gstreamer/ext/alsa/gstalsasink.c
Sebastian Dröge 49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00

983 lines
28 KiB
C

/* 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 ""
#define SPDIF_PERIOD_SIZE 1536
#define SPDIF_BUFFER_SIZE 15360
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) 24, "
"depth = (int) 24, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
"audio/x-raw-int, "
"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 32, "
"depth = (int) 24, "
"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 ];"
"audio/x-iec958")
);
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_PARAM_STATIC_STRINGS));
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 | G_PARAM_STATIC_STRINGS));
}
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_malloc (&params);
GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
"SPDIF (%d)", alsa->channels, alsa->rate,
snd_pcm_format_name (alsa->format), alsa->iec958);
/* 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 */
if (alsa->iec958) {
/* Try to use big endian first else fallback to le and swap bytes */
if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
alsa->format = SND_PCM_FORMAT_S16_LE;
alsa->need_swap = TRUE;
GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
} else {
alsa->need_swap = FALSE;
}
}
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, NULL),
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 && !alsa->iec958) {
/* 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;
}
GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
}
if (period_time != -1 && !alsa->iec958) {
/* 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;
}
GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
}
/* Set buffer size and period size manually for SPDIF */
if (G_UNLIKELY (alsa->iec958)) {
snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
&buffer_size), buffer_size);
CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
&period_size, NULL), period_size);
}
/* 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);
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
alsa->period_size);
snd_pcm_hw_params_free (params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for playback: no configurations available: %s",
snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
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);
snd_pcm_hw_params_free (params);
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));
snd_pcm_hw_params_free (params);
return -EINVAL;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
}
static int
set_swparams (GstAlsaSink * alsa)
{
int err;
snd_pcm_sw_params_t *params;
snd_pcm_sw_params_malloc (&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);
#if GST_CHECK_ALSA_VERSION(1,0,16)
/* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
#else
/* align all transfers to 1 sample */
CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
#endif
/* write the parameters to the playback device */
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
snd_pcm_sw_params_free (params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to determine current swparams for playback: %s",
snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
start_threshold:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set start threshold mode for playback: %s",
snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
set_avail:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set avail min for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
#if !GST_CHECK_ALSA_VERSION(1,0,16)
set_align:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set transfer align for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
#endif
set_sw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set sw params for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
}
static gboolean
alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
{
/* Initialize our boolean */
alsa->iec958 = FALSE;
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;
case GST_BUFTYPE_IEC958:
alsa->format = SND_PCM_FORMAT_S16_BE;
alsa->iec958 = TRUE;
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,
(_("Could not open audio device for playback. "
"Device is being used by another application.")),
("Device '%s' is busy", alsa->device));
} else {
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
(_("Could not open audio device for playback.")),
("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 (spec->format == GST_IEC958) {
snd_pcm_close (alsa->handle);
alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa));
if (G_UNLIKELY (!alsa->handle)) {
goto no_iec958;
}
}
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;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
return TRUE;
/* ERRORS */
no_iec958:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
("Could not open IEC958 (SPDIF) device for playback"));
return FALSE;
}
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;
if (alsa->handle) {
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 = data;
alsa = GST_ALSA_SINK (asink);
if (alsa->iec958 && alsa->need_swap) {
guint i;
GST_DEBUG_OBJECT (asink, "swapping bytes");
for (i = 0; i < length / 2; i++) {
ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);
}
}
GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
cptr = length / alsa->bytes_per_sample;
GST_ALSA_SINK_LOCK (asink);
while (cptr > 0) {
err = snd_pcm_writei (alsa->handle, ptr, cptr);
GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
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 += snd_pcm_frames_to_bytes (alsa->handle, err);
cptr -= err;
}
GST_ALSA_SINK_UNLOCK (asink);
return length - (cptr * alsa->bytes_per_sample);
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;
int res;
alsa = GST_ALSA_SINK (asink);
res = snd_pcm_delay (alsa->handle, &delay);
if (G_UNLIKELY (res < 0)) {
/* on errors, report 0 delay */
GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
delay = 0;
}
if (G_UNLIKELY (delay < 0)) {
/* make sure we never return a negative delay */
GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
delay = 0;
}
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;
}
}