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vlc_aout.h
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vlc_aout.h
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/*****************************************************************************
* vlc_aout.h : audio output interface
*****************************************************************************
* Copyright (C) 2002-2011 VLC authors and VideoLAN
*
* Authors: Christophe Massiot <[email protected]>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
#ifndef VLC_AOUT_H
#define VLC_AOUT_H 1
#include <assert.h>
/**
* \defgroup audio_output Audio output
* \ingroup output
* @{
* \file
* Audio output modules interface
*/
/* Buffers which arrive in advance of more than AOUT_MAX_ADVANCE_TIME
* will be considered as bogus and be trashed */
#define AOUT_MAX_ADVANCE_TIME (AOUT_MAX_PREPARE_TIME + VLC_TICK_FROM_SEC(1))
/* Buffers which arrive in advance of more than AOUT_MAX_PREPARE_TIME
* will cause the calling thread to sleep */
#define AOUT_MAX_PREPARE_TIME VLC_TICK_FROM_SEC(2)
/* Buffers which arrive after pts - AOUT_MIN_PREPARE_TIME will be trashed
* to avoid too heavy resampling */
#define AOUT_MIN_PREPARE_TIME AOUT_MAX_PTS_ADVANCE
/* Tolerance values from EBU Recommendation 37 */
/** Maximum advance of actual audio playback time to coded PTS,
* above which downsampling will be performed */
#define AOUT_MAX_PTS_ADVANCE VLC_TICK_FROM_MS(40)
/** Maximum delay of actual audio playback time from coded PTS,
* above which upsampling will be performed */
#define AOUT_MAX_PTS_DELAY VLC_TICK_FROM_MS(60)
/* Max acceptable resampling (in %) */
#define AOUT_MAX_RESAMPLING 10
#include "vlc_es.h"
#define AOUT_FMTS_IDENTICAL( p_first, p_second ) ( \
((p_first)->i_format == (p_second)->i_format) \
&& AOUT_FMTS_SIMILAR(p_first, p_second) )
/* Check if i_rate == i_rate and i_channels == i_channels */
#define AOUT_FMTS_SIMILAR( p_first, p_second ) ( \
((p_first)->i_rate == (p_second)->i_rate) \
&& ((p_first)->channel_type == (p_second)->channel_type) \
&& ((p_first)->i_physical_channels == (p_second)->i_physical_channels)\
&& ((p_first)->i_chan_mode == (p_second)->i_chan_mode) )
#define AOUT_FMT_LINEAR( p_format ) \
(aout_BitsPerSample((p_format)->i_format) != 0)
#define VLC_CODEC_SPDIFL VLC_FOURCC('s','p','d','i')
#define VLC_CODEC_SPDIFB VLC_FOURCC('s','p','d','b')
#define AOUT_FMT_SPDIF( p_format ) \
( ((p_format)->i_format == VLC_CODEC_SPDIFL) \
|| ((p_format)->i_format == VLC_CODEC_SPDIFB) \
|| ((p_format)->i_format == VLC_CODEC_A52) \
|| ((p_format)->i_format == VLC_CODEC_DTS) )
#define AOUT_FMT_HDMI( p_format ) \
( (p_format)->i_format == VLC_CODEC_EAC3 \
||(p_format)->i_format == VLC_CODEC_DTSHD \
||(p_format)->i_format == VLC_CODEC_TRUEHD \
||(p_format)->i_format == VLC_CODEC_MLP \
)
/* Values used for the audio-channels object variable */
#define AOUT_VAR_CHAN_UNSET 0 /* must be zero */
#define AOUT_VAR_CHAN_STEREO 1
#define AOUT_VAR_CHAN_RSTEREO 2
#define AOUT_VAR_CHAN_LEFT 3
#define AOUT_VAR_CHAN_RIGHT 4
#define AOUT_VAR_CHAN_DOLBYS 5
#define AOUT_VAR_CHAN_HEADPHONES 6
#define AOUT_VAR_CHAN_MONO 7
/*****************************************************************************
* Main audio output structures
*****************************************************************************/
/* Size of a frame for S/PDIF output. */
#define AOUT_SPDIF_SIZE 6144
/* Number of samples in an A/52 frame. */
#define A52_FRAME_NB 1536
/* FIXME to remove once aout.h is cleaned a bit more */
#include <vlc_block.h>
struct vlc_audio_output_events {
void (*timing_report)(audio_output_t *, vlc_tick_t system_now, vlc_tick_t pts);
void (*volume_report)(audio_output_t *, float);
void (*mute_report)(audio_output_t *, bool);
void (*policy_report)(audio_output_t *, bool);
void (*device_report)(audio_output_t *, const char *);
void (*hotplug_report)(audio_output_t *, const char *, const char *);
void (*restart_request)(audio_output_t *, unsigned);
int (*gain_request)(audio_output_t *, float);
};
/** Audio output object
*
* The audio output object is the abstraction for rendering decoded
* (or pass-through) audio samples. In addition to playing samples,
* the abstraction exposes controls for pause/resume, flush/drain,
* changing the volume or mut flag, and listing and changing output device.
*
* An audio output can be in one of three different states:
* stopped, playing or paused.
* The audio output is always created in stopped state and is always destroyed
* in that state also. It is moved from stopped to playing state by start(),
* and from playing or paused states back to stopped state by stop().
**/
struct audio_output
{
struct vlc_object_t obj;
void *sys; /**< Private data for callbacks */
int (*start)(audio_output_t *, audio_sample_format_t *fmt);
/**< Starts a new stream (mandatory, cannot be NULL).
*
* This callback changes the audio output from stopped to playing state
* (if succesful). After the callback returns, time_get(), play(),
* pause(), flush() and eventually stop() callbacks may be called.
*
* \param fmt input stream sample format upon entry,
* output stream sample format upon return [IN/OUT]
* \return VLC_SUCCESS on success, non-zero on failure
*
* \note This callback can only be called while the audio output is in
* stopped state. There can be only one stream per audio output at a time.
*
* \note This callbacks needs not be reentrant.
*/
void (*stop)(audio_output_t *);
/**< Stops the existing stream (mandatory, cannot be NULL).
*
* This callback terminates the current audio stream,
* and returns the audio output to stopped state.
*
* \note This callback needs not be reentrant.
*/
int (*time_get)(audio_output_t *, vlc_tick_t *delay);
/**< Estimates playback buffer latency (mandatory, cannot be NULL).
*
* This callback computes an estimation of the delay until the current
* tail of the audio output buffer would be rendered. This is essential
* for (lip) synchronization and long term drift between the audio output
* clock and the media upstream clock (if any).
*
* If the audio output clock is exactly synchronized with the system
* monotonic clock (i.e. vlc_tick_now()), then aout_TimeGetDefault() can
* implement this callback. In that case, drain must be implemented (since
* the default implementation uses the delay to wait for the end of the
* stream).
*
* This callback is called before the first play() in order to get the
* initial delay (the hw latency). Most modules won't be able to know this
* latency before the first play. In that case, they should return -1 and
* handle the first play() date, cf. play() documentation.
*
* \param delay pointer to the delay until the next sample to be written
* to the playback buffer is rendered [OUT]
* \return 0 on success, non-zero on failure or lack of data
*
* \note This callback cannot be called in stopped state.
*/
void (*play)(audio_output_t *, block_t *block, vlc_tick_t date);
/**< Queues a block of samples for playback (mandatory, cannot be NULL).
*
* The first play() date (after a flush()/start()) will be most likely in
* the future. Modules that don't know the hw latency before a first play
* (when they return -1 from the first time_get()) will need to handle
* this. They can play a silence buffer with 'length = date - now()', or
* configure their render callback to start at the given date.
*
* \param block block of audio samples
* \param date intended system time to render the first sample
*
* \note This callback cannot be called in stopped state.
*/
void (*pause)( audio_output_t *, bool pause, vlc_tick_t date);
/**< Pauses or resumes playback (mandatory, cannot be NULL).
*
* This callback pauses or resumes audio playback as quickly as possible.
* When pausing, it is desirable to stop producing sound immediately, but
* retain already queued audio samples in the buffer to play when later
* when resuming.
*
* If pausing is impossible, then aout_PauseDefault() can provide a
* fallback implementation of this callback.
*
* \param pause pause if true, resume from pause if false
* \param date timestamp when the pause or resume was requested
*
* \note This callback cannot be called in stopped state.
*/
void (*flush)( audio_output_t *);
/**< Flushes the playback buffers (mandatory, cannot be NULL).
*
* \param wait true to wait for playback of pending buffers (drain),
* false to discard pending buffers (flush)
*
* \note This callback cannot be called in stopped state.
*/
void (*drain)(audio_output_t *);
/**< Drain the playback buffers (can be NULL).
*
* If NULL, the caller will wait for the delay returned by time_get before
* calling stop().
*/
int (*volume_set)(audio_output_t *, float volume);
/**< Changes playback volume (optional, may be NULL).
*
* \param volume requested volume (0. = mute, 1. = nominal)
*
* \note The volume is always a positive number.
*
* \warning A stream may or may not have been started when called.
* \warning This callback may be called concurrently with
* time_get(), play(), pause() or flush().
* It will however be protected against concurrent calls to
* start(), stop(), volume_set(), mute_set() or device_select().
*/
int (*mute_set)(audio_output_t *, bool mute);
/**< Changes muting (optinal, may be NULL).
*
* \param mute true to mute, false to unmute
* \warning The same constraints apply as with volume_set().
*/
int (*device_select)(audio_output_t *, const char *id);
/**< Selects an audio output device (optional, may be NULL).
*
* \param id nul-terminated device unique identifier.
* \return 0 on success, non-zero on failure.
*
* \warning The same constraints apply as with volume_set().
*/
struct {
bool headphones; /**< Default to false, set it to true if the current
sink is using headphones */
} current_sink_info;
/**< Current sink informations set by the module from the start() function */
const struct vlc_audio_output_events *events;
};
typedef enum
{
AOUT_CHANIDX_DISABLE = -1,
AOUT_CHANIDX_LEFT,
AOUT_CHANIDX_RIGHT,
AOUT_CHANIDX_MIDDLELEFT,
AOUT_CHANIDX_MIDDLERIGHT,
AOUT_CHANIDX_REARLEFT,
AOUT_CHANIDX_REARRIGHT,
AOUT_CHANIDX_REARCENTER,
AOUT_CHANIDX_CENTER,
AOUT_CHANIDX_LFE,
AOUT_CHANIDX_MAX
} vlc_chan_order_idx_t;
static_assert(AOUT_CHANIDX_MAX == AOUT_CHAN_MAX, "channel count mismatch");
#define AOUT_CHAN_REMAP_INIT { \
AOUT_CHANIDX_LEFT, \
AOUT_CHANIDX_RIGHT, \
AOUT_CHANIDX_MIDDLELEFT, \
AOUT_CHANIDX_MIDDLERIGHT, \
AOUT_CHANIDX_REARLEFT, \
AOUT_CHANIDX_REARRIGHT, \
AOUT_CHANIDX_REARCENTER, \
AOUT_CHANIDX_CENTER, \
AOUT_CHANIDX_LFE, \
}
/**
* It describes the audio channel order VLC expect.
*/
static const uint32_t pi_vlc_chan_order_wg4[] =
{
AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT,
AOUT_CHAN_MIDDLELEFT, AOUT_CHAN_MIDDLERIGHT,
AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT, AOUT_CHAN_REARCENTER,
AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0
};
#define AOUT_RESTART_FILTERS 0x1
#define AOUT_RESTART_OUTPUT (AOUT_RESTART_FILTERS|0x2)
#define AOUT_RESTART_STEREOMODE (AOUT_RESTART_OUTPUT|0x4)
/*****************************************************************************
* Prototypes
*****************************************************************************/
/**
* This function computes the reordering needed to go from pi_chan_order_in to
* pi_chan_order_out.
* If pi_chan_order_in or pi_chan_order_out is NULL, it will assume that vlc
* internal (WG4) order is requested.
*/
VLC_API unsigned aout_CheckChannelReorder( const uint32_t *, const uint32_t *,
uint32_t mask, uint8_t *table );
VLC_API void aout_ChannelReorder(void *, size_t, uint8_t, const uint8_t *, vlc_fourcc_t);
VLC_API void aout_Interleave(void *dst, const void *const *planes,
unsigned samples, unsigned channels,
vlc_fourcc_t fourcc);
VLC_API void aout_Deinterleave(void *dst, const void *src, unsigned samples,
unsigned channels, vlc_fourcc_t fourcc);
/**
* This function will compute the extraction parameter into pi_selection to go
* from i_channels with their type given by pi_order_src[] into the order
* describe by pi_order_dst.
* It will also set :
* - *pi_channels as the number of channels that will be extracted which is
* lower (in case of non understood channels type) or equal to i_channels.
* - the layout of the channels (*pi_layout).
*
* It will return true if channel extraction is really needed, in which case
* aout_ChannelExtract must be used
*
* XXX It must be used when the source may have channel type not understood
* by VLC. In this case the channel type pi_order_src[] must be set to 0.
* XXX It must also be used if multiple channels have the same type.
*/
VLC_API bool aout_CheckChannelExtraction( int *pi_selection, uint32_t *pi_layout, int *pi_channels, const uint32_t pi_order_dst[AOUT_CHAN_MAX], const uint32_t *pi_order_src, int i_channels );
/**
* Do the actual channels extraction using the parameters created by
* aout_CheckChannelExtraction.
*
* XXX this function does not work in place (p_dst and p_src must not overlap).
* XXX Only 8, 16, 32, 64 bits per sample are supported.
*/
VLC_API void aout_ChannelExtract( void *p_dst, int i_dst_channels, const void *p_src, int i_src_channels, int i_sample_count, const int *pi_selection, int i_bits_per_sample );
/* */
static inline unsigned aout_FormatNbChannels(const audio_sample_format_t *fmt)
{
return vlc_popcount(fmt->i_physical_channels);
}
VLC_API unsigned int aout_BitsPerSample( vlc_fourcc_t i_format ) VLC_USED;
VLC_API void aout_FormatPrepare( audio_sample_format_t * p_format );
VLC_API void aout_FormatPrint(vlc_object_t *, const char *,
const audio_sample_format_t *);
#define aout_FormatPrint(o, t, f) aout_FormatPrint(VLC_OBJECT(o), t, f)
VLC_API const char * aout_FormatPrintChannels( const audio_sample_format_t * ) VLC_USED;
#define AOUT_VOLUME_DEFAULT 256
#define AOUT_VOLUME_MAX 512
VLC_API float aout_VolumeGet (audio_output_t *);
VLC_API int aout_VolumeSet (audio_output_t *, float);
VLC_API int aout_VolumeUpdate (audio_output_t *, int, float *);
VLC_API int aout_MuteGet (audio_output_t *);
VLC_API int aout_MuteSet (audio_output_t *, bool);
VLC_API char *aout_DeviceGet (audio_output_t *);
VLC_API int aout_DeviceSet (audio_output_t *, const char *);
VLC_API int aout_DevicesList (audio_output_t *, char ***, char ***);
/**
* Report change of configured audio volume to the core and UI.
*/
static inline void aout_VolumeReport(audio_output_t *aout, float volume)
{
aout->events->volume_report(aout, volume);
}
/**
* Report change of muted flag to the core and UI.
*/
static inline void aout_MuteReport(audio_output_t *aout, bool mute)
{
aout->events->mute_report(aout, mute);
}
/**
* Report audio policy status.
* \param cork true to request a cork, false to undo any pending cork.
*/
static inline void aout_PolicyReport(audio_output_t *aout, bool cork)
{
aout->events->policy_report(aout, cork);
}
/**
* Report change of output device.
*/
static inline void aout_DeviceReport(audio_output_t *aout, const char *id)
{
aout->events->device_report(aout, id);
}
/**
* Report a device hot-plug event.
* @param id device ID
* @param name human-readable device name (NULL for hot unplug)
*/
static inline void aout_HotplugReport(audio_output_t *aout,
const char *id, const char *name)
{
aout->events->hotplug_report(aout, id, name);
}
/**
* Request a change of software audio amplification.
* \param gain linear amplitude gain (must be positive)
* \warning Values in excess 1.0 may cause overflow and distorsion.
*/
static inline int aout_GainRequest(audio_output_t *aout, float gain)
{
return aout->events->gain_request(aout, gain);
}
static inline void aout_RestartRequest(audio_output_t *aout, unsigned mode)
{
aout->events->restart_request(aout, mode);
}
/**
* Default implementation for audio_output_t.time_get
*/
static inline int aout_TimeGetDefault(audio_output_t *aout,
vlc_tick_t *restrict delay)
{
(void) aout; (void) delay;
return -1;
}
/**
* Default implementation for audio_output_t.pause
*
* \warning This default callback implementation is suboptimal as it will
* discard some audio samples.
* Do not use this unless there are really no possible better alternatives.
*/
static inline void aout_PauseDefault(audio_output_t *aout, bool paused,
vlc_tick_t date)
{
if (paused)
aout->flush(aout);
(void) date;
}
/* Audio output filters */
/**
* Enable or disable an audio filter ("audio-filter")
*
* \param aout a valid audio output
* \param name a valid filter name
* \param add true to add the filter, false to remove it
* \return 0 on success, non-zero on failure.
*/
VLC_API int aout_EnableFilter(audio_output_t *aout, const char *name, bool add);
typedef struct
{
/**
* If the remap order differs from the WG4 order, a remap audio filter will
* be inserted to remap channels according to this array.
*/
int remap[AOUT_CHANIDX_MAX];
/**
* If true, a filter will be inserted to add a headphones effect (like a
* binauralizer audio filter).
*/
bool headphones;
} aout_filters_cfg_t;
#define AOUT_FILTERS_CFG_INIT (aout_filters_cfg_t) \
{ .remap = AOUT_CHAN_REMAP_INIT, \
.headphones = false, \
};
typedef struct aout_filters aout_filters_t;
VLC_API aout_filters_t *aout_FiltersNew(vlc_object_t *,
const audio_sample_format_t *,
const audio_sample_format_t *,
const aout_filters_cfg_t *cfg) VLC_USED;
#define aout_FiltersNew(o,inf,outf,remap) \
aout_FiltersNew(VLC_OBJECT(o),inf,outf,remap)
VLC_API void aout_FiltersDelete(vlc_object_t *, aout_filters_t *);
#define aout_FiltersDelete(o,f) \
aout_FiltersDelete(VLC_OBJECT(o),f)
VLC_API bool aout_FiltersAdjustResampling(aout_filters_t *, int);
VLC_API block_t *aout_FiltersPlay(aout_filters_t *, block_t *, float rate);
VLC_API block_t *aout_FiltersDrain(aout_filters_t *);
VLC_API void aout_FiltersFlush(aout_filters_t *);
VLC_API void aout_FiltersChangeViewpoint(aout_filters_t *, const vlc_viewpoint_t *vp);
VLC_API vout_thread_t *aout_filter_GetVout(filter_t *, const video_format_t *);
/** @} */
#endif /* VLC_AOUT_H */