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gstreamer/subprojects/gst-plugins-base/ext/alsa/gstalsasink.c
Robert Rosengren 6bb1257c99 alsasink: Fix for being stuck in stop_streaming_threads state 
Moving from PLAYING to NULL will set the stop_streaming_threads to TRUE,
but when moving back upwards its not reset to FALSE (as only done in
uncalled init and resume callbacks).

Fix by reseting value in the prepare callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4309>
2023-03-31 16:51:11 +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;
alsasink->stop_streaming_threads = TRUE;
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 */
alsa->stop_streaming_threads = FALSE;
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) {
if (alsa->stop_streaming_threads)
goto write_error;
/* 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;
alsa->stop_streaming_threads = 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);
}
alsa->stop_streaming_threads = FALSE;
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");
alsa->stop_streaming_threads = TRUE;
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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