mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-10 03:19:40 +00:00
144b320afe
Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.c: (plugin_init): Set rank to NONE so that it doesn't get autoplugged by autoaudiosink (which didn't happen previously because the klass string didn't contain anything autoaudiosink was looking for).
840 lines
23 KiB
C
840 lines
23 KiB
C
/* Based on a plugin from Martin Soto's Seamless DVD Player.
|
|
* Copyright (C) 2003, 2004 Martin Soto <martinsoto@users.sourceforge.net>
|
|
* 2005-6 Michael Smith <msmith@fluendo.com>
|
|
*
|
|
* 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.
|
|
*/
|
|
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
#include <unistd.h>
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/gstaudioclock.h>
|
|
#include <gst/base/gstbasesink.h>
|
|
|
|
#include "alsaspdifsink.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (alsaspdifsink_debug);
|
|
#define GST_CAT_DEFAULT (alsaspdifsink_debug)
|
|
|
|
/* The magic audio-type we pretend to be for AC3 output */
|
|
#define AC3_CHANNELS 2
|
|
#define AC3_BITS 16
|
|
|
|
/* Define AC3 FORMAT as big endian. Fall back to swapping
|
|
* on sound devices that don't support it */
|
|
#define AC3_FORMAT_BE SND_PCM_FORMAT_S16_BE
|
|
#define AC3_FORMAT_LE SND_PCM_FORMAT_S16_LE
|
|
|
|
/* The size in bytes of an IEC958 frame. */
|
|
#define IEC958_FRAME_SIZE 6144
|
|
|
|
/* Size in bytes of an ALSA PCM frame (4, for this case). */
|
|
#define ALSASPDIFSINK_BYTES_PER_FRAME ((AC3_BITS / 8) * AC3_CHANNELS)
|
|
|
|
#if 0
|
|
/* The duration of a single IEC958 frame. */
|
|
#define IEC958_FRAME_DURATION (32 * GST_MSECOND)
|
|
|
|
/* Maximal synchronization difference. Measures will be taken if
|
|
block timestamps differ from actual playing time in more than this
|
|
value. */
|
|
#define MAX_SYNC_DIFF (IEC958_FRAME_DURATION * 0.8)
|
|
|
|
/* Playing time for the given number of ALSA PCM frames. */
|
|
#define ALSASPDIFSINK_TIME_PER_FRAMES(sink, frames) \
|
|
(((GstClockTime) (frames) * GST_SECOND) / AC3_RATE)
|
|
|
|
/* Number of ALSA PCM frames for the given playing time. */
|
|
#define ALSASPDIFSINK_FRAMES_PER_TIME(sink, time) \
|
|
(((GstClockTime) AC3_RATE * (time)) / GST_SECOND)
|
|
#endif
|
|
|
|
/* ElementFactory information. */
|
|
static GstElementDetails alsaspdifsink_details = {
|
|
"S/PDIF ALSA audiosink",
|
|
"Sink/Audio",
|
|
"Feeds audio to S/PDIF interfaces through the ALSA sound driver",
|
|
"Martin Soto <martinsoto@users.sourceforge.net>\n"
|
|
"Michael Smith <msmith@fluendo.com>"
|
|
};
|
|
|
|
/* AlsaSPDIFSink signals and args */
|
|
enum
|
|
{
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_CARD,
|
|
PROP_DEVICE
|
|
};
|
|
|
|
static GstStaticPadTemplate alsaspdifsink_sink_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-iec958")
|
|
);
|
|
|
|
#define _do_init(bla) \
|
|
GST_DEBUG_CATEGORY_INIT (alsaspdifsink_debug, "alsaspdifsink", 0, \
|
|
"ALSA S/PDIF audio sink element");
|
|
|
|
GST_BOILERPLATE_FULL (AlsaSPDIFSink, alsaspdifsink, GstBaseSink,
|
|
GST_TYPE_BASE_SINK, _do_init);
|
|
|
|
static void alsaspdifsink_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec);
|
|
static void alsaspdifsink_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec);
|
|
static gboolean alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event);
|
|
static GstFlowReturn alsaspdifsink_render (GstBaseSink * bsink,
|
|
GstBuffer * buf);
|
|
static void alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end);
|
|
static gboolean alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps);
|
|
|
|
static gboolean alsaspdifsink_open (AlsaSPDIFSink * sink);
|
|
static void alsaspdifsink_close (AlsaSPDIFSink * sink);
|
|
|
|
static GstClock *alsaspdifsink_provide_clock (GstElement * elem);
|
|
static GstClockTime alsaspdifsink_get_time (GstClock * clock,
|
|
gpointer user_data);
|
|
static void alsaspdifsink_dispose (GObject * object);
|
|
|
|
static GstStateChangeReturn alsaspdifsink_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
static int alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink);
|
|
|
|
/* Alsa error handler to suppress messages from within the ALSA library */
|
|
static void ignore_alsa_err (const char *file, int line, const char *function,
|
|
int err, const char *fmt, ...);
|
|
|
|
static void
|
|
alsaspdifsink_base_init (gpointer g_class)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_set_details (element_class, &alsaspdifsink_details);
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&alsaspdifsink_sink_factory));
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_class_init (AlsaSPDIFSinkClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstBaseSinkClass *gstbasesink_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstbasesink_class = (GstBaseSinkClass *) klass;
|
|
|
|
gobject_class->set_property = alsaspdifsink_set_property;
|
|
gobject_class->get_property = alsaspdifsink_get_property;
|
|
gobject_class->dispose = alsaspdifsink_dispose;
|
|
|
|
gstelement_class->change_state = alsaspdifsink_change_state;
|
|
gstelement_class->provide_clock = alsaspdifsink_provide_clock;
|
|
|
|
gstbasesink_class->event = alsaspdifsink_event;
|
|
gstbasesink_class->render = alsaspdifsink_render;
|
|
gstbasesink_class->get_times = alsaspdifsink_get_times;
|
|
gstbasesink_class->set_caps = alsaspdifsink_set_caps;
|
|
|
|
#if 0
|
|
/* We ignore the device property anyway, so don't install it
|
|
* we don't want the user supplying just any device string for us.
|
|
* At most we might want a card number and an iec958.%d device name
|
|
* to attempt */
|
|
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", G_PARAM_READWRITE));
|
|
#endif
|
|
g_object_class_install_property (gobject_class, PROP_CARD,
|
|
g_param_spec_int ("card", "Card",
|
|
"ALSA card number for the SPDIF device to use",
|
|
0, G_MAXINT, 0, G_PARAM_READWRITE));
|
|
|
|
snd_lib_error_set_handler (ignore_alsa_err);
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_init (AlsaSPDIFSink * sink, AlsaSPDIFSinkClass * g_class)
|
|
{
|
|
/* Create the provided clock. */
|
|
sink->clock = gst_audio_clock_new ("clock", alsaspdifsink_get_time, sink);
|
|
|
|
sink->card = 0;
|
|
sink->device = g_strdup ("default");
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_dispose (GObject * object)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
|
|
|
|
if (sink->clock)
|
|
gst_object_unref (sink->clock);
|
|
sink->clock = NULL;
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
AlsaSPDIFSink *sink;
|
|
|
|
sink = ALSASPDIFSINK (object);
|
|
|
|
switch (prop_id) {
|
|
/*
|
|
case PROP_DEVICE:
|
|
if(sink->device)
|
|
g_free(sink->device);
|
|
sink->device = g_strdup(g_value_get_string(value));
|
|
break;
|
|
*/
|
|
case PROP_CARD:
|
|
sink->card = g_value_get_int (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
static void
|
|
alsaspdifsink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
AlsaSPDIFSink *sink;
|
|
|
|
sink = ALSASPDIFSINK (object);
|
|
|
|
switch (prop_id) {
|
|
/*
|
|
case PROP_DEVICE:
|
|
g_value_set_string(value, sink->device);
|
|
break;
|
|
*/
|
|
case PROP_CARD:
|
|
g_value_set_int (value, sink->card);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
|
|
|
|
if (!gst_structure_get_int (gst_caps_get_structure (caps, 0), "rate",
|
|
&sink->rate))
|
|
sink->rate = 48000;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstClock *
|
|
alsaspdifsink_provide_clock (GstElement * elem)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (elem);
|
|
|
|
return GST_CLOCK (gst_object_ref (sink->clock));
|
|
}
|
|
|
|
static GstClockTime
|
|
alsaspdifsink_get_time (GstClock * clock, gpointer user_data)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (user_data);
|
|
|
|
return sink->frames * sink->rate / 1536;
|
|
}
|
|
|
|
static gboolean
|
|
alsaspdifsink_open (AlsaSPDIFSink * sink)
|
|
{
|
|
char *pcm_name = sink->device;
|
|
snd_pcm_hw_params_t *params;
|
|
snd_pcm_sw_params_t *sw_params;
|
|
unsigned int rate, buffer_time, period_time, tmp;
|
|
snd_pcm_uframes_t avail_min;
|
|
int err, step;
|
|
char devstr[256]; /* Storage for local 'default' device string */
|
|
GstClockTime time;
|
|
|
|
snd_pcm_hw_params_alloca (¶ms);
|
|
snd_pcm_sw_params_alloca (&sw_params);
|
|
|
|
/*
|
|
* Try and open our default iec958 device. Fall back to searching on card x
|
|
* if this fails, which should only happen on older alsa setups
|
|
*/
|
|
|
|
/* The string will be one of these:
|
|
* SPDIF_CON: Non-audio flag not set:
|
|
* spdif:{AES0 0x0 AES1 0x82 AES2 0x0 AES3 0x2}
|
|
* SPDIF_CON: Non-audio flag set:
|
|
* spdif:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}
|
|
*/
|
|
sprintf (devstr,
|
|
"iec958:{CARD %d AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
|
|
sink->card,
|
|
IEC958_AES0_NONAUDIO,
|
|
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
|
|
0, IEC958_AES3_CON_FS_48000);
|
|
|
|
GST_DEBUG_OBJECT (sink, "Generated device string \"%s\"", devstr);
|
|
pcm_name = devstr;
|
|
|
|
err = snd_pcm_open (&(sink->pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0);
|
|
if (err < 0) {
|
|
GST_DEBUG_OBJECT ("Open failed for %s - searching for IEC958 manually\n",
|
|
pcm_name);
|
|
|
|
err = alsaspdifsink_find_pcm_device (sink);
|
|
if (err == 0 && sink->pcm == NULL) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Could not open IEC958/SPDIF output device"), GST_ERROR_SYSTEM);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("snd_pcm_open: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
|
|
return FALSE;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_any (sink->pcm, params);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Broken configuration for this PCM: "
|
|
"no configurations available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
/* Set interleaved access. */
|
|
err = snd_pcm_hw_params_set_access (sink->pcm, params,
|
|
SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Access type not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_BE);
|
|
if (err < 0) {
|
|
/* Try LE output and swap data */
|
|
err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_LE);
|
|
sink->need_swap = TRUE;
|
|
} else
|
|
sink->need_swap = FALSE;
|
|
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Sample format not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_channels (sink->pcm, params, AC3_CHANNELS);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Channels count not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
rate = sink->rate;
|
|
err = snd_pcm_hw_params_set_rate_near (sink->pcm, params, &rate, 0);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Rate not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
buffer_time = 500000;
|
|
err = snd_pcm_hw_params_set_buffer_time_near (sink->pcm, params,
|
|
&buffer_time, 0);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Buffer time not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
time = buffer_time * 1000;
|
|
GST_DEBUG_OBJECT (sink, "buffer size set to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (time));
|
|
|
|
step = 2;
|
|
period_time = 10000 * 2;
|
|
do {
|
|
period_time /= 2;
|
|
tmp = period_time;
|
|
|
|
err = snd_pcm_hw_params_set_period_time_near (sink->pcm, params, &tmp, 0);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Period time not available"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
if (tmp == period_time) {
|
|
period_time /= 3;
|
|
tmp = period_time;
|
|
err = snd_pcm_hw_params_set_period_time_near (sink->pcm, params, &tmp, 0);
|
|
if (tmp == period_time) {
|
|
period_time = 10000 * 2;
|
|
}
|
|
}
|
|
} while (buffer_time == period_time && period_time > 10000);
|
|
|
|
if (buffer_time == period_time) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Buffer time and period time match, could not use"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
err = snd_pcm_hw_params (sink->pcm, params);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("PCM hw_params failed: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
err = snd_pcm_sw_params_current (sink->pcm, sw_params);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Cannot retrieve software params"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
|
|
avail_min = 48000 * 0.15;
|
|
err = snd_pcm_sw_params_set_avail_min (sink->pcm, sw_params, avail_min);
|
|
if (err < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("Cannot set avail min"), GST_ERROR_SYSTEM);
|
|
goto __close;
|
|
}
|
|
snd_pcm_sw_params_get_avail_min (sw_params, &avail_min);
|
|
GST_DEBUG_OBJECT (sink, "Avail min set to:%lu frames", avail_min);
|
|
|
|
return TRUE;
|
|
|
|
__close:
|
|
snd_pcm_close (sink->pcm);
|
|
sink->pcm = NULL;
|
|
return FALSE;
|
|
}
|
|
|
|
|
|
static void
|
|
alsaspdifsink_close (AlsaSPDIFSink * sink)
|
|
{
|
|
if (sink->pcm) {
|
|
snd_pcm_close (sink->pcm);
|
|
sink->pcm = NULL;
|
|
}
|
|
}
|
|
|
|
/* Try and find an IEC958 PCM device and mixer on card 0 and open it
|
|
* This function is only used on older ALSA installs that don't have the
|
|
* correct iec958 alias stuff set up, and relies on there being only
|
|
* one IEC958 PCM device (relies IEC958 in the device name) and one IEC958
|
|
* mixer control for doing the settings.
|
|
*/
|
|
static int
|
|
alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink)
|
|
{
|
|
int err = -1, dev, idx, count;
|
|
const gchar *ctl_name = "hw:0";
|
|
const gchar *spdif_name = SND_CTL_NAME_IEC958 ("", PLAYBACK, NONE);
|
|
int card = sink->card;
|
|
gchar pcm_name[24];
|
|
snd_pcm_t *pcm = NULL;
|
|
snd_ctl_t *ctl;
|
|
snd_ctl_card_info_t *info;
|
|
snd_ctl_elem_list_t *clist;
|
|
snd_ctl_elem_id_t *cid;
|
|
snd_pcm_info_t *pinfo;
|
|
|
|
GST_WARNING ("Opening IEC958 named device failed. Trying to autodetect");
|
|
|
|
snd_ctl_card_info_alloca (&info);
|
|
snd_pcm_info_alloca (&pinfo);
|
|
|
|
if ((err = snd_ctl_open (&ctl, ctl_name, card)) < 0)
|
|
return err;
|
|
|
|
/* Find a mixer for IEC958 settings */
|
|
snd_ctl_elem_list_alloca (&clist);
|
|
if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
|
|
goto beach;
|
|
|
|
if ((err =
|
|
snd_ctl_elem_list_alloc_space (clist,
|
|
snd_ctl_elem_list_get_count (clist))) < 0)
|
|
goto beach;
|
|
if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
|
|
goto beach;
|
|
|
|
count = snd_ctl_elem_list_get_used (clist);
|
|
for (idx = 0; idx < count; idx++) {
|
|
if (strstr (snd_ctl_elem_list_get_name (clist, idx), spdif_name) != NULL)
|
|
break;
|
|
}
|
|
if (idx == count) {
|
|
/* No SPDIF mixer availble */
|
|
err = 0;
|
|
goto beach;
|
|
}
|
|
snd_ctl_elem_id_alloca (&cid);
|
|
snd_ctl_elem_list_get_id (clist, idx, cid);
|
|
|
|
/* Now find a PCM device for IEC 958 */
|
|
if ((err = snd_ctl_card_info (ctl, info)) < 0)
|
|
goto beach;
|
|
dev = -1;
|
|
do {
|
|
if (snd_ctl_pcm_next_device (ctl, &dev) < 0)
|
|
goto beach;
|
|
if (dev < 0)
|
|
break; /* No more devices */
|
|
|
|
/* Filter for playback devices */
|
|
snd_pcm_info_set_device (pinfo, dev);
|
|
snd_pcm_info_set_subdevice (pinfo, 0);
|
|
snd_pcm_info_set_stream (pinfo, SND_PCM_STREAM_PLAYBACK);
|
|
if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0) {
|
|
if (err != -ENOENT)
|
|
goto beach; /* Genuine error */
|
|
|
|
/* Device has no playback streams */
|
|
continue;
|
|
}
|
|
if (strstr (snd_pcm_info_get_name (pinfo), "IEC958") == NULL)
|
|
continue; /* Not the device we are looking for */
|
|
|
|
count = snd_pcm_info_get_subdevices_count (pinfo);
|
|
GST_LOG_OBJECT (sink, "Device %d has %d subdevices\n", dev,
|
|
snd_pcm_info_get_subdevices_count (pinfo));
|
|
for (idx = 0; idx < count; idx++) {
|
|
snd_pcm_info_set_subdevice (pinfo, idx);
|
|
|
|
if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0)
|
|
goto beach;
|
|
|
|
g_assert (snd_pcm_info_get_stream (pinfo) == SND_PCM_STREAM_PLAYBACK);
|
|
|
|
GST_LOG_OBJECT (sink, "Found playback stream on dev %d sub-d %d\n", dev,
|
|
idx);
|
|
|
|
/* Found a suitable PCM device, let's open it */
|
|
g_snprintf (pcm_name, 24, "hw:%d,%d", card, dev);
|
|
if ((err =
|
|
snd_pcm_open (&(pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
|
|
goto beach;
|
|
|
|
break;
|
|
}
|
|
} while (pcm == NULL);
|
|
|
|
if (pcm != NULL) {
|
|
snd_ctl_elem_value_t *cval;
|
|
snd_aes_iec958_t iec958;
|
|
|
|
/* Have a PCM device and a mixer, set things up */
|
|
snd_ctl_elem_value_alloca (&cval);
|
|
snd_ctl_elem_value_set_id (cval, cid);
|
|
snd_ctl_elem_value_get_iec958 (cval, &iec958);
|
|
iec958.status[0] = IEC958_AES0_NONAUDIO;
|
|
iec958.status[1] = IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER;
|
|
iec958.status[2] = 0;
|
|
iec958.status[3] = IEC958_AES3_CON_FS_48000;
|
|
snd_ctl_elem_value_set_iec958 (cval, &iec958);
|
|
|
|
sink->pcm = pcm;
|
|
pcm = NULL;
|
|
err = 0;
|
|
}
|
|
|
|
beach:
|
|
if (pcm)
|
|
snd_pcm_close (pcm);
|
|
snd_ctl_close (ctl);
|
|
return err;
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_write_frame (AlsaSPDIFSink * sink, guchar * buf)
|
|
{
|
|
snd_pcm_sframes_t res;
|
|
int num_frames = IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME;
|
|
|
|
/* If we couldn't output big endian when we opened the devic, then
|
|
* we need to swap here */
|
|
if (sink->need_swap) {
|
|
int i;
|
|
guchar tmp;
|
|
|
|
for (i = 0; i < IEC958_FRAME_SIZE; i += 2) {
|
|
tmp = buf[i];
|
|
buf[i] = buf[i + 1];
|
|
buf[i + 1] = tmp;
|
|
}
|
|
}
|
|
|
|
res = 0;
|
|
do {
|
|
if (res == -EPIPE) {
|
|
/* Underrun. */
|
|
GST_INFO_OBJECT (sink, "buffer underrun");
|
|
res = snd_pcm_prepare (sink->pcm);
|
|
} else if (res == -ESTRPIPE) {
|
|
/* Suspend. */
|
|
while ((res = snd_pcm_resume (sink->pcm)) == -EAGAIN) {
|
|
GST_DEBUG_OBJECT (sink, "sleeping for suspend");
|
|
g_usleep (100000);
|
|
}
|
|
|
|
if (res < 0) {
|
|
res = snd_pcm_prepare (sink->pcm);
|
|
}
|
|
}
|
|
|
|
if (res >= 0) {
|
|
res = snd_pcm_writei (sink->pcm, (void *) buf, num_frames);
|
|
}
|
|
|
|
if (res > 0) {
|
|
num_frames -= res;
|
|
}
|
|
|
|
} while (res == -EPIPE || num_frames > 0);
|
|
|
|
sink->frames++;
|
|
|
|
if (res < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("writei returned error: %s", snd_strerror (res)), GST_ERROR_SYSTEM);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
snd_pcm_drop (sink->pcm);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
snd_pcm_start (sink->pcm);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
/* Like GstBaseAudioSink, we set these to NONE */
|
|
*start = GST_CLOCK_TIME_NONE;
|
|
*end = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
#if 0
|
|
static GstClockTime
|
|
alsaspdifsink_current_delay (AlsaSPDIFSink * sink)
|
|
{
|
|
snd_pcm_sframes_t delay;
|
|
int err;
|
|
|
|
err = snd_pcm_delay (sink->pcm, &delay);
|
|
if (err < 0 || delay < 0) {
|
|
return 0;
|
|
}
|
|
|
|
return ALSASPDIFSINK_TIME_PER_FRAMES (sink, delay);
|
|
}
|
|
|
|
static void
|
|
generate_iec958_zero_frame (guchar * buffer)
|
|
{
|
|
/* 2 sync words, 16 bits each */
|
|
buffer[0] = 0xF8;
|
|
buffer[1] = 0x72;
|
|
buffer[2] = 0x4E;
|
|
buffer[3] = 0x1F;
|
|
|
|
/* 16-bit burst-info. Contains data type (zero here, for 'null data'),
|
|
stream number (we output '0' for this always), and a few other bits.
|
|
As it happens, all-zero is the correct value.
|
|
*/
|
|
buffer[4] = 0;
|
|
buffer[5] = 0;
|
|
|
|
/* 16-bit frame size. Also zero */
|
|
buffer[6] = 0;
|
|
buffer[7] = 0;
|
|
|
|
memset (buffer + 8, 0, IEC958_FRAME_SIZE - 8);
|
|
}
|
|
#endif
|
|
|
|
static GstFlowReturn
|
|
alsaspdifsink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
|
|
|
|
#if 0
|
|
GstClockTime next_write;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
|
|
sink->cur_ts = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
next_write = gst_element_get_time (GST_ELEMENT (sink)) +
|
|
alsaspdifsink_current_delay (sink);
|
|
|
|
/*
|
|
fprintf (stderr, "Drift: % 0.6fs, delay: % 0.6fs\r",
|
|
GST_TIME_ARGS (GST_CLOCK_DIFF (sink->cur_ts, next_write)),
|
|
GST_TIME_ARGS (alsaspdifsink_current_delay (sink)));
|
|
*/
|
|
|
|
/* If we're too far behind, send empty IEC958 frames. */
|
|
if (sink->cur_ts > next_write + MAX_SYNC_DIFF) {
|
|
int frames = (int) (
|
|
((double) (sink->cur_ts - next_write)) /
|
|
(double) IEC958_FRAME_DURATION + 0.5);
|
|
int i;
|
|
|
|
for (i = 0; i < frames; i++) {
|
|
static guchar frame[IEC958_FRAME_SIZE];
|
|
|
|
generate_iec958_zero_frame (frame);
|
|
|
|
alsaspdifsink_write_frame (sink, frame);
|
|
}
|
|
}
|
|
/* If we're too far ahead, just drop this buffer */
|
|
else if (sink->cur_ts + MAX_SYNC_DIFF < next_write) {
|
|
goto end;
|
|
}
|
|
#endif
|
|
|
|
GST_LOG_OBJECT (sink, "Writing %d bytes to spdif out", GST_BUFFER_SIZE (buf));
|
|
if (GST_BUFFER_SIZE (buf) == IEC958_FRAME_SIZE)
|
|
alsaspdifsink_write_frame (sink, GST_BUFFER_DATA (buf));
|
|
else
|
|
GST_WARNING_OBJECT (sink, "Ignoring buffer of incorrect size");
|
|
|
|
#if 0
|
|
end:
|
|
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DURATION (buf)))
|
|
sink->cur_ts = GST_BUFFER_DURATION (buf);
|
|
#endif
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/* Drop error output from within alsalib on the floor */
|
|
static void
|
|
ignore_alsa_err (const char *file, int line, const char *function,
|
|
int err, const char *fmt, ...)
|
|
{
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
alsaspdifsink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (element);
|
|
GstStateChangeReturn ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
sink->frames = 0;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
if (!alsaspdifsink_open (sink)) {
|
|
GST_WARNING_OBJECT (sink, "Failed to open alsa device");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
GST_INFO_OBJECT (sink, "Parent change_state returned %d", ret);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
alsaspdifsink_close (sink);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
/* no rank so it doesn't get autoplugged by autoaudiosink */
|
|
if (!gst_element_register (plugin, "alsaspdifsink", GST_RANK_NONE,
|
|
GST_TYPE_ALSASPDIFSINK)) {
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"alsaspdif",
|
|
"Alsa plugin for S/PDIF output",
|
|
plugin_init,
|
|
VERSION, GST_LICENSE_UNKNOWN, PACKAGE, "http://www.fluendo.com");
|