gstreamer/ext/alsa/gstalsasink.c

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/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
*
* 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.
*/
/**
* SECTION:element-alsasink
* @short_description: play audio to an ALSA device
* @see_also: alsasrc, alsamixer
*
* <refsect2>
* <para>
* This element renders raw audio samples using the ALSA api.
* </para>
* <title>Example pipelines</title>
* <para>
* Play an Ogg/Vorbis file.
* </para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink
* </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 "gstalsa.h"
#include "gstalsasink.h"
#include "gstalsadeviceprobe.h"
#include <gst/gst-i18n-plugin.h>
/* elementfactory information */
static const GstElementDetails gst_alsasink_details =
GST_ELEMENT_DETAILS ("Audio sink (ALSA)",
"Sink/Audio",
"Output to a sound card via ALSA",
"Wim Taymans <wim@fluendo.com>");
#define DEFAULT_DEVICE "default"
#define DEFAULT_DEVICE_NAME ""
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME
};
static void gst_alsasink_init_interfaces (GType type);
GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,
GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);
static void gst_alsasink_finalise (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);
static gint output_ref; /* 0 */
static snd_output_t *output; /* NULL */
static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static GstStaticPadTemplate alsasink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " ALSA_SINK_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_SINK_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_alsasink_finalise (GObject * object)
{
GstAlsaSink *sink = GST_ALSA_SINK (object);
g_free (sink->device);
g_mutex_free (sink->alsa_lock);
g_static_mutex_lock (&output_mutex);
--output_ref;
if (output_ref == 0) {
snd_output_close (output);
output = NULL;
}
g_static_mutex_unlock (&output_mutex);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_alsasink_init_interfaces (GType type)
{
gst_alsa_type_add_device_property_probe_interface (type);
}
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_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);
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_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_DEVICE_NAME,
G_PARAM_READABLE));
}
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:
g_free (sink->device);
sink->device = g_value_dup_string (value);
/* setting NULL restores the default device */
if (sink->device == NULL) {
sink->device = g_strdup (DEFAULT_DEVICE);
}
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;
case PROP_DEVICE_NAME:
g_value_take_string (value,
gst_alsa_find_device_name (GST_OBJECT_CAST (sink),
sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class)
{
GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
alsasink->device = g_strdup (DEFAULT_DEVICE);
alsasink->handle = NULL;
alsasink->cached_caps = NULL;
alsasink->alsa_lock = g_mutex_new ();
g_static_mutex_lock (&output_mutex);
if (output_ref == 0) {
snd_output_stdio_attach (&output, stdout, 0);
++output_ref;
}
g_static_mutex_unlock (&output_mutex);
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) \
goto error; \
} G_STMT_END;
static GstCaps *
gst_alsasink_getcaps (GstBaseSink * bsink)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSink *sink = GST_ALSA_SINK (bsink);
GstCaps *caps;
if (sink->handle == NULL) {
GST_DEBUG_OBJECT (sink, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (sink->cached_caps) {
GST_LOG_OBJECT (sink, "Returning cached caps");
return gst_caps_ref (sink->cached_caps);
}
element_class = GST_ELEMENT_GET_CLASS (sink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
g_return_val_if_fail (pad_template != NULL, NULL);
caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,
gst_pad_template_get_caps (pad_template));
if (caps) {
sink->cached_caps = gst_caps_ref (caps);
}
GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static int
set_hwparams (GstAlsaSink * alsa)
{
guint rrate;
gint err, dir;
snd_pcm_hw_params_t *params;
guint period_time, buffer_time;
snd_pcm_hw_params_alloca (&params);
GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s)",
alsa->channels, alsa->rate, snd_pcm_format_name (alsa->format));
/* start with requested values, if we cannot configure alsa for those values,
* we set these values to -1, which will leave the default alsa values */
buffer_time = alsa->buffer_time;
period_time = alsa->period_time;
retry:
/* 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;
/* get and dump some limits */
{
guint min, max;
snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir);
snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir);
GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
alsa->buffer_time, min, max);
snd_pcm_hw_params_get_period_time_min (params, &min, &dir);
snd_pcm_hw_params_get_period_time_max (params, &max, &dir);
GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
alsa->period_time, min, max);
snd_pcm_hw_params_get_periods_min (params, &min, &dir);
snd_pcm_hw_params_get_periods_max (params, &max, &dir);
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
}
/* now try to configure the buffer time and period time, if one
* of those fail, we fall back to the defaults and emit a warning. */
if (buffer_time != -1) {
/* set the buffer time */
if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&buffer_time, &dir)) < 0) {
GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set buffer time %i for playback: %s",
buffer_time, snd_strerror (err)));
/* disable buffer_time the next round */
buffer_time = -1;
goto retry;
}
}
if (period_time != -1) {
/* set the period time */
if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&period_time, &dir)) < 0) {
GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set period time %i for playback: %s",
period_time, snd_strerror (err)));
/* disable period_time the next round */
period_time = -1;
goto retry;
}
}
/* write the parameters to device */
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
/* now get the configured values */
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);
Printf format fixes. Original commit message from CVS: * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_device_property_probe_get_values): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad), (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain): * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad): * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream), (gst_ogg_parse_chain): * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header): * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup), (gst_vorbis_enc_buffer_check_discontinuous): * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_handle_track_seek): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_push_full): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push): * gst/audioresample/resample.c: (resample_input_pushthrough): * gst/playback/gstplaybasebin.c: (queue_out_of_data): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients): * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset), (wavpack_type_find): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create): * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy), (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/volume.c: (GST_START_TEST): Printf format fixes.
2006-10-05 15:55:21 +00:00
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
alsa->period_size);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for playback: no configurations available: %s",
snd_strerror (err)));
return err;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for playback: %s", snd_strerror (err)));
return err;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for playback: %s", snd_strerror (err)));
return err;
}
no_channels:
{
gchar *msg = NULL;
if ((alsa->channels) == 1)
msg = g_strdup (_("Could not open device for playback in mono mode."));
if ((alsa->channels) == 2)
msg = g_strdup (_("Could not open device for playback in stereo mode."));
if ((alsa->channels) > 2)
msg =
g_strdup_printf (_
("Could not open device for playback 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 playback: %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_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for playback: %s", snd_strerror (err)));
return err;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for playback: %s", snd_strerror (err)));
return err;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for playback: %s", snd_strerror (err)));
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, SETTINGS, (NULL),
("Unable to determine current swparams for playback: %s",
snd_strerror (err)));
return err;
}
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;
}
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
alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
{
switch (spec->type) {
case GST_BUFTYPE_LINEAR:
GST_DEBUG_OBJECT (alsa,
"Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,
spec->width, spec->sign, spec->bigend);
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);
GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
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), ("Playback open error on device '%s': %s", alsa->device,
snd_strerror (err)));
}
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;
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;
}
}
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, 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_alsasink_close (GstAudioSink * asink)
{
GstAlsaSink *alsa = GST_ALSA_SINK (asink);
gint err;
CHECK (snd_pcm_close (alsa->handle), close_error);
alsa->handle = NULL;
gst_caps_replace (&alsa->cached_caps, NULL);
return TRUE;
/* ERRORS */
close_error:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL),
("Playback close error: %s", snd_strerror (err)));
return FALSE;
}
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (GstAlsaSink * 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_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;
GST_ALSA_SINK_LOCK (asink);
while (cptr > 0) {
err = snd_pcm_writei (alsa->handle, ptr, cptr);
GST_DEBUG_OBJECT (asink, "written %d result %d", cptr, err);
if (err < 0) {
GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
if (err == -EAGAIN) {
continue;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto write_error;
}
continue;
}
ptr += err * alsa->channels;
cptr -= err;
}
GST_ALSA_SINK_UNLOCK (asink);
return length - cptr;
write_error:
{
GST_ALSA_SINK_UNLOCK (asink);
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)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
GST_ALSA_SINK_LOCK (asink);
GST_DEBUG_OBJECT (alsa, "drop");
CHECK (snd_pcm_drop (alsa->handle), drop_error);
GST_DEBUG_OBJECT (alsa, "prepare");
CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
GST_DEBUG_OBJECT (alsa, "reset done");
GST_ALSA_SINK_UNLOCK (asink);
return;
/* ERRORS */
drop_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
prepare_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
}