gstreamer/ext/alsa/gstalsasink.c
Andy Wingo 69d36f02ce gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2005-08-08  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.

* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.

* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.

* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.

* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.

* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
2005-08-08 16:42:10 +00:00

719 lines
18 KiB
C

/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
*
* gstalsasink.c:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <getopt.h>
#include <alsa/asoundlib.h>
#include "gstalsa.h"
#include "gstalsasink.h"
/* elementfactory information */
static GstElementDetails gst_alsasink_details =
GST_ELEMENT_DETAILS ("Audio Sink (ALSA)",
"Sink/Audio",
"Output to a sound card via ALSA",
"Wim Taymans <wim@fluendo.com>");
enum
{
PROP_0,
PROP_DEVICE,
};
static void gst_alsasink_base_init (gpointer g_class);
static void gst_alsasink_class_init (GstAlsaSinkClass * klass);
static void gst_alsasink_init (GstAlsaSink * alsasink);
static void gst_alsasink_dispose (GObject * object);
static void gst_alsasink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_alsasink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink);
static gboolean gst_alsasink_open (GstAudioSink * asink);
static gboolean gst_alsasink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
static gboolean gst_alsasink_close (GstAudioSink * asink);
static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_alsasink_delay (GstAudioSink * asink);
static void gst_alsasink_reset (GstAudioSink * asink);
/* AlsaSink signals and args */
enum
{
LAST_SIGNAL
};
static GstStaticPadTemplate alsasink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
#else
"endianness = (int) { BIG_ENDIAN, LITTLE_ENDIAN }, "
#endif
//"signed = (boolean) { TRUE, FALSE }, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
static GstElementClass *parent_class = NULL;
/* static guint gst_alsasink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_alsasink_get_type (void)
{
static GType alsasink_type = 0;
if (!alsasink_type) {
static const GTypeInfo alsasink_info = {
sizeof (GstAlsaSinkClass),
gst_alsasink_base_init,
NULL,
(GClassInitFunc) gst_alsasink_class_init,
NULL,
NULL,
sizeof (GstAlsaSink),
0,
(GInstanceInitFunc) gst_alsasink_init,
};
alsasink_type =
g_type_register_static (GST_TYPE_AUDIO_SINK, "GstAlsaSink",
&alsasink_info, 0);
}
return alsasink_type;
}
static void
gst_alsasink_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_alsasink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_alsasink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&alsasink_sink_factory));
}
static void
gst_alsasink_class_init (GstAlsaSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_AUDIO_SINK);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_alsasink_dispose);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"ALSA device, as defined in an asound configuration file",
"default", G_PARAM_READWRITE));
}
static void
gst_alsasink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAlsaSink *sink;
sink = GST_ALSA_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
if (sink->device)
g_free (sink->device);
sink->device = g_strdup (g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAlsaSink *sink;
sink = GST_ALSA_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, sink->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasink_init (GstAlsaSink * alsasink)
{
GST_DEBUG ("initializing alsasink");
alsasink->device = g_strdup ("default");
}
static GstCaps *
gst_alsasink_getcaps (GstBaseSink * bsink)
{
return NULL;
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) \
goto error; \
} G_STMT_END;
static int
set_hwparams (GstAlsaSink * alsa)
{
guint rrate;
gint err, dir;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca (&params);
GST_DEBUG ("Negotiating to %d channels @ %d Hz", alsa->channels, alsa->rate);
/* choose all parameters */
CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
/* set the interleaved read/write format */
CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
wrong_access);
/* set the sample format */
CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
no_sample_format);
/* set the count of channels */
CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
no_channels);
/* set the stream rate */
rrate = alsa->rate;
CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, 0),
no_rate);
if (rrate != alsa->rate)
goto rate_match;
if (alsa->buffer_time != -1) {
/* set the buffer time */
CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, &dir), buffer_time);
}
if (alsa->period_time != -1) {
/* set the period time */
CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&alsa->period_time, &dir), period_time);
}
/* write the parameters to device */
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
buffer_size);
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),
period_size);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Broken configuration for playback: no configurations available: %s",
snd_strerror (err)), (NULL));
return err;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Access type not available for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Sample format not available for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
no_channels:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Channels count (%i) not available for playbacks: %s",
alsa->channels, snd_strerror (err)), (NULL));
return err;
}
no_rate:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Rate %iHz not available for playback: %s",
alsa->rate, snd_strerror (err)), (NULL));
return err;
}
rate_match:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Rate doesn't match (requested %iHz, get %iHz)",
alsa->rate, err), (NULL));
return -EINVAL;
}
buffer_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set buffer time %i for playback: %s",
alsa->buffer_time, snd_strerror (err)), (NULL));
return err;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to get buffer size for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
period_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set period time %i for playback: %s", alsa->period_time,
snd_strerror (err)), (NULL));
return err;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to get period size for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set hw params for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
}
static int
set_swparams (GstAlsaSink * alsa)
{
int err;
snd_pcm_sw_params_t *params;
snd_pcm_sw_params_alloca (&params);
/* get the current swparams */
CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
(alsa->buffer_size / alsa->period_size) * alsa->period_size),
start_threshold);
/* allow the transfer when at least period_size samples can be processed */
CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
alsa->period_size), set_avail);
/* align all transfers to 1 sample */
CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
/* write the parameters to the playback device */
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to determine current swparams for playback: %s",
snd_strerror (err)), (NULL));
return err;
}
start_threshold:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set start threshold mode for playback: %s",
snd_strerror (err)), (NULL));
return err;
}
set_avail:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set avail min for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
set_align:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set transfer align for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
set_sw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Unable to set sw params for playback: %s", snd_strerror (err)),
(NULL));
return err;
}
}
static gboolean
alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
{
switch (spec->type) {
case GST_BUFTYPE_LINEAR:
alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
break;
case GST_BUFTYPE_FLOAT:
switch (spec->format) {
case GST_FLOAT32_LE:
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
break;
case GST_FLOAT32_BE:
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
break;
case GST_FLOAT64_LE:
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
break;
case GST_FLOAT64_BE:
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
break;
default:
goto error;
}
break;
case GST_BUFTYPE_A_LAW:
alsa->format = SND_PCM_FORMAT_A_LAW;
break;
case GST_BUFTYPE_MU_LAW:
alsa->format = SND_PCM_FORMAT_MU_LAW;
break;
default:
goto error;
}
alsa->rate = spec->rate;
alsa->channels = spec->channels;
alsa->buffer_time = spec->buffer_time;
alsa->period_time = spec->latency_time;
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
return TRUE;
/* ERRORS */
error:
{
return FALSE;
}
}
static gboolean
gst_alsasink_open (GstAudioSink * asink)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK), open_error);
return TRUE;
/* ERRORS */
open_error:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Playback open error: %s", snd_strerror (err)), (NULL));
return FALSE;
}
}
static gboolean
gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
if (!alsasink_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
alsa->bytes_per_sample = spec->bytes_per_sample;
spec->segsize = alsa->period_size * spec->bytes_per_sample;
spec->segtotal = alsa->buffer_size / alsa->period_size;
spec->silence_sample[0] = 0;
spec->silence_sample[1] = 0;
spec->silence_sample[2] = 0;
spec->silence_sample[3] = 0;
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Error parsing spec"), (NULL));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Could not set device to blocking: %s", snd_strerror (err)), (NULL));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Setting of hwparams failed: %s", snd_strerror (err)), (NULL));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Setting of swparams failed: %s", snd_strerror (err)), (NULL));
return FALSE;
}
}
static gboolean
gst_alsasink_unprepare (GstAudioSink * asink)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
CHECK (snd_pcm_drop (alsa->handle), drop);
CHECK (snd_pcm_hw_free (alsa->handle), hw_free);
CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);
return TRUE;
/* ERRORS */
drop:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Could not drop samples: %s", snd_strerror (err)), (NULL));
return FALSE;
}
hw_free:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Could not free hw params: %s", snd_strerror (err)), (NULL));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("Could not set device to nonblocking: %s", snd_strerror (err)),
(NULL));
return FALSE;
}
}
static gboolean
gst_alsasink_close (GstAudioSink * asink)
{
GstAlsaSink *alsa = GST_ALSA_SINK (asink);
snd_pcm_close (alsa->handle);
return TRUE;
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (snd_pcm_t * handle, gint err)
{
GST_DEBUG ("xrun recovery %d", err);
if (err == -EPIPE) { /* under-run */
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING ("Can't recovery from underrun, prepare failed: %s",
snd_strerror (err));
return 0;
} else if (err == -ESTRPIPE) {
while ((err = snd_pcm_resume (handle)) == -EAGAIN)
g_usleep (100); /* wait until the suspend flag is released */
if (err < 0) {
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING ("Can't recovery from suspend, prepare failed: %s",
snd_strerror (err));
}
return 0;
}
return err;
}
static guint
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstAlsaSink *alsa;
gint err;
gint cptr;
gint16 *ptr;
alsa = GST_ALSA_SINK (asink);
cptr = length / alsa->bytes_per_sample;
ptr = data;
while (cptr > 0) {
err = snd_pcm_writei (alsa->handle, ptr, cptr);
if (err < 0) {
if (err == -EAGAIN) {
GST_DEBUG ("Write error: %s", snd_strerror (err));
continue;
} else if (xrun_recovery (alsa->handle, err) < 0) {
goto write_error;
}
continue;
}
ptr += err * alsa->channels;
cptr -= err;
}
return length - cptr;
write_error:
{
return length; /* skip one period */
}
}
static guint
gst_alsasink_delay (GstAudioSink * asink)
{
GstAlsaSink *alsa;
snd_pcm_sframes_t delay;
alsa = GST_ALSA_SINK (asink);
snd_pcm_delay (alsa->handle, &delay);
return delay;
}
static void
gst_alsasink_reset (GstAudioSink * asink)
{
#if 0
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
CHECK (snd_pcm_drop (alsa->handle), drop_error);
CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
return;
/* ERRORS */
drop_error:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("alsa-reset: pcm drop error: %s", snd_strerror (err)), (NULL));
return;
}
prepare_error:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
("alsa-reset: pcm prepare error: %s", snd_strerror (err)), (NULL));
return;
}
#endif
}