gstreamer/ext/alsa/gstalsa.h
Wim Taymans 1bb09c4352 ext/alsa/: Use alsa trigger_tstamp to get the timestamp of the first sample in the buffer for more precise sync. Some...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_start), (gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
(gst_alsa_src_update_avail), (gst_alsa_src_loop):
Use alsa trigger_tstamp to get the timestamp of the first
sample in the buffer for more precise sync. Some cleanups.
2004-06-24 12:53:17 +00:00

224 lines
7.1 KiB
C

/*
* Copyright (C) 2001 CodeFactory AB
* Copyright (C) 2001 Thomas Nyberg <thomas@codefactory.se>
* Copyright (C) 2001-2002 Andy Wingo <apwingo@eos.ncsu.edu>
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free
* Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __GST_ALSA_H__
#define __GST_ALSA_H__
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include <alsa/control.h>
#include <alsa/error.h>
#include <gst/gst.h>
GST_DEBUG_CATEGORY_EXTERN (alsa_debug);
#define GST_CAT_DEFAULT alsa_debug
/* error checking for standard alsa functions */
/* NOTE: these functions require a GObject *this and can only be used in
functions that return TRUE on success and FALSE on error */
#define SIMPLE_ERROR_CHECK(value) G_STMT_START{ \
int err = (value); \
if (err < 0) { \
GST_WARNING_OBJECT (this, "\"" #value "\": %s", snd_strerror (err)); \
return FALSE; \
} \
}G_STMT_END
#ifdef G_HAVE_ISO_VARARGS
#define ERROR_CHECK(value, ...) G_STMT_START{ \
int err = (value); \
if (err < 0) { \
GST_WARNING_OBJECT (this, __VA_ARGS__, snd_strerror (err)); \
return FALSE; \
} \
}G_STMT_END
#elif defined(G_HAVE_GNUC_VARARGS)
#define ERROR_CHECK(value, args...) G_STMT_START{ \
int err = (value); \
if (err < 0) { \
GST_WARNING_OBJECT (this, ## args, snd_strerror (err)); \
return FALSE; \
} \
}G_STMT_END
#else
#define ERROR_CHECK(value, args...) G_STMT_START{ \
int err = (value); \
if (err < 0) { \
GST_WARNING_OBJECT (this, snd_strerror (err)); \
return FALSE; \
} \
}G_STMT_END
#endif
#define GST_ALSA_MIN_RATE 8000
#define GST_ALSA_MAX_RATE 192000
#define GST_ALSA_MAX_TRACKS 64 /* we don't support more than 64 tracks */
#define GST_ALSA_MAX_CHANNELS 32 /* tracks can have up to 32 channels */
/* Mono is 1 channel ; the 5.1 standard is 6 channels. The value for
GST_ALSA_MAX_CHANNELS comes from alsa/mixer.h. */
/* Max allowed discontinuity in time units between timestamp and playback
pointer before killing/inserting samples. This should be big enough to allow
smoothing errors on different video formats. */
#define GST_ALSA_DEFAULT_DISCONT (GST_SECOND / 10)
G_BEGIN_DECLS
#define GST_ALSA(obj) (G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_ALSA, GstAlsa))
#define GST_ALSA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_ALSA, GstAlsaClass))
#define GST_ALSA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_ALSA, GstAlsaClass))
#define GST_IS_ALSA(obj) (G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_ALSA))
#define GST_IS_ALSA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_ALSA))
#define GST_TYPE_ALSA (gst_alsa_get_type())
enum {
GST_ALSA_OPEN = GST_ELEMENT_FLAG_LAST,
GST_ALSA_RUNNING,
GST_ALSA_CAPS_NEGO,
GST_ALSA_FLAG_LAST = GST_ELEMENT_FLAG_LAST + 3
};
typedef enum {
GST_ALSA_CAPS_PAUSE = 0,
GST_ALSA_CAPS_RESUME,
GST_ALSA_CAPS_SYNC_START
/* add more */
} GstAlsaPcmCaps;
#define GST_ALSA_CAPS_IS_SET(obj, flag) (GST_ALSA (obj)->pcm_caps & (1<<(flag)))
#define GST_ALSA_CAPS_SET(obj, flag, set) G_STMT_START{ \
if (set) { (GST_ALSA (obj)->pcm_caps |= (1<<(flag))); } \
else { (GST_ALSA (obj)->pcm_caps &= ~(1<<(flag))); } \
}G_STMT_END
typedef struct _GstAlsaClock GstAlsaClock;
typedef struct _GstAlsaClockClass GstAlsaClockClass;
typedef struct _GstAlsa GstAlsa;
typedef struct _GstAlsaClass GstAlsaClass;
typedef int (*GstAlsaTransmitFunction) (GstAlsa *this, snd_pcm_sframes_t *avail);
typedef struct {
snd_pcm_format_t format;
guint rate;
gint channels;
} GstAlsaFormat;
struct _GstAlsa {
GstElement parent;
/* array of GstAlsaPads */
GstPad * pad[GST_ALSA_MAX_TRACKS];
gchar * device;
gchar * cardname;
snd_pcm_t * handle;
guint pcm_caps; /* capabilities of the pcm device, see GstAlsaPcmCaps */
snd_output_t * out;
GstAlsaFormat * format; /* NULL if undefined */
gboolean mmap; /* use mmap transmit (fast) or read/write (sloooow) */
GstAlsaTransmitFunction transmit;
/* latency / performance parameters */
snd_pcm_uframes_t period_size;
unsigned int period_count;
gboolean autorecover;
/* clocking */
GstAlsaClock * clock; /* our provided clock */
snd_pcm_uframes_t played; /* samples transmitted since last sync
This thing actually is our master clock.
We will event insert silent samples or
drop some to sync to incoming timestamps.
*/
snd_pcm_uframes_t captured;
GstClockTime max_discont; /* max difference between current
playback timestamp and buffers timestamps
*/
};
struct _GstAlsaClass {
GstElementClass parent_class;
snd_pcm_stream_t stream;
/* different transmit functions */
GstAlsaTransmitFunction transmit_mmap;
GstAlsaTransmitFunction transmit_rw;
/* autodetected devices available */
GList *devices;
};
GType gst_alsa_get_type (void);
void gst_alsa_set_eos (GstAlsa * this);
GstPadLinkReturn gst_alsa_link (GstPad * pad,
const GstCaps * caps);
GstCaps * gst_alsa_get_caps (GstPad * pad);
GstCaps * gst_alsa_fixate (GstPad * pad,
const GstCaps * caps);
GstCaps * gst_alsa_caps (snd_pcm_format_t format,
gint rate,
gint channels);
/* audio processing functions */
inline snd_pcm_sframes_t gst_alsa_update_avail (GstAlsa * this);
inline gboolean gst_alsa_pcm_wait (GstAlsa * this);
inline gboolean gst_alsa_start (GstAlsa * this);
gboolean gst_alsa_xrun_recovery (GstAlsa * this);
/* format conversions */
inline snd_pcm_uframes_t gst_alsa_timestamp_to_samples (GstAlsa * this,
GstClockTime time);
inline GstClockTime gst_alsa_samples_to_timestamp (GstAlsa * this,
snd_pcm_uframes_t samples);
inline snd_pcm_uframes_t gst_alsa_bytes_to_samples (GstAlsa * this,
guint bytes);
inline guint gst_alsa_samples_to_bytes (GstAlsa * this,
snd_pcm_uframes_t samples);
inline GstClockTime gst_alsa_bytes_to_timestamp (GstAlsa * this,
guint bytes);
inline guint gst_alsa_timestamp_to_bytes (GstAlsa * this,
GstClockTime time);
/* debugging functions (useful in gdb) - require running with --gst-debug=alsa:4 or better */
void gst_alsa_sw_params_dump (GstAlsa * this,
snd_pcm_sw_params_t * sw_params);
void gst_alsa_hw_params_dump (GstAlsa * this,
snd_pcm_hw_params_t * hw_params);
G_END_DECLS
#endif /* __GST_ALSA_H__ */