+++ /dev/null
-/* 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)
-
-#define IEC958_SAMPLES_PER_FRAME (IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME)
-
-#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
-
-/* 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 void alsaspdifsink_finalize (GObject * object);
-
-static GstStateChangeReturn alsaspdifsink_change_state (GstElement * element,
- GstStateChange transition);
-static int alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink);
-static gboolean alsaspdifsink_set_params (AlsaSPDIFSink * sink);
-static snd_pcm_sframes_t alsaspdifsink_delay (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_simple (element_class, "S/PDIF ALSA audiosink",
- "Sink/Audio",
- "Feeds audio to S/PDIF interfaces through the ALSA sound driver",
- "Martin Soto <martinsoto@users.sourceforge.net>, "
- "Michael Smith <msmith@fluendo.com>");
- 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;
- gobject_class->finalize = alsaspdifsink_finalize;
-
- 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. */
-#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
- sink->clock =
- gst_audio_clock_new_full ("clock", alsaspdifsink_get_time,
- gst_object_ref (sink), (GDestroyNotify) gst_object_unref);
-#else
- sink->clock = gst_audio_clock_new ("clock", alsaspdifsink_get_time, sink);
-#endif
-
- 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_finalize (GObject * object)
-{
- AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
-
- g_free (sink->device);
- sink->device = NULL;
-
- G_OBJECT_CLASS (parent_class)->finalize (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;
-
- if (!alsaspdifsink_set_params (sink)) {
- GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
- ("Cannot set ALSA hardware parameters"), GST_ERROR_SYSTEM);
- return FALSE;
- }
-
- 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)
-{
- GstClockTime result;
- snd_pcm_sframes_t raw, delay, samples;
- AlsaSPDIFSink *sink = ALSASPDIFSINK (user_data);
-
- raw = samples = sink->frames * IEC958_SAMPLES_PER_FRAME;
- delay = alsaspdifsink_delay (sink);
-
- if (samples > delay)
- samples -= delay;
- else
- samples = 0;
-
- result = gst_util_uint64_scale_int (samples, GST_SECOND, sink->rate);
- GST_LOG_OBJECT (sink, "Samples raw: %d, delay: %d, real: %d, "
- "Time: %" GST_TIME_FORMAT, (int) raw, (int) delay, (int) samples,
- GST_TIME_ARGS (result));
- return result;
-}
-
-static gboolean
-alsaspdifsink_open (AlsaSPDIFSink * sink)
-{
- char *pcm_name = sink->device;
- int err;
- char devstr[256]; /* Storage for local 'default' device string */
-
- /*
- * 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 (sink,
- "Open failed for %s - searching for IEC958 manually\n", pcm_name);
-
- err = alsaspdifsink_find_pcm_device (sink);
- if (err == 0 && sink->pcm == NULL)
- goto open_failed;
- }
- if (err < 0)
- goto failed;
-
- return TRUE;
-
- /* ERRORS */
-open_failed:
- {
- GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
- ("Could not open IEC958/SPDIF output device"), GST_ERROR_SYSTEM);
- return FALSE;
- }
-failed:
- {
- GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
- ("snd_pcm_open: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
- return FALSE;
- }
-}
-
-static gboolean
-alsaspdifsink_set_params (AlsaSPDIFSink * sink)
-{
- snd_pcm_hw_params_t *params;
- unsigned int rate;
- int err;
-
- snd_pcm_hw_params_malloc (¶ms);
-
- 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 __error;
- }
-
- /* 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 __error;
- }
-
- err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_BE);
- if (err < 0) {
- /* Try LE output and swap data */
- GST_DEBUG_OBJECT (sink, "PCM format S16_BE not supported, trying S16_LE");
- 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 __error;
- }
-
- 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 __error;
- }
-
- rate = sink->rate;
- GST_DEBUG_OBJECT (sink, "Setting S/PDIF sample rate: %d", 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 __error;
- }
-
- 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 __error;
- }
-
- snd_pcm_hw_params_free (params);
-
- return TRUE;
-
- /* ERRORS */
-__error:
- {
- snd_pcm_hw_params_free (params);
- 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 = NULL;
- snd_ctl_card_info_t *info = NULL;
- snd_ctl_elem_list_t *clist = NULL;
- snd_ctl_elem_id_t *cid = NULL;
- snd_pcm_info_t *pinfo = NULL;
-
- GST_WARNING ("Opening IEC958 named device failed. Trying to autodetect");
-
- if ((err = snd_ctl_open (&ctl, ctl_name, card)) < 0)
- return err;
-
- snd_ctl_card_info_malloc (&info);
- snd_pcm_info_malloc (&pinfo);
-
- /* Find a mixer for IEC958 settings */
- snd_ctl_elem_list_malloc (&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_malloc (&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_malloc (&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);
- snd_ctl_elem_value_free (cval);
-
- sink->pcm = pcm;
- pcm = NULL;
- err = 0;
- }
-
-beach:
- if (pcm)
- snd_pcm_close (pcm);
- if (clist)
- snd_ctl_elem_list_clear (clist);
- if (ctl)
- snd_ctl_close (ctl);
- if (clist)
- snd_ctl_elem_list_free (clist);
- if (cid)
- snd_ctl_elem_id_free (cid);
- if (info)
- snd_ctl_card_info_free (info);
- if (pinfo)
- snd_pcm_info_free (pinfo);
- 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;
-}
-
-static snd_pcm_sframes_t
-alsaspdifsink_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 delay;
-}
-
-#if 0
-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;
- gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->clock), 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, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
+++ /dev/null
-dnl Configure Paths for Alsa
-dnl Some modifications by Richard Boulton <richard-alsa@tartarus.org>
-dnl Christopher Lansdown <lansdoct@cs.alfred.edu>
-dnl Jaroslav Kysela <perex@suse.cz>
-dnl Last modification: alsa.m4,v 1.23 2004/01/16 18:14:22 tiwai Exp
-dnl AM_PATH_ALSA([MINIMUM-VERSION [, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
-dnl Test for libasound, and define ALSA_CFLAGS and ALSA_LIBS as appropriate.
-dnl enables arguments --with-alsa-prefix=
-dnl --with-alsa-enc-prefix=
-dnl --disable-alsatest
-dnl
-dnl For backwards compatibility, if ACTION_IF_NOT_FOUND is not specified,
-dnl and the alsa libraries are not found, a fatal AC_MSG_ERROR() will result.
-dnl
-AC_DEFUN([AM_PATH_ALSA],
-[dnl Save the original CFLAGS, LDFLAGS, and LIBS
-alsa_save_CFLAGS="$CFLAGS"
-alsa_save_LDFLAGS="$LDFLAGS"
-alsa_save_LIBS="$LIBS"
-alsa_found=yes
-
-dnl
-dnl Get the cflags and libraries for alsa
-dnl
-AC_ARG_WITH(alsa-prefix,
- AC_HELP_STRING([--with-alsa-prefix=PFX],
- [prefix where Alsa library is installed(optional)]),
- [alsa_prefix="$withval"], [alsa_prefix=""])
-
-AC_ARG_WITH(alsa-inc-prefix,
- AC_HELP_STRING([--with-alsa-inc-prefix=PFX],
- [prefix where include libraries are (optional)]),
- [alsa_inc_prefix="$withval"], [alsa_inc_prefix=""])
-
-dnl FIXME: this is not yet implemented
-dnl AC_ARG_ENABLE(alsatest,
-dnl AC_HELP_STRING([--disable-alsatest],
-dnl [do not try to compile and run a test Alsa program],
-dnl [enable_alsatest=no], [enable_alsatest=yes])
-
-dnl Add any special include directories
-AC_MSG_CHECKING(for ALSA CFLAGS)
-if test "$alsa_inc_prefix" != "" ; then
- ALSA_CFLAGS="$ALSA_CFLAGS -I$alsa_inc_prefix"
- CFLAGS="$CFLAGS -I$alsa_inc_prefix"
-fi
-AC_MSG_RESULT($ALSA_CFLAGS)
-
-dnl add any special lib dirs
-AC_MSG_CHECKING(for ALSA LDFLAGS)
-if test "$alsa_prefix" != "" ; then
- ALSA_LIBS="$ALSA_LIBS -L$alsa_prefix"
- LDFLAGS="$LDFLAGS $ALSA_LIBS"
-fi
-
-dnl add the alsa library
-ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread"
-LIBS=`echo $LIBS | sed 's/-lm//'`
-LIBS=`echo $LIBS | sed 's/-ldl//'`
-LIBS=`echo $LIBS | sed 's/-lpthread//'`
-LIBS=`echo $LIBS | sed 's/ //'`
-LIBS="$ALSA_LIBS $LIBS"
-AC_MSG_RESULT($ALSA_LIBS)
-
-dnl Check for a working version of libasound that is of the right version.
-min_alsa_version=ifelse([$1], ,0.1.1,$1)
-AC_MSG_CHECKING(for libasound headers version >= $min_alsa_version)
-no_alsa=""
- alsa_min_major_version=`echo $min_alsa_version | \
- sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\1/'`
- alsa_min_minor_version=`echo $min_alsa_version | \
- sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\2/'`
- alsa_min_micro_version=`echo $min_alsa_version | \
- sed 's/\([[0-9]]*\).\([[0-9]]*\).\([[0-9]]*\)/\3/'`
-
-AC_LANG_SAVE
-AC_LANG_C
-AC_TRY_COMPILE([
-#include <alsa/asoundlib.h>
-], [
-void main(void)
-{
-/* ensure backward compatibility */
-#if !defined(SND_LIB_MAJOR) && defined(SOUNDLIB_VERSION_MAJOR)
-#define SND_LIB_MAJOR SOUNDLIB_VERSION_MAJOR
-#endif
-#if !defined(SND_LIB_MINOR) && defined(SOUNDLIB_VERSION_MINOR)
-#define SND_LIB_MINOR SOUNDLIB_VERSION_MINOR
-#endif
-#if !defined(SND_LIB_SUBMINOR) && defined(SOUNDLIB_VERSION_SUBMINOR)
-#define SND_LIB_SUBMINOR SOUNDLIB_VERSION_SUBMINOR
-#endif
-
-# if(SND_LIB_MAJOR > $alsa_min_major_version)
- exit(0);
-# else
-# if(SND_LIB_MAJOR < $alsa_min_major_version)
-# error not present
-# endif
-
-# if(SND_LIB_MINOR > $alsa_min_minor_version)
- exit(0);
-# else
-# if(SND_LIB_MINOR < $alsa_min_minor_version)
-# error not present
-# endif
-
-# if(SND_LIB_SUBMINOR < $alsa_min_micro_version)
-# error not present
-# endif
-# endif
-# endif
-exit(0);
-}
-],
- [AC_MSG_RESULT(found.)],
- [AC_MSG_RESULT(not present.)
- ifelse([$3], , [AC_MSG_ERROR(Sufficiently new version of libasound not found.)])
- alsa_found=no]
-)
-AC_LANG_RESTORE
-
-dnl Now that we know that we have the right version, let's see if we have the library and not just the headers.
-if test "x$enable_alsatest" = "xyes"; then
-AC_CHECK_LIB([asound], [snd_ctl_open],,
- [ifelse([$3], , [AC_MSG_ERROR(No linkable libasound was found.)])
- alsa_found=no]
-)
-fi
-
-if test "x$alsa_found" = "xyes" ; then
- ifelse([$2], , :, [$2])
- LIBS=`echo $LIBS | sed 's/-lasound//g'`
- LIBS=`echo $LIBS | sed 's/ //'`
- LIBS="-lasound $LIBS"
-fi
-if test "x$alsa_found" = "xno" ; then
- ifelse([$3], , :, [$3])
- CFLAGS="$alsa_save_CFLAGS"
- LDFLAGS="$alsa_save_LDFLAGS"
- LIBS="$alsa_save_LIBS"
- ALSA_CFLAGS=""
- ALSA_LIBS=""
-fi
-
-dnl That should be it. Now just export out symbols:
-AC_SUBST(ALSA_CFLAGS)
-AC_SUBST(ALSA_LIBS)
-])
-
projects / platform / upstream / gstreamer.git / commitdiff