaused (void)

void	Filter (AVFrame , AVCodecContext )
	Send audio frame to filter and enqueue it.

void	Enqueue (uint16_t , int, AVFrame )
	Send audio data to ringbuffer.

bool	IsBufferFull (void)

void	FlushBuffers (void)
	Flush audio buffers.

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

int	GetFreeBytes (void)
	Get free bytes in audio ringbuffer.

int64_t	GetHardwareOutputPtsMs (void)
	Get the hardware output PTS in milliseconds.

int64_t	GetHardwareOutputPtsTimebaseUnits (void)
	Get the hardware output PTS in timebase units.

int	GetPassthrough (void) const

bool	HasPts (void)

int64_t	GetInputPtsMs (void)

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

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	SetPassthrough (int)
	Set audio passthrough mask.

void	SetAutoAES (bool appendAes)

void	SetTimebase (AVRational *timebase)

void	FlushAlsaBuffers (void)
	Flush alsa buffers.

bool	CyclicCall (void)
	Cyclic audio playback call.

void	DropSamplesOlderThanPtsMs (int64_t)
	Drop samples older than the given PTS.

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

void	ClockDriftCompensation (void)
	Calculate clock drift compensation.

Private Member Functions
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.

void	Normalize (uint16_t *, int)
	Normalize audio.

void	Compress (uint16_t *, int)
	Compress audio.

void	SoftAmplify (int16_t *, int)
	Software amplifier.

int	InitFilter (AVCodecContext *)
	Init filter.

void	EnqueueFrame (AVFrame *)
	Place samples in audio output queue.

void	HandleError (int)
	handle error

char *	OpenAlsaDevice (const char *, int)
	Open alsa device.

char *	FindAlsaDevice (const char , const char , int)
	Find alsa device giving some search hints.

int	AlsaSetup (int channels, int sample_rate, int passthrough)
	Setup alsa audio for requested format.

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 alsa audio output module.

void	AlsaExit (void)
	Cleanup alsa audio output module.

int64_t	PtsToMs (int64_t pts)

int64_t	MsToPts (int64_t ptsMs)

int	MsToFrames (int milliseconds)

int	FramesToMs (int frames)

double	FramesToMsDouble (int frames)

int64_t	GetOutputPtsMsInternal (void)

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

cAudioThread *	m_pAudioThread = nullptr
	pointer to audio thread

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

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
	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 unsigned	RINGBUFFER_SIZE = 3 * 5 * 7 * 8 * 2 * 1000
	default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)

Detailed Description

cSoftHdAudio - Audio class

Definition at line 51 of file audio.h.

Constructor & Destructor Documentation

◆ cSoftHdAudio()

cSoftHdAudio::cSoftHdAudio ( cSoftHdDevice * device )

cSoftHdAudio constructor

Definition at line 61 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

◆ AlsaExit()

void cSoftHdAudio::AlsaExit ( void )

private

Cleanup alsa audio output module.

Definition at line 1555 of file audio.cpp.

References m_pAlsaMixer, m_pAlsaMixerElem, and m_pAlsaPCMHandle.

Referenced by Exit().

◆ AlsaInit()

void cSoftHdAudio::AlsaInit ( void )

private

Initialize alsa audio output module.

Definition at line 1539 of file audio.cpp.

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

Referenced by LazyInit().

◆ AlsaInitMixer()

void cSoftHdAudio::AlsaInitMixer ( void )

private

Initialize alsa mixer.

Definition at line 1360 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 1281 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

channels	Channels requested
sample_rate	SampleRate requested
passthrough	use pass-through (AC-3, ...) device

Return values

0	everything ok
1	didn't support hw channels, downmix set -> retest
-1	something gone wrong

Todo:: FIXME: remove pointer for freq + channels

Definition at line 1437 of file audio.cpp.

References FlushAlsaBuffers(), L_SOUND, LOGDEBUG2, LOGERROR, LOGINFO, LOGWARNING, m_alsaBufferSizeFrames, m_alsaUseMmap, m_downmix, m_hwNumChannels, m_hwSampleRate, m_pAlsaPCMHandle, m_pAudioThread, MsToFrames(), and cAudioThread::Stop().

Referenced by InitFilter(), and Setup().

◆ AlsaSetVolume()

void cSoftHdAudio::AlsaSetVolume ( int volume )

private

Set alsa mixer volume (0-1000)

Parameters

volume volume (0 .. 1000)

Definition at line 1414 of file audio.cpp.

References m_alsaRatio, m_pAlsaMixer, and m_pAlsaMixerElem.

Referenced by SetVolume().

◆ 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 739 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 1588 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.

Parameters

samples	sample buffer
count	number of bytes in sample buffer

Definition at line 233 of file audio.cpp.

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

Referenced by EnqueueFrame().

◆ CyclicCall()

bool cSoftHdAudio::CyclicCall ( void )

Cyclic audio playback call.

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

Returns: true if data was written, false otherwise

Definition at line 1109 of file audio.cpp.

References cSoftHdRingbuffer::GetReadPointer(), HandleError(), LOGERROR, LOGWARNING, m_alsaUseMmap, m_fillLevel, m_mutex, m_pAlsaPCMHandle, m_passthrough, m_paused, m_pauseMutex, m_pRingbuffer, m_softVolume, m_volume, cSoftHdRingbuffer::ReadAdvance(), SoftAmplify(), and cBufferFillLevelLowPassFilter::WroteFrames().

Referenced by cAudioThread::Action().

◆ 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 570 of file audio.cpp.

References GetOutputPtsMsInternal(), HasPts(), 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
	)

Send audio data to ringbuffer.

Parameters

buffer	data buffer
count	number of bytes in data buffer
frame	decoded frame (used to get frame parameters)

Definition at line 632 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 cAudioDecoder::DecodePassthrough(), and EnqueueFrame().

◆ EnqueueFrame()

void cSoftHdAudio::EnqueueFrame ( AVFrame * frame )

private

Place samples in audio output queue.

Parameters

frame audio frame

Definition at line 602 of file audio.cpp.

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

Referenced by Filter().

◆ Exit()

void cSoftHdAudio::Exit ( void )

Cleanup audio output module.

Definition at line 1029 of file audio.cpp.

References AlsaExit(), L_SOUND, LOGDEBUG2, m_initialized, m_pAudioThread, m_pFilterGraph, and cAudioThread::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 772 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 706 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

devname	interface identification (e.g. "pcm")
hint	string to compare with device name hints
passthrough	set, 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 1245 of file audio.cpp.

References LOGWARNING, and OpenAlsaDevice().

Referenced by AlsaInitPCMDevice().

◆ FlushAlsaBuffers()

void cSoftHdAudio::FlushAlsaBuffers ( void )

Flush alsa buffers.

Definition at line 1066 of file audio.cpp.

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

Referenced by AlsaSetup(), and FlushBuffers().

◆ FlushBuffers()

void cSoftHdAudio::FlushBuffers ( void )

Flush audio buffers.

Stop alsa player if running, otherwise flush the alsa buffers and force a filter init

Definition at line 816 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::ClearAudio().

◆ FramesToMs()

int cSoftHdAudio::FramesToMs ( int frames )

inlineprivate

Definition at line 196 of file audio.h.

References m_hwSampleRate.

Referenced by GetHardwareOutputPtsMs(), and GetOutputPtsMsInternal().

◆ FramesToMsDouble()

double cSoftHdAudio::FramesToMsDouble ( int frames )

inlineprivate

Definition at line 197 of file audio.h.

References m_hwSampleRate.

Referenced by ClockDriftCompensation().

◆ GetFreeBytes()

int cSoftHdAudio::GetFreeBytes ( void )

Get free bytes in audio ringbuffer.

Definition at line 839 of file audio.cpp.

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

◆ 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 888 of file audio.cpp.

References AV_NOPTS_VALUE, FramesToMs(), GetOutputPtsMsInternal(), L_SOUND, LOGDEBUG2, 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 912 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 71 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 867 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 874 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 69 of file audio.h.

References m_passthrough.

◆ GetUsedBytes()

int cSoftHdAudio::GetUsedBytes ( void )

Get used bytes in audio ringbuffer.

Definition at line 849 of file audio.cpp.

References m_pRingbuffer, and cSoftHdRingbuffer::UsedBytes().

Referenced by cVideoRender::LogDroppedDuped().

◆ HandleError()

void cSoftHdAudio::HandleError ( int error )

private

handle error

Definition at line 1052 of file audio.cpp.

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

Referenced by CyclicCall().

◆ HasPts()

bool cSoftHdAudio::HasPts ( 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 filter.

Return values

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

Definition at line 395 of file audio.cpp.

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

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::DisplayFrame().

◆ LazyInit()

void cSoftHdAudio::LazyInit ( void )

Initialize audio output module.

Definition at line 1018 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 195 of file audio.h.

References m_hwSampleRate.

Referenced by AlsaSetup(), and DropSamplesOlderThanPtsMs().

◆ MsToPts()

int64_t cSoftHdAudio::MsToPts ( int64_t ptsMs )

inlineprivate

Definition at line 194 of file audio.h.

References m_pTimebase.

Referenced by GetHardwareOutputPtsTimebaseUnits().

◆ Normalize()

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

private

Normalize audio.

Parameters

samples	sample buffer
count	number of bytes in sample buffer

Definition at line 152 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 alsa device.

Parameters

device	alsa device
passthrough	set, if this is a passthrough device

Returns: the alsa device if successful, NULL otherwise

Definition at line 1193 of file audio.cpp.

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

Referenced by AlsaInitPCMDevice(), and FindAlsaDevice().

◆ ProcessEvents()

void cSoftHdAudio::ProcessEvents ( void )

Process queued events and forward to event receiver.

Definition at line 1571 of file audio.cpp.

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

Referenced by cAudioThread::Action().

◆ PtsToMs()

int64_t cSoftHdAudio::PtsToMs ( int64_t pts )

inlineprivate

Definition at line 193 of file audio.h.

References m_pTimebase.

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

◆ 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

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

Definition at line 975 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

band	setting frequenz bands
onoff	set using equalizer

Definition at line 319 of file audio.cpp.

References m_equalizerBand, m_filterChanged, and m_useEqualizer.

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

◆ 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 963 of file audio.cpp.

References m_normalize, and m_normalizeMaxFactor.

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

◆ SetPassthrough()

void cSoftHdAudio::SetPassthrough ( int mask )

Set audio passthrough mask.

Parameters

mask	passthrough mask (as a bitmask)

Definition at line 1004 of file audio.cpp.

References m_passthrough, and m_pitchPpm.

Referenced by cSoftHdDevice::SetPassthrough().

◆ SetPaused()

void cSoftHdAudio::SetPaused ( bool pause )

Set audio playback paused state.

Parameters

pause true to pause, false to resume

Definition at line 948 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

delta value (/1000) to reduce stereo volume

Definition at line 993 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
	)

Setup alsa.

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

Parameters

AudioCtx	AVCodec audio decoding context
samplerate	stream samplerate
channels	stream nb of channels
passthrough	passthrough enabled

Return values

0	everything ok
err	something gone wrong

Definition at line 683 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

volume volume (0 .. 1000)

Definition at line 925 of file audio.cpp.

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

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

◆ SoftAmplify()

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

private

Software amplifier.

Parameters

samples	sample buffer
count	number of bytes in sample buffer

Todo:: FIXME: this does hard clipping

Definition at line 290 of file audio.cpp.

References m_amplifier, m_bytesPerSample, and m_volume.

Referenced by CyclicCall().

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 92 of file audio.h.

Referenced by IsBufferFull().

◆ m_alsaBufferSizeFrames

int cSoftHdAudio::m_alsaBufferSizeFrames = 0

private

alsa buffer size in frames

Definition at line 99 of file audio.h.

Referenced by AlsaSetup(), and ClockDriftCompensation().

◆ m_alsaRatio

int cSoftHdAudio::m_alsaRatio

private

internal -> mixer ratio * 1000

Definition at line 181 of file audio.h.

Referenced by AlsaInitMixer(), and AlsaSetVolume().

◆ m_alsaUseMmap

bool cSoftHdAudio::m_alsaUseMmap

private

use mmap

Definition at line 182 of file audio.h.

Referenced by AlsaSetup(), and CyclicCall().

◆ m_amplifier

int cSoftHdAudio::m_amplifier

private

software volume amplify factor

Definition at line 145 of file audio.h.

Referenced by SetVolume(), and SoftAmplify().

◆ m_appendAES

bool cSoftHdAudio::m_appendAES

private

flag ato utomatic append AES

Definition at line 126 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 107 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 140 of file audio.h.

Referenced by EnqueueFrame(), and SetCompression().

◆ m_compressionFactor

int cSoftHdAudio::m_compressionFactor = 0

private

current compression factor

Definition at line 141 of file audio.h.

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

◆ m_compressionMaxFactor

int cSoftHdAudio::m_compressionMaxFactor

private

max. compression factor

Definition at line 142 of file audio.h.

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

◆ m_downmix

int cSoftHdAudio::m_downmix

private

set stereo downmix

Definition at line 117 of file audio.h.

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

◆ m_equalizerBand

float cSoftHdAudio::m_equalizerBand[18]

private

equalizer band

Definition at line 151 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 113 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 96 of file audio.h.

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

◆ m_filterChanged

int cSoftHdAudio::m_filterChanged = 0

private

filter has changed

Definition at line 158 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 159 of file audio.h.

Referenced by CheckForFilterReady(), and InitFilter().

◆ m_hwNumChannels

unsigned int cSoftHdAudio::m_hwNumChannels = 0

private

number of hardware channels

Definition at line 109 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 108 of file audio.h.

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

◆ m_initialized

bool cSoftHdAudio::m_initialized = false

private

class initialized

Definition at line 106 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 119 of file audio.h.

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

◆ m_lastPidInvocation

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

private

last time the PID controller was invoked

Definition at line 98 of file audio.h.

Referenced by ClockDriftCompensation().

◆ m_mutex

std::mutex cSoftHdAudio::m_mutex

private

mutex for thread safety

Definition at line 111 of file audio.h.

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

◆ m_normalize

bool cSoftHdAudio::m_normalize

private

flag to use volume normalize

Definition at line 129 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 132 of file audio.h.

Referenced by FlushAlsaBuffers(), and Normalize().

◆ m_normalizeCounter

int cSoftHdAudio::m_normalizeCounter

private

normalize sample counter

Definition at line 131 of file audio.h.

Referenced by FlushAlsaBuffers(), and Normalize().

◆ m_normalizeFactor

int cSoftHdAudio::m_normalizeFactor

private

current normalize factor

Definition at line 135 of file audio.h.

Referenced by FlushAlsaBuffers(), and Normalize().

◆ m_normalizeIndex

int cSoftHdAudio::m_normalizeIndex

private

index into normalize average table

Definition at line 133 of file audio.h.

Referenced by Normalize().

◆ m_normalizeMaxFactor

int cSoftHdAudio::m_normalizeMaxFactor

private

max. normalize factor

Definition at line 137 of file audio.h.

Referenced by Normalize(), and SetNormalize().

◆ m_normalizeMinFactor

const int cSoftHdAudio::m_normalizeMinFactor = 100

private

min. normalize factor

Definition at line 136 of file audio.h.

Referenced by Normalize().

◆ m_normalizeReady

int cSoftHdAudio::m_normalizeReady

private

index normalize counter

Definition at line 134 of file audio.h.

Referenced by FlushAlsaBuffers(), and Normalize().

◆ m_normalizeSamples

const int cSoftHdAudio::m_normalizeSamples = 4096

private

number of normalize samples

Definition at line 130 of file audio.h.

Referenced by Normalize().

◆ m_packetCounter

int cSoftHdAudio::m_packetCounter = 0

private

packet counter for logging

Definition at line 100 of file audio.h.

Referenced by ClockDriftCompensation().

◆ m_pAlsaMixer

snd_mixer_t* cSoftHdAudio::m_pAlsaMixer = nullptr

private

alsa mixer handle

Definition at line 179 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 180 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 178 of file audio.h.

Referenced by AlsaExit(), AlsaInitPCMDevice(), AlsaSetup(), ClockDriftCompensation(), CyclicCall(), DropSamplesOlderThanPtsMs(), Enqueue(), FlushAlsaBuffers(), GetHardwareOutputPtsMs(), GetOutputPtsMsInternal(), HandleError(), and OpenAlsaDevice().

◆ m_passthrough

int cSoftHdAudio::m_passthrough

private

passthrough mask

Definition at line 123 of file audio.h.

Referenced by AlsaInitPCMDevice(), ClockDriftCompensation(), CyclicCall(), GetPassthrough(), HandleError(), SetPassthrough(), and SetVolume().

◆ m_pAudioThread

cAudioThread* cSoftHdAudio::m_pAudioThread = nullptr

private

pointer to audio thread

Definition at line 103 of file audio.h.

Referenced by AlsaSetup(), and Exit().

◆ m_paused

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

private

audio is paused

Definition at line 120 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 112 of file audio.h.

Referenced by CyclicCall(), and SetPaused().

◆ m_pBuffersinkCtx

AVFilterContext* cSoftHdAudio::m_pBuffersinkCtx

private

Definition at line 162 of file audio.h.

Referenced by FilterGetFrame(), and InitFilter().

◆ m_pBuffersrcCtx

AVFilterContext* cSoftHdAudio::m_pBuffersrcCtx

private

Definition at line 161 of file audio.h.

Referenced by Filter(), and InitFilter().

◆ m_pConfig

cSoftHdConfig* cSoftHdAudio::m_pConfig

private

pointer to config

Definition at line 94 of file audio.h.

Referenced by cSoftHdAudio().

◆ m_pDevice

cSoftHdDevice* cSoftHdAudio::m_pDevice

private

pointer to device

Definition at line 93 of file audio.h.

◆ m_pEventReceiver

IEventReceiver* cSoftHdAudio::m_pEventReceiver

private

pointer to event receiver

Definition at line 95 of file audio.h.

Referenced by ProcessEvents().

◆ m_pFilterGraph

AVFilterGraph* cSoftHdAudio::m_pFilterGraph = nullptr

private

Definition at line 160 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 97 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 115 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 114 of file audio.h.

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

◆ m_pMixerChannel

const char* cSoftHdAudio::m_pMixerChannel

private

mixer channel name

Definition at line 155 of file audio.h.

Referenced by AlsaInitMixer().

◆ m_pMixerDevice

const char* cSoftHdAudio::m_pMixerDevice = nullptr

private

mixer device name (not used)

Definition at line 154 of file audio.h.

Referenced by AlsaInitMixer().

◆ m_pPassthroughDevice

const char* cSoftHdAudio::m_pPassthroughDevice

private

passthrough device name

Definition at line 125 of file audio.h.

Referenced by AlsaInitPCMDevice().

◆ m_pPCMDevice

const char* cSoftHdAudio::m_pPCMDevice

private

PCM device name.

Definition at line 124 of file audio.h.

Referenced by AlsaInitPCMDevice().

◆ m_pRingbuffer

cSoftHdRingbuffer cSoftHdAudio::m_pRingbuffer {RINGBUFFER_SIZE}

private

sample ring buffer

Definition at line 168 of file audio.h.

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

◆ m_pTimebase

AVRational* cSoftHdAudio::m_pTimebase

private

pointer to AVCodecContext pkts_timebase

Definition at line 110 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 122 of file audio.h.

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

◆ m_stereoDescent

int cSoftHdAudio::m_stereoDescent

private

volume descent for stereo

Definition at line 146 of file audio.h.

Referenced by SetStereoDescent(), and SetVolume().

◆ m_useEqualizer

int cSoftHdAudio::m_useEqualizer

private

flag to use equalizer

Definition at line 150 of file audio.h.

Referenced by InitFilter(), and SetEq().

◆ m_volume

int cSoftHdAudio::m_volume

private

current volume (0 .. 1000)

Definition at line 147 of file audio.h.

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

◆ 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 167 of file audio.h.

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

Public Member Functions

Private Member Functions

Private Attributes

Static Private Attributes

Detailed Description

Constructor & Destructor Documentation

◆ cSoftHdAudio()

Member Function Documentation

◆ AlsaExit()

◆ AlsaInit()

◆ AlsaInitMixer()

◆ AlsaInitPCMDevice()

◆ AlsaSetup()

◆ AlsaSetVolume()

◆ CheckForFilterReady()

◆ ClockDriftCompensation()

◆ Compress()

◆ CyclicCall()

◆ DropSamplesOlderThanPtsMs()

◆ Enqueue()

◆ EnqueueFrame()

◆ Exit()

◆ Filter()

◆ FilterGetFrame()

◆ FindAlsaDevice()

◆ FlushAlsaBuffers()

◆ FlushBuffers()

◆ FramesToMs()

◆ FramesToMsDouble()

◆ GetFreeBytes()

◆ GetHardwareOutputPtsMs()

◆ GetHardwareOutputPtsTimebaseUnits()

◆ GetInputPtsMs()

◆ GetOutputPtsMs()

◆ GetOutputPtsMsInternal()

◆ GetPassthrough()

◆ GetUsedBytes()

◆ HandleError()

◆ HasPts()

◆ InitFilter()

◆ IsBufferFull()

◆ IsPaused()

◆ LazyInit()

◆ MsToFrames()

◆ MsToPts()

◆ Normalize()

◆ OpenAlsaDevice()

◆ ProcessEvents()

◆ PtsToMs()

◆ SetAutoAES()

◆ SetCompression()

◆ SetDownmix()

◆ SetEq()

◆ SetNormalize()

◆ SetPassthrough()

◆ SetPaused()

◆ SetSoftvol()

◆ SetStereoDescent()

◆ SetTimebase()

◆ Setup()

◆ SetVolume()

◆ SoftAmplify()

Member Data Documentation

◆ AUDIO_MIN_BUFFER_FREE

◆ m_alsaBufferSizeFrames

◆ m_alsaRatio

◆ m_alsaUseMmap

◆ m_amplifier

◆ m_appendAES

◆ m_bytesPerSample

◆ m_compression

◆ m_compressionFactor

◆ m_compressionMaxFactor

◆ m_downmix

◆ m_equalizerBand

◆ m_eventQueue

◆ m_fillLevel

◆ m_filterChanged

◆ m_filterReady

◆ m_hwNumChannels