gstreamer/ext/alsa/gstalsasrc.c

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/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
*
* gstalsasrc.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.
*/
/**
* SECTION:element-alsasrc
* @short_description: capture audio from an alsa device
* @see_also: alsasink, alsamixer
*
* <refsect2>
* <para>
* This element reads data from an audio card using the ALSA API.
* </para>
* <title>Example pipelines</title>
* <para>
* Record from a sound card using ALSA and encode to Ogg/Vorbis.
* </para>
* <programlisting>
* gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
* </programlisting>
* </refsect2>
*
* Last reviewed on 2006-03-01 (0.10.4)
*/
#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 "gstalsasrc.h"
#include "gstalsadeviceprobe.h"
#include <gst/gst-i18n-plugin.h>
/* elementfactory information */
make GstElementDetails const Original commit message from CVS: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsasrc.c: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/ogg/gstogmparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst/audioconvert/gstaudioconvert.c: * gst/audiorate/gstaudiorate.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: * gst/typefind/gsttypefindfunctions.c: (plugin_init): * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/v4l/gstv4ljpegsrc.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * sys/v4l/gstv4lsrc.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: * tests/check/libs/cddabasesrc.c: make GstElementDetails const
2006-04-28 19:46:37 +00:00
static const GstElementDetails gst_alsasrc_details =
GST_ELEMENT_DETAILS ("Audio source (ALSA)",
"Source/Audio",
"Read from a sound card via ALSA",
"Wim Taymans <wim@fluendo.com>");
#define DEFAULT_PROP_DEVICE "default"
#define DEFAULT_PROP_DEVICE_NAME ""
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
};
static void gst_alsasrc_init_interfaces (GType type);
GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,
GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);
GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
static void gst_alsasrc_dispose (GObject * object);
static void gst_alsasrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_alsasrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc);
static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
static guint gst_alsasrc_delay (GstAudioSrc * asrc);
static void gst_alsasrc_reset (GstAudioSrc * asrc);
/* AlsaSrc signals and args */
enum
{
LAST_SIGNAL
};
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static GstStaticPadTemplate alsasrc_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 32, "
"depth = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
"audio/x-raw-int, "
"endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static void
gst_alsasrc_dispose (GObject * object)
{
GstAlsaSrc *src = GST_ALSA_SRC (object);
g_free (src->device);
src->device = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static gboolean
gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
{
/* only support this one interface (wrapped by GstImplementsInterface) */
g_assert (interface_type == GST_TYPE_MIXER);
return gst_alsasrc_mixer_supported (this, interface_type);
}
static void
gst_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = (gpointer) gst_alsasrc_interface_supported;
}
static void
gst_alsasrc_init_interfaces (GType type)
{
static const GInterfaceInfo implements_iface_info = {
(GInterfaceInitFunc) gst_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo mixer_iface_info = {
(GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
NULL,
NULL,
};
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_iface_info);
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
gst_alsa_type_add_device_property_probe_interface (type);
}
static void
gst_alsasrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_alsasrc_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&alsasrc_src_factory));
}
static void
gst_alsasrc_class_init (GstAlsaSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_alsasrc_dispose);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasrc_get_property);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasrc_set_property);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_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_PROP_DEVICE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device",
DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE));
}
static void
gst_alsasrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (src->device);
src->device = g_value_dup_string (value);
if (src->device == NULL) {
src->device = g_strdup (DEFAULT_PROP_DEVICE);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
if (src->handle) {
snd_pcm_info_t *info;
snd_pcm_info_malloc (&info);
snd_pcm_info (src->handle, info);
g_value_set_string (value, snd_pcm_info_get_name (info));
snd_pcm_info_free (info);
} else {
g_value_set_string (value, NULL);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
{
GST_DEBUG_OBJECT (alsasrc, "initializing");
alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
alsasrc->cached_caps = NULL;
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) \
goto error; \
} G_STMT_END;
static GstCaps *
gst_alsasrc_getcaps (GstBaseSrc * bsrc)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSrc *src;
GstCaps *caps;
src = GST_ALSA_SRC (bsrc);
if (src->handle == NULL) {
GST_DEBUG_OBJECT (src, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (src->cached_caps) {
GST_LOG_OBJECT (src, "Returning cached caps");
return gst_caps_ref (src->cached_caps);
}
element_class = GST_ELEMENT_GET_CLASS (src);
pad_template = gst_element_class_get_pad_template (element_class, "src");
g_return_val_if_fail (pad_template != NULL, NULL);
caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
gst_pad_template_get_caps (pad_template));
if (caps) {
src->cached_caps = gst_caps_ref (caps);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static int
set_hwparams (GstAlsaSrc * alsa)
{
guint rrate;
gint err, dir;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca (&params);
/* 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, SETTINGS, (NULL),
("Broken configuration for recording: no configurations available: %s",
snd_strerror (err)));
return err;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for recording: %s", snd_strerror (err)));
return err;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for recording: %s", snd_strerror (err)));
return err;
}
no_channels:
{
gchar *msg = NULL;
if ((alsa->channels) == 1)
msg = g_strdup (_("Could not open device for recording in mono mode."));
if ((alsa->channels) == 2)
msg = g_strdup (_("Could not open device for recording in stereo mode."));
if ((alsa->channels) > 2)
msg =
g_strdup_printf (_
("Could not open device for recording in %d-channel mode"),
alsa->channels);
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));
g_free (msg);
return err;
}
no_rate:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate %iHz not available for recording: %s",
alsa->rate, snd_strerror (err)));
return err;
}
rate_match:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
return -EINVAL;
}
buffer_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set buffer time %i for recording: %s",
alsa->buffer_time, snd_strerror (err)));
return err;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for recording: %s", snd_strerror (err)));
return err;
}
period_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set period time %i for recording: %s", alsa->period_time,
snd_strerror (err)));
return err;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for recording: %s", snd_strerror (err)));
return err;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for recording: %s", snd_strerror (err)));
return err;
}
}
static int
set_swparams (GstAlsaSrc * 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 */
#if 0
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);
#endif
/* 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 recording device */
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to determine current swparams for playback: %s",
snd_strerror (err)));
return err;
}
#if 0
start_threshold:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set start threshold mode for playback: %s",
snd_strerror (err)));
return err;
}
set_avail:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set avail min for playback: %s", snd_strerror (err)));
return err;
}
#endif
set_align:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set transfer align for playback: %s", snd_strerror (err)));
return err;
}
set_sw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set sw params for playback: %s", snd_strerror (err)));
return err;
}
}
static gboolean
alsasrc_parse_spec (GstAlsaSrc * 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_alsasrc_open (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
SND_PCM_NONBLOCK), open_error);
if (!alsa->mixer)
alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
return TRUE;
/* ERRORS */
open_error:
{
if (err == -EBUSY) {
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, (NULL), ("Device '%s' is busy",
alsa->device));
} else {
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
(NULL), ("Recording open error on device '%s': %s", alsa->device,
snd_strerror (err)));
}
return FALSE;
}
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_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);
CHECK (snd_pcm_prepare (alsa->handle), prepare_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, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_alsasrc_unprepare (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
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, SETTINGS, (NULL),
("Could not drop samples: %s", snd_strerror (err)));
return FALSE;
}
hw_free:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not free hw params: %s", snd_strerror (err)));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to nonblocking: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_alsasrc_close (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
snd_pcm_close (alsa->handle);
if (alsa->mixer) {
gst_alsa_mixer_free (alsa->mixer);
alsa->mixer = NULL;
}
gst_caps_replace (&alsa->cached_caps, NULL);
return TRUE;
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
{
GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
if (err == -EPIPE) { /* under-run */
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING_OBJECT (alsa,
"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_OBJECT (alsa,
"Can't recovery from suspend, prepare failed: %s",
snd_strerror (err));
}
return 0;
}
return err;
}
static guint
gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
GstAlsaSrc *alsa;
gint err;
gint cptr;
gint16 *ptr;
alsa = GST_ALSA_SRC (asrc);
cptr = length / alsa->bytes_per_sample;
ptr = data;
while (cptr > 0) {
if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
if (err == -EAGAIN) {
GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
continue;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto read_error;
}
continue;
}
ptr += err * alsa->channels;
cptr -= err;
}
return length - cptr;
read_error:
{
return length; /* skip one period */
}
}
static guint
gst_alsasrc_delay (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
snd_pcm_sframes_t delay;
alsa = GST_ALSA_SRC (asrc);
snd_pcm_delay (alsa->handle, &delay);
return CLAMP (delay, 0, alsa->buffer_size);
}
static void
gst_alsasrc_reset (GstAudioSrc * asrc)
{
#if 0
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
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
}