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gstreamer/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c
Xavier Claessens b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h 
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-alsasink
* @title: alsasink
* @see_also: alsasrc
*
* This element renders audio samples using the ALSA audio API.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! autoaudiosink
* ]|
*
* Play an Ogg/Vorbis file and output audio via ALSA.
*
*/
#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 "gstalsaelements.h"
#include "gstalsa.h"
#include "gstalsasink.h"
#include <gst/audio/gstaudioiec61937.h>
#include <glib/gi18n-lib.h>
#ifndef ESTRPIPE
#define ESTRPIPE EPIPE
#endif
#define DEFAULT_DEVICE "default"
#define DEFAULT_DEVICE_NAME ""
#define DEFAULT_CARD_NAME ""
#define SPDIF_PERIOD_SIZE 1536
#define SPDIF_BUFFER_SIZE 15360
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_CARD_NAME,
PROP_LAST
};
#define gst_alsasink_parent_class parent_class
G_DEFINE_TYPE (GstAlsaSink, gst_alsasink, GST_TYPE_AUDIO_SINK);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsasink, "alsasink", GST_RANK_PRIMARY,
GST_TYPE_ALSA_SINK, alsa_element_init (plugin));
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, GstCaps * filter);
static gboolean gst_alsasink_query (GstBaseSink * bsink, GstQuery * query);
static gboolean gst_alsasink_open (GstAudioSink * asink);
static gboolean gst_alsasink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec);
static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
static gboolean gst_alsasink_close (GstAudioSink * asink);
static gint gst_alsasink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_alsasink_delay (GstAudioSink * asink);
static void gst_alsasink_pause (GstAudioSink * asink);
static void gst_alsasink_resume (GstAudioSink * asink);
static void gst_alsasink_stop (GstAudioSink * asink);
static gboolean gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps);
static GstBuffer *gst_alsasink_payload (GstAudioBaseSink * sink,
GstBuffer * buf);
static gint output_ref; /* 0 */
static snd_output_t *output; /* NULL */
static GMutex output_mutex;
static GstStaticPadTemplate alsasink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
PASSTHROUGH_CAPS)
);
static void
gst_alsasink_finalise (GObject * object)
{
GstAlsaSink *sink = GST_ALSA_SINK (object);
g_free (sink->device);
g_mutex_clear (&sink->alsa_lock);
g_mutex_clear (&sink->delay_lock);
g_mutex_lock (&output_mutex);
--output_ref;
if (output_ref == 0) {
snd_output_close (output);
output = NULL;
}
g_mutex_unlock (&output_mutex);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_alsasink_class_init (GstAlsaSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioBaseSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_alsasink_finalise;
gobject_class->get_property = gst_alsasink_get_property;
gobject_class->set_property = gst_alsasink_set_property;
gst_element_class_set_static_metadata (gstelement_class,
"Audio sink (ALSA)", "Sink/Audio",
"Output to a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_static_pad_template (gstelement_class,
&alsasink_sink_factory);
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_alsasink_query);
gstbaseaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_alsasink_payload);
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->stop = GST_DEBUG_FUNCPTR (gst_alsasink_stop);
gstaudiosink_class->pause = GST_DEBUG_FUNCPTR (gst_alsasink_pause);
gstaudiosink_class->resume = GST_DEBUG_FUNCPTR (gst_alsasink_resume);
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));
g_object_class_install_property (gobject_class, PROP_CARD_NAME,
g_param_spec_string ("card-name", "Card name",
"Human-readable name of the sound card", DEFAULT_CARD_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS |
GST_PARAM_DOC_SHOW_DEFAULT));
}
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;
case PROP_CARD_NAME:
g_value_take_string (value,
gst_alsa_find_card_name (GST_OBJECT_CAST (sink),
sink->device, SND_PCM_STREAM_PLAYBACK));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasink_init (GstAlsaSink * alsasink)
{
GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
alsasink->device = g_strdup (DEFAULT_DEVICE);
alsasink->handle = NULL;
alsasink->cached_caps = NULL;
alsasink->is_paused = FALSE;
alsasink->after_paused = FALSE;
alsasink->hw_support_pause = FALSE;
g_mutex_init (&alsasink->alsa_lock);
g_mutex_init (&alsasink->delay_lock);
g_mutex_lock (&output_mutex);
if (output_ref == 0) {
snd_output_stdio_attach (&output, stdout, 0);
++output_ref;
}
g_mutex_unlock (&output_mutex);
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) { \
GST_WARNING_OBJECT (alsa, "Error %d (%s) calling " #call, err, snd_strerror (err)); \
goto error; \
} \
} G_STMT_END;
static GstCaps *
gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSink *sink = GST_ALSA_SINK (bsink);
GstCaps *caps, *templ_caps;
GST_OBJECT_LOCK (sink);
if (sink->handle == NULL) {
GST_OBJECT_UNLOCK (sink);
GST_DEBUG_OBJECT (sink, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (sink->cached_caps) {
if (filter) {
caps = gst_caps_intersect_full (filter, sink->cached_caps,
GST_CAPS_INTERSECT_FIRST);
GST_OBJECT_UNLOCK (sink);
GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT " with "
"filter %" GST_PTR_FORMAT " applied: %" GST_PTR_FORMAT,
sink->cached_caps, filter, caps);
return caps;
} else {
caps = gst_caps_ref (sink->cached_caps);
GST_OBJECT_UNLOCK (sink);
GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT, caps);
return caps;
}
}
element_class = GST_ELEMENT_GET_CLASS (sink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
if (pad_template == NULL) {
GST_OBJECT_UNLOCK (sink);
g_assert_not_reached ();
return NULL;
}
templ_caps = gst_pad_template_get_caps (pad_template);
caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->device,
sink->handle, templ_caps);
gst_caps_unref (templ_caps);
if (caps) {
sink->cached_caps = gst_caps_ref (caps);
}
GST_OBJECT_UNLOCK (sink);
GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static gboolean
gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps)
{
GstPad *pad = GST_BASE_SINK (alsa)->sinkpad;
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
pad_caps = gst_pad_query_caps (pad, caps);
if (!pad_caps || gst_caps_is_empty (pad_caps)) {
if (pad_caps)
gst_caps_unref (pad_caps);
ret = FALSE;
goto done;
}
gst_caps_unref (pad_caps);
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps))
goto done;
/* parse helper expects this set, so avoid nasty warning
* will be set properly later on anyway */
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
goto done;
/* Make sure input is framed (one frame per buffer) and can be payloaded */
switch (spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
{
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
goto done;
}
default:{
}
}
ret = TRUE;
done:
gst_caps_replace (&spec.caps, NULL);
return ret;
}
static gboolean
gst_alsasink_query (GstBaseSink * sink, GstQuery * query)
{
GstAlsaSink *alsa = GST_ALSA_SINK (sink);
gboolean ret;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ACCEPT_CAPS:
{
GstCaps *caps;
gst_query_parse_accept_caps (query, &caps);
ret = gst_alsasink_acceptcaps (alsa, caps);
gst_query_set_accept_caps_result (query, ret);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
break;
}
return ret;
}
static int
set_hwparams (GstAlsaSink * alsa)
{
guint rrate;
gint err = 0;
snd_pcm_hw_params_t *params, *params_copy;
snd_pcm_hw_params_malloc (&params);
snd_pcm_hw_params_malloc (&params_copy);
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);
/* 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);
#ifndef GST_DISABLE_GST_DEBUG
/* get and dump some limits */
{
guint min, max;
snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_periods_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
}
#endif
/* Keep a copy of initial params struct that can be used later */
snd_pcm_hw_params_copy (params_copy, params);
if (!alsa->iec958) {
/* Following pulseaudio's approach in
* https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f
* we'll try various configuration to set the period time and buffer time as some
* driver can be picky on the order of the calls.
*/
if (alsa->buffer_time != -1 && alsa->period_time != -1) {
if (((err = snd_pcm_hw_params_set_period_time_near (alsa->handle,
params, &alsa->period_time, NULL)) >= 0)
&& ((err =
snd_pcm_hw_params_set_buffer_time_near (alsa->handle,
params, &alsa->buffer_time, NULL)) >= 0)) {
GST_DEBUG_OBJECT (alsa, "period time %u buffer time %u set correctly",
alsa->period_time, alsa->buffer_time);
goto success;
}
/* Try the new order with previous params struct as current one might
have partial settings from the order that was tried unsuccessfully */
snd_pcm_hw_params_copy (params, params_copy);
if (((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle,
params, &alsa->buffer_time, NULL)) >= 0)
&& ((err =
snd_pcm_hw_params_set_period_time_near (alsa->handle,
params, &alsa->period_time, NULL)) >= 0)) {
GST_DEBUG_OBJECT (alsa, "buffer time %u period time %u set correctly",
alsa->buffer_time, alsa->period_time);
goto success;
}
}
/* now try to configure the period time and buffer time exclusively
* if both fail we fall back to the defaults */
if (alsa->period_time != -1) {
snd_pcm_hw_params_copy (params, params_copy);
/* set the period time */
if ((err =
snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&alsa->period_time, NULL)) < 0) {
GST_DEBUG_OBJECT (alsa, "Unable to set period time %i for playback: %s",
alsa->period_time, snd_strerror (err));
} else {
GST_DEBUG_OBJECT (alsa, "period time %u set correctly",
alsa->period_time);
goto success;
}
}
if (alsa->buffer_time != -1) {
snd_pcm_hw_params_copy (params, params_copy);
/* set the buffer time */
if ((err =
snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, NULL)) < 0) {
GST_DEBUG_OBJECT (alsa, "Unable to set buffer time %i for playback: %s",
alsa->buffer_time, snd_strerror (err));
} else {
GST_DEBUG_OBJECT (alsa, "buffer time %u set correctly",
alsa->buffer_time);
goto success;
}
}
} else {
/* Set buffer size and period size manually for SPDIF */
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);
goto success;
}
/* Set nothing if all above failed */
snd_pcm_hw_params_copy (params, params_copy);
GST_DEBUG_OBJECT (alsa, "Not setting period time and buffer time");
success:
/* 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,
NULL), period_size);
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
alsa->period_size);
/* Check if hardware supports pause */
alsa->hw_support_pause = snd_pcm_hw_params_can_pause (params);
GST_DEBUG_OBJECT (alsa, "Hw support pause: %s",
alsa->hw_support_pause ? "yes" : "no");
goto exit;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for playback: no configurations available: %s",
snd_strerror (err)));
goto exit;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for playback: %s", snd_strerror (err)));
goto exit;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for playback: %s", snd_strerror (err)));
goto exit;
}
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, ("%s", msg),
("%s", snd_strerror (err)));
g_free (msg);
goto exit;
}
no_rate:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate %iHz not available for playback: %s",
alsa->rate, snd_strerror (err)));
goto exit;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for playback: %s", snd_strerror (err)));
goto exit;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for playback: %s", snd_strerror (err)));
goto exit;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for playback: %s", snd_strerror (err)));
}
exit:
{
snd_pcm_hw_params_free (params);
snd_pcm_hw_params_free (params_copy);
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, GstAudioRingBufferSpec * spec)
{
/* Initialize our boolean */
alsa->iec958 = FALSE;
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_U8:
alsa->format = SND_PCM_FORMAT_U8;
break;
case GST_AUDIO_FORMAT_S8:
alsa->format = SND_PCM_FORMAT_S8;
break;
case GST_AUDIO_FORMAT_S16LE:
alsa->format = SND_PCM_FORMAT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
alsa->format = SND_PCM_FORMAT_S16_BE;
break;
case GST_AUDIO_FORMAT_U16LE:
alsa->format = SND_PCM_FORMAT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
alsa->format = SND_PCM_FORMAT_U16_BE;
break;
case GST_AUDIO_FORMAT_S24_32LE:
alsa->format = SND_PCM_FORMAT_S24_LE;
break;
case GST_AUDIO_FORMAT_S24_32BE:
alsa->format = SND_PCM_FORMAT_S24_BE;
break;
case GST_AUDIO_FORMAT_U24_32LE:
alsa->format = SND_PCM_FORMAT_U24_LE;
break;
case GST_AUDIO_FORMAT_U24_32BE:
alsa->format = SND_PCM_FORMAT_U24_BE;
break;
case GST_AUDIO_FORMAT_S32LE:
alsa->format = SND_PCM_FORMAT_S32_LE;
break;
case GST_AUDIO_FORMAT_S32BE:
alsa->format = SND_PCM_FORMAT_S32_BE;
break;
case GST_AUDIO_FORMAT_U32LE:
alsa->format = SND_PCM_FORMAT_U32_LE;
break;
case GST_AUDIO_FORMAT_U32BE:
alsa->format = SND_PCM_FORMAT_U32_BE;
break;
case GST_AUDIO_FORMAT_S24LE:
alsa->format = SND_PCM_FORMAT_S24_3LE;
break;
case GST_AUDIO_FORMAT_S24BE:
alsa->format = SND_PCM_FORMAT_S24_3BE;
break;
case GST_AUDIO_FORMAT_U24LE:
alsa->format = SND_PCM_FORMAT_U24_3LE;
break;
case GST_AUDIO_FORMAT_U24BE:
alsa->format = SND_PCM_FORMAT_U24_3BE;
break;
case GST_AUDIO_FORMAT_S20LE:
alsa->format = SND_PCM_FORMAT_S20_3LE;
break;
case GST_AUDIO_FORMAT_S20BE:
alsa->format = SND_PCM_FORMAT_S20_3BE;
break;
case GST_AUDIO_FORMAT_U20LE:
alsa->format = SND_PCM_FORMAT_U20_3LE;
break;
case GST_AUDIO_FORMAT_U20BE:
alsa->format = SND_PCM_FORMAT_U20_3BE;
break;
case GST_AUDIO_FORMAT_S18LE:
alsa->format = SND_PCM_FORMAT_S18_3LE;
break;
case GST_AUDIO_FORMAT_S18BE:
alsa->format = SND_PCM_FORMAT_S18_3BE;
break;
case GST_AUDIO_FORMAT_U18LE:
alsa->format = SND_PCM_FORMAT_U18_3LE;
break;
case GST_AUDIO_FORMAT_U18BE:
alsa->format = SND_PCM_FORMAT_U18_3BE;
break;
case GST_AUDIO_FORMAT_F32LE:
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
break;
case GST_AUDIO_FORMAT_F32BE:
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
break;
case GST_AUDIO_FORMAT_F64LE:
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
break;
case GST_AUDIO_FORMAT_F64BE:
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
break;
default:
goto error;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
alsa->format = SND_PCM_FORMAT_A_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
alsa->format = SND_PCM_FORMAT_MU_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
alsa->format = SND_PCM_FORMAT_S16_BE;
alsa->iec958 = TRUE;
break;
default:
goto error;
}
alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
alsa->buffer_time = spec->buffer_time;
alsa->period_time = spec->latency_time;
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
(alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
return TRUE;
/* ERRORS */
error:
{
return FALSE;
}
}
static gboolean
gst_alsasink_open (GstAudioSink * asink)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
/* open in non-blocking mode, we'll use snd_pcm_wait() for space to become
* available. */
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, GstAudioRingBufferSpec * spec)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
if (alsa->iec958) {
snd_pcm_close (alsa->handle);
alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa), alsa->device);
if (G_UNLIKELY (!alsa->handle)) {
goto no_iec958;
}
}
if (!alsasink_parse_spec (alsa, spec))
goto spec_parse;
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
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);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SINK (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
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;
}
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;
alsa = GST_ALSA_SINK (asink);
snd_pcm_drop (alsa->handle);
snd_pcm_hw_free (alsa->handle);
return TRUE;
}
static gboolean
gst_alsasink_close (GstAudioSink * asink)
{
GstAlsaSink *alsa = GST_ALSA_SINK (asink);
GST_OBJECT_LOCK (asink);
if (alsa->handle) {
snd_pcm_close (alsa->handle);
alsa->handle = NULL;
}
gst_caps_replace (&alsa->cached_caps, NULL);
GST_OBJECT_UNLOCK (asink);
return TRUE;
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
{
GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
if (err == -EPIPE) { /* under-run */
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING_OBJECT (alsa,
"Can't recover from underrun, prepare failed: %s",
snd_strerror (err));
gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
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 recover from suspend, prepare failed: %s",
snd_strerror (err));
}
if (err == 0)
gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
return 0;
}
return err;
}
static gint
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstAlsaSink *alsa;
gint err;
gint cptr;
guint8 *ptr = data;
alsa = GST_ALSA_SINK (asink);
if (alsa->iec958 && alsa->need_swap) {
guint i;
guint16 *ptr_tmp = (guint16 *) ptr;
GST_DEBUG_OBJECT (asink, "swapping bytes");
for (i = 0; i < length / 2; i++) {
ptr_tmp[i] = GUINT16_SWAP_LE_BE (ptr_tmp[i]);
}
}
GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
cptr = length / alsa->bpf;
GST_ALSA_SINK_LOCK (asink);
while (cptr > 0) {
/* start by doing a blocking wait for free space. Set the timeout
* to 4 times the period time */
err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
if (err < 0) {
GST_DEBUG_OBJECT (asink, "wait error, %d", err);
} else {
GST_DELAY_SINK_LOCK (asink);
err = snd_pcm_writei (alsa->handle, ptr, cptr);
GST_DELAY_SINK_UNLOCK (asink);
}
if (err < 0) {
GST_DEBUG_OBJECT (asink, "Write error: %s (%d)", snd_strerror (err), err);
if (err == -EAGAIN) {
/* will continue out of the if/else group */
} else if (err == -ENODEV) {
goto device_disappeared;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto write_error;
}
/* Unlock so that _reset() can run and break an otherwise infinit loop
* here */
GST_ALSA_SINK_UNLOCK (asink);
g_thread_yield ();
GST_ALSA_SINK_LOCK (asink);
continue;
} else if (err == 0 && alsa->hw_support_pause) {
/* We might be already paused, if so, just bail */
if (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PAUSED)
break;
}
GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
cptr -= err;
}
GST_ALSA_SINK_UNLOCK (asink);
return length - (cptr * alsa->bpf);
write_error:
{
GST_ALSA_SINK_UNLOCK (asink);
return length; /* skip one period */
}
device_disappeared:
{
GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
(_("Error outputting to audio device. "
"The device has been disconnected.")), (NULL));
goto write_error;
}
}
static guint
gst_alsasink_delay (GstAudioSink * asink)
{
GstAlsaSink *alsa;
snd_pcm_sframes_t delay;
int res = 0;
alsa = GST_ALSA_SINK (asink);
GST_DELAY_SINK_LOCK (asink);
if (alsa->is_paused == TRUE) {
delay = alsa->pos_in_buffer;
alsa->is_paused = FALSE;
alsa->after_paused = TRUE;
} else {
if (alsa->after_paused == TRUE) {
delay = alsa->pos_in_buffer;
alsa->after_paused = FALSE;
} else {
res = snd_pcm_delay (alsa->handle, &delay);
}
}
GST_DELAY_SINK_UNLOCK (asink);
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_pause (GstAudioSink * asink)
{
GstAlsaSink *alsa;
gint err;
snd_pcm_sframes_t delay;
alsa = GST_ALSA_SINK (asink);
if (alsa->hw_support_pause == TRUE) {
GST_ALSA_SINK_LOCK (asink);
snd_pcm_delay (alsa->handle, &delay);
alsa->pos_in_buffer = delay;
CHECK (snd_pcm_pause (alsa->handle, 1), pause_error);
GST_DEBUG_OBJECT (alsa, "pause done");
alsa->is_paused = TRUE;
GST_ALSA_SINK_UNLOCK (asink);
} else {
gst_alsasink_stop (asink);
}
return;
pause_error:
{
GST_ERROR_OBJECT (alsa, "alsa-pause: pcm pause error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
}
static void
gst_alsasink_resume (GstAudioSink * asink)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
if (alsa->hw_support_pause == TRUE) {
GST_ALSA_SINK_LOCK (asink);
CHECK (snd_pcm_pause (alsa->handle, 0), resume_error);
GST_DEBUG_OBJECT (alsa, "resume done");
GST_ALSA_SINK_UNLOCK (asink);
}
return;
resume_error:
{
GST_ERROR_OBJECT (alsa, "alsa-resume: pcm resume error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
}
static void
gst_alsasink_stop (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, "stop done");
GST_ALSA_SINK_UNLOCK (asink);
return;
/* ERRORS */
drop_error:
{
GST_ERROR_OBJECT (alsa, "alsa-stop: pcm drop error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
prepare_error:
{
GST_ERROR_OBJECT (alsa, "alsa-stop: pcm prepare error: %s",
snd_strerror (err));
GST_ALSA_SINK_UNLOCK (asink);
return;
}
}
static GstBuffer *
gst_alsasink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
GstAlsaSink *alsa;
alsa = GST_ALSA_SINK (sink);
if (alsa->iec958) {
GstBuffer *out;
gint framesize;
GstMapInfo iinfo, oinfo;
framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
if (framesize <= 0)
return NULL;
out = gst_buffer_new_and_alloc (framesize);
gst_buffer_map (buf, &iinfo, GST_MAP_READ);
gst_buffer_map (out, &oinfo, GST_MAP_WRITE);
if (!gst_audio_iec61937_payload (iinfo.data, iinfo.size,
oinfo.data, oinfo.size, &sink->ringbuffer->spec, G_BIG_ENDIAN)) {
gst_buffer_unmap (buf, &iinfo);
gst_buffer_unmap (out, &oinfo);
gst_buffer_unref (out);
return NULL;
}
gst_buffer_unmap (buf, &iinfo);
gst_buffer_unmap (out, &oinfo);
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
return out;
}
return gst_buffer_ref (buf);
}
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