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eqMerge remote-tracking branch 'origin/master' into 0.11

Browse source 
Conflicts:
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-read-common.c
	gst/rtpmanager/gstrtpssrcdemux.c
This commit is contained in:
Tim-Philipp Müller 2012-01-12 23:48:50 +00:00
commit 8580dd86c9
13 changed files with 255 additions and 58 deletions
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2
Android.mk
View file

@ -92,7 +92,7 @@ CONFIGURE_TARGETS += gst-plugins-good-configure
-include $(GST_PLUGINS_GOOD_TOP)/gst/law/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/level/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/matroska/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/multifile/Android.mk
#-include $(GST_PLUGINS_GOOD_TOP)/gst/multifile/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/multipart/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/isomp4/Android.mk
-include $(GST_PLUGINS_GOOD_TOP)/gst/replaygain/Android.mk

4
ext/flac/gstflacenc.c
View file

@ -1047,7 +1047,9 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
ret = gst_flac_enc_process_stream_headers (flacenc);
flacenc->got_headers = TRUE;
}
} else if (flacenc->got_headers && samples == 0) {
}
if (flacenc->got_headers && samples == 0) {
/* header fixup, push downstream directly */
GST_DEBUG_OBJECT (flacenc, "Fixing up headers at pos=%" G_GUINT64_FORMAT
", size=%u", flacenc->offset, (guint) bytes);

21
ext/jack/gstjack.c
View file

@ -46,6 +46,27 @@ gst_jack_connect_get_type (void)
return (GType) jack_connect_type;
}
GType
gst_jack_transport_get_type (void)
{
static volatile gsize type = 0;
if (g_once_init_enter (&type)) {
static const GEnumValue enum_values[] = {
{GST_JACK_TRANSPORT_AUTONOMOUS,
"No transport support", "autonomous"},
{GST_JACK_TRANSPORT_MASTER,
"Start and stop transport with state changes", "master"},
{GST_JACK_TRANSPORT_SLAVE,
"Follow transport state changes", "slave"},
{0, NULL, NULL},
};
GType tmp = g_enum_register_static ("GstJackTransport", enum_values);
g_once_init_leave (&type, tmp);
}
return (GType) type;
}
static gpointer
gst_jack_client_copy (gpointer jclient)

22
ext/jack/gstjack.h
View file

@ -37,17 +37,32 @@
*
* Specify how the output ports will be connected.
*/
typedef enum {
GST_JACK_CONNECT_NONE,
GST_JACK_CONNECT_AUTO,
GST_JACK_CONNECT_AUTO_FORCED
} GstJackConnect;
/**
* GstJackTransport:
* @GST_JACK_TRANSPORT_AUTONOMOUS: no transport support
* @GST_JACK_TRANSPORT_MASTER: start and stop transport with state-changes
* @GST_JACK_TRANSPORT_SLAVE: follow transport state changes
*
* The jack transport state allow to sync multiple clients. This enum defines a
* client behaviour regarding to the transport mechanism.
*/
typedef enum {
GST_JACK_TRANSPORT_AUTONOMOUS,
GST_JACK_TRANSPORT_MASTER,
GST_JACK_TRANSPORT_SLAVE
} GstJackTransport;
typedef jack_default_audio_sample_t sample_t;
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type ())
#define GST_TYPE_JACK_TRANSPORT (gst_jack_transport_get_type ())
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define GST_JACK_FORMAT_STR "F32LE"
@ -57,5 +72,6 @@ typedef jack_default_audio_sample_t sample_t;
GType gst_jack_client_get_type(void);
GType gst_jack_connect_get_type(void);
GType gst_jack_transport_get_type(void);
#endif // _GST_JACK_H_

49
ext/jack/gstjackaudioclient.c
View file

@ -55,6 +55,10 @@ typedef struct
gint n_clients;
GList *src_clients;
GList *sink_clients;
/* transport state handling */
gint cur_ts;
GstState transport_state;
} GstJackAudioConnection;
/* an object sharing a jack_client_t connection. */
@ -66,7 +70,7 @@ struct _GstJackAudioClient
gboolean active;
gboolean deactivate;
void (*shutdown) (void *arg);
JackShutdownCallback shutdown;
JackProcessCallback process;
JackBufferSizeCallback buffer_size;
JackSampleRateCallback sample_rate;
@ -87,6 +91,27 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
GList *walk;
int res = 0;
jack_transport_state_t ts = jack_transport_query (conn->client, NULL);
if (ts != conn->cur_ts) {
conn->cur_ts = ts;
switch (ts) {
case JackTransportStopped:
GST_DEBUG ("transport state is 'stopped'");
conn->transport_state = GST_STATE_PAUSED;
break;
case JackTransportStarting:
GST_DEBUG ("transport state is 'starting'");
conn->transport_state = GST_STATE_READY;
break;
case JackTransportRolling:
GST_DEBUG ("transport state is 'rolling'");
conn->transport_state = GST_STATE_PLAYING;
break;
default:
break;
}
}
g_mutex_lock (conn->lock);
/* call sources first, then sinks. Sources will either push data into the
@ -117,7 +142,6 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
}
}
g_mutex_unlock (conn->lock);
return res;
}
@ -224,6 +248,8 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server,
conn->n_clients = 0;
conn->src_clients = NULL;
conn->sink_clients = NULL;
conn->cur_ts = -1;
conn->transport_state = GST_STATE_VOID_PENDING;
/* set our callbacks */
jack_set_process_callback (jclient, jack_process_cb, conn);
@ -525,3 +551,22 @@ gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
return 0;
}
/**
* gst_jack_audio_client_get_transport_state:
* @client: a #GstJackAudioClient
*
* Check the current transport state. The client can use this to request a state
* change from the application.
*
* Returns: the state, %GST_STATE_VOID_PENDING for no change in the transport
* state
*/
GstState
gst_jack_audio_client_get_transport_state (GstJackAudioClient * client)
{
GstState state = client->conn->transport_state;
client->conn->transport_state = GST_STATE_VOID_PENDING;
return state;
}

2
ext/jack/gstjackaudioclient.h
View file

@ -54,6 +54,8 @@ jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *
gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active);
GstState gst_jack_audio_client_get_transport_state (GstJackAudioClient *client);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_CLIENT_H__ */

59
ext/jack/gstjackaudiosink.c
View file

@ -197,6 +197,18 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* handle transport state requisitions */
if (sink->transport == GST_JACK_TRANSPORT_SLAVE) {
GstState state = gst_jack_audio_client_get_transport_state (sink->client);
if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (sink) != state)) {
GST_DEBUG_OBJECT (sink, "requesting state change: %s",
gst_element_state_get_name (state));
gst_element_post_message (GST_ELEMENT (sink),
gst_message_new_request_state (GST_OBJECT (sink), state));
}
}
/* get target buffers */
for (i = 0; i < channels; i++) {
sink->buffers[i] =
@ -564,6 +576,13 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "start");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
jack_transport_start (client);
}
return TRUE;
}
@ -576,6 +595,13 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "pause");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
jack_transport_stop (client);
}
return TRUE;
}
@ -588,6 +614,13 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "stop");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
jack_transport_stop (client);
}
return TRUE;
}
@ -654,7 +687,8 @@ enum
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
#define DEFAULT_PROP_CLIENT_NAME NULL
#define DEFAULT_PROP_CLIENT_NAME NULL
#define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
enum
{
@ -663,6 +697,7 @@ enum
PROP_SERVER,
PROP_CLIENT,
PROP_CLIENT_NAME,
PROP_TRANSPORT,
PROP_LAST
};
@ -730,6 +765,19 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
/**
* GstJackAudioSink:transport
*
* The jack transport behaviour for the client.
*
* Since: 0.10.31
*/
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_enum ("transport", "Transport mode",
"Jack transport behaviour of the client",
GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_details_simple (gstelement_class, "Audio Sink (Jack)",
"Sink/Audio", "Output audio to a JACK server",
"Wim Taymans <wim.taymans@gmail.com>");
@ -757,8 +805,9 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink)
sink->jclient = NULL;
sink->ports = NULL;
sink->port_count = 0;
sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
sink->buffers = NULL;
sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
sink->transport = DEFAULT_PROP_TRANSPORT;
}
static void
@ -802,6 +851,9 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
sink->jclient = g_value_get_boxed (value);
}
break;
case PROP_TRANSPORT:
sink->transport = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -829,6 +881,9 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
case PROP_CLIENT:
g_value_set_boxed (value, sink->jclient);
break;
case PROP_TRANSPORT:
g_value_set_enum (value, sink->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;

1
ext/jack/gstjackaudiosink.h
View file

@ -59,6 +59,7 @@ struct _GstJackAudioSink {
gchar *server;
jack_client_t *jclient;
gchar *client_name;
GstJackTransport transport;
/* our client */
GstJackAudioClient *client;

60
ext/jack/gstjackaudiosrc.c
View file

@ -217,6 +217,18 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* handle transport state requisitions */
if (src->transport == GST_JACK_TRANSPORT_SLAVE) {
GstState state = gst_jack_audio_client_get_transport_state (src->client);
if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (src) != state)) {
GST_DEBUG_OBJECT (src, "requesting state change: %s",
gst_element_state_get_name (state));
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_request_state (GST_OBJECT (src), state));
}
}
/* get input buffers */
for (i = 0; i < channels; i++)
src->buffers[i] =
@ -576,6 +588,13 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "start");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_start (client);
}
return TRUE;
}
@ -588,6 +607,13 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "pause");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_stop (client);
}
return TRUE;
}
@ -600,6 +626,13 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "stop");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
jack_transport_stop (client);
}
return TRUE;
}
@ -658,6 +691,7 @@ enum
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
#define DEFAULT_PROP_CLIENT_NAME NULL
#define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
enum
{
@ -666,6 +700,7 @@ enum
PROP_SERVER,
PROP_CLIENT,
PROP_CLIENT_NAME,
PROP_TRANSPORT,
PROP_LAST
};
@ -751,6 +786,19 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
/**
* GstJackAudioSink:transport
*
* The jack transport behaviour for the client.
*
* Since: 0.10.31
*/
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_enum ("transport", "Transport mode",
"Jack transport behaviour of the client",
GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
@ -769,11 +817,6 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
gst_jack_audio_client_init ();
}
/* initialize the new element
* instantiate pads and add them to element
* set pad calback functions
* initialize instance structure
*/
static void
gst_jack_audio_src_init (GstJackAudioSrc * src)
{
@ -785,6 +828,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src)
src->port_count = 0;
src->buffers = NULL;
src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
src->transport = DEFAULT_PROP_TRANSPORT;
}
static void
@ -826,6 +870,9 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id,
src->jclient = g_value_get_boxed (value);
}
break;
case PROP_TRANSPORT:
src->transport = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -851,6 +898,9 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id,
case PROP_CLIENT:
g_value_set_boxed (value, src->jclient);
break;
case PROP_TRANSPORT:
g_value_set_enum (value, src->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;

1
ext/jack/gstjackaudiosrc.h
View file

@ -76,6 +76,7 @@ struct _GstJackAudioSrc
gchar *server;
jack_client_t *jclient;
gchar *client_name;
GstJackTransport transport;
/* our client */
GstJackAudioClient *client;

72
gst/matroska/matroska-mux.c
View file

@ -668,6 +668,17 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstObject * parent,
return gst_pad_event_default (pad, parent, event);
}
static void
gst_matroska_mux_free_codec_priv (GstMatroskaTrackContext * context)
{
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
}
static void
gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context,
const guint * clut)
@ -695,7 +706,7 @@ gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context,
sclut = g_strjoinv (",", clutv);
/* build codec private; only palette for now */
g_free (context->codec_priv);
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = (guint8 *) g_strdup_printf ("palette: %s", sclut);
/* include terminating 0 */
context->codec_priv_size = strlen ((gchar *) context->codec_priv) + 1;
@ -1046,18 +1057,13 @@ skip_details:
gst_matroska_mux_set_codec_id (context,
GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC);
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = (gpointer) bih;
context->codec_priv_size = size;
} else if (!strcmp (mimetype, "video/x-h264")) {
gst_matroska_mux_set_codec_id (context,
GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
/* Create avcC header */
if (codec_buf != NULL) {
context->codec_priv_size = gst_buffer_get_size (codec_buf);
@ -1069,11 +1075,7 @@ skip_details:
gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_THEORA);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
streamheader = gst_structure_get_value (structure, "streamheader");
if (!theora_streamheader_to_codecdata (streamheader, context)) {
@ -1108,6 +1110,7 @@ skip_details:
/* global headers may be in codec data */
if (codec_buf != NULL) {
gst_matroska_mux_free_codec_priv (context);
context->codec_priv_size = gst_buffer_get_size (codec_buf);
context->codec_priv = g_malloc0 (context->codec_priv_size);
gst_buffer_extract (codec_buf, 0, context->codec_priv, -1);
@ -1154,6 +1157,7 @@ skip_details:
gst_buffer_extract (codec_data_buf, 0, priv_data, -1);
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = priv_data;
context->codec_priv_size = priv_data_size;
}
@ -1237,6 +1241,7 @@ xiphN_streamheader_to_codecdata (const GValue * streamheader,
offset += gst_buffer_get_size (buf[i]);
}
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = priv_data;
context->codec_priv_size = priv_data_size;
@ -1421,6 +1426,7 @@ flac_streamheader_to_codecdata (const GValue * streamheader,
return FALSE;
}
gst_matroska_mux_free_codec_priv (context);
context->codec_priv_size = gst_buffer_get_size (buffer) - 9;
context->codec_priv = g_malloc (context->codec_priv_size);
gst_buffer_extract (buffer, 9, context->codec_priv, -1);
@ -1430,9 +1436,7 @@ flac_streamheader_to_codecdata (const GValue * streamheader,
bufval = &g_array_index (bufarr, GValue, i);
if (G_VALUE_TYPE (bufval) != GST_TYPE_BUFFER) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
gst_matroska_mux_free_codec_priv (context);
GST_WARNING ("streamheaders array does not contain GstBuffers");
return FALSE;
}
@ -1487,6 +1491,7 @@ speex_streamheader_to_codecdata (const GValue * streamheader,
return FALSE;
}
gst_matroska_mux_free_codec_priv (context);
context->codec_priv_size = gst_buffer_get_size (buffer);
context->codec_priv = g_malloc (context->codec_priv_size);
gst_buffer_extract (buffer, 0, context->codec_priv, -1);
@ -1494,9 +1499,7 @@ speex_streamheader_to_codecdata (const GValue * streamheader,
bufval = &g_array_index (bufarr, GValue, 1);
if (G_VALUE_TYPE (bufval) != GST_TYPE_BUFFER) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
gst_matroska_mux_free_codec_priv (context);
GST_WARNING ("streamheaders array does not contain GstBuffers");
return FALSE;
}
@ -1729,11 +1732,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_VORBIS);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
streamheader = gst_structure_get_value (structure, "streamheader");
if (!vorbis_streamheader_to_codecdata (streamheader, context)) {
@ -1745,11 +1744,8 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
const GValue *streamheader;
gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_FLAC);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
streamheader = gst_structure_get_value (structure, "streamheader");
if (!flac_streamheader_to_codecdata (streamheader, context)) {
@ -1761,11 +1757,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
const GValue *streamheader;
gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_SPEEX);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
streamheader = gst_structure_get_value (structure, "streamheader");
if (!speex_streamheader_to_codecdata (streamheader, context)) {
@ -1823,6 +1815,8 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
gst_buffer_extract (codec_data_buf, 0, priv_data, -1);
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = priv_data;
context->codec_priv_size = priv_data_size;
}
@ -1907,6 +1901,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
}
gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_ACM);
gst_matroska_mux_free_codec_priv (context);
context->codec_priv = (gpointer) codec_priv;
context->codec_priv_size = codec_priv_size;
}
@ -1981,11 +1976,7 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
gst_matroska_mux_set_codec_id (context,
GST_MATROSKA_CODEC_ID_SUBTITLE_KATE);
if (context->codec_priv != NULL) {
g_free (context->codec_priv);
context->codec_priv = NULL;
context->codec_priv_size = 0;
}
gst_matroska_mux_free_codec_priv (context);
streamheader = gst_structure_get_value (structure, "streamheader");
if (!kate_streamheader_to_codecdata (streamheader, context)) {
@ -2030,8 +2021,7 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
return TRUE;
}
if (context->codec_priv != NULL)
g_free (context->codec_priv);
gst_matroska_mux_free_codec_priv (context);
priv_data = g_malloc0 (priv_data_size);
memcpy (priv_data, priv_buffer_data, priv_data_size);

3
gst/matroska/matroska-read-common.c
View file

@ -650,6 +650,9 @@ gst_matroska_read_common_parse_attached_file (GstMatroskaReadCommon * common,
else
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_ATTACHMENT,
tagsample, NULL);
/* the list adds it own ref */
gst_buffer_unref (tagsample);
}
g_free (filename);

17
tests/check/pipelines/wavenc.c
View file

@ -54,7 +54,7 @@ bus_handler (GstBus * bus, GstMessage * message, gpointer data)
/*
* gst-launch \
* audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! wavenc ! filesink location=/tmp/mc.wav \
* audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! audioconvert ! wavenc ! filesink location=/tmp/mc.wav \
* audiotestsrc freq=880 num-buffers=100 ! i.
* ...
*
@ -65,7 +65,7 @@ static void
make_n_channel_wav (const gint channels, const GValueArray * arr)
{
GstElement *pipeline;
GstElement **audiotestsrc, *interleave, *wavenc, *fakesink;
GstElement **audiotestsrc, *interleave, *wavenc, *conv, *fakesink;
GstBus *bus;
GMainLoop *loop;
guint i;
@ -81,10 +81,21 @@ make_n_channel_wav (const gint channels, const GValueArray * arr)
g_object_set (interleave, "channel-positions", arr, NULL);
gst_bin_add (GST_BIN (pipeline), interleave);
if (G_BYTE_ORDER == G_BIG_ENDIAN) {
/* we're not here to test orc; audioconvert misbehaves on ppc32 */
g_setenv ("ORC_CODE", "backup", 1);
conv = gst_element_factory_make ("audioconvert", NULL);
} else {
conv = gst_element_factory_make ("identity", NULL);
}
fail_unless (conv != NULL);
gst_bin_add (GST_BIN (pipeline), conv);
fail_unless (gst_element_link (interleave, conv));
wavenc = gst_element_factory_make ("wavenc", NULL);
fail_unless (wavenc != NULL);
gst_bin_add (GST_BIN (pipeline), wavenc);
fail_unless (gst_element_link (interleave, wavenc));
fail_unless (gst_element_link (conv, wavenc));
fakesink = gst_element_factory_make ("fakesink", NULL);
fail_unless (fakesink != NULL);