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

Browse source 
Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstamrparse.c
	gst/audioparsers/gstdcaparse.c
	gst/audioparsers/gstflacparse.c
	gst/effectv/gstradioac.c
	gst/effectv/gstradioac.h
	gst/effectv/gstripple.c

Some possible FIXMEs remaining in the audio parser getcaps functions.
This commit is contained in:
Tim-Philipp Müller 2011-11-26 13:34:10 +00:00
commit be0d6baac5
22 changed files with 750 additions and 117 deletions
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4
ext/pulse/pulseaudiosink.c
View file

@ -535,7 +535,7 @@ dbin_event_probe (GstPad * pad, GstPadProbeInfo * info, gpointer data)
static void
pad_added_cb (GstElement * dbin, GstPad * pad, gpointer * data)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
GstPulseAudioSink *pbin;
GstPad *sinkpad = NULL;
pbin = GST_PULSE_AUDIO_SINK (data);
@ -584,11 +584,13 @@ gst_pulse_audio_sink_add_dbin (GstPulseAudioSink * pbin)
/* Trap the newsegment events that we feed the decodebin and discard them */
sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
if (pbin->event_probe_id == 0)
pbin->event_probe_id =
gst_pad_add_probe (sinkpad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
dbin_event_probe, gst_object_ref (pbin),
(GDestroyNotify) gst_object_unref);
gst_object_unref (sinkpad);
sinkpad = NULL;

409
ext/pulse/pulsesrc.c
View file

@ -43,6 +43,9 @@
#include <gst/base/gstbasesrc.h>
#include <gst/gsttaglist.h>
#ifdef HAVE_PULSE_1_0
#include <gst/interfaces/streamvolume.h>
#endif
#include "pulsesrc.h"
#include "pulseutil.h"
@ -55,6 +58,12 @@ GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define DEFAULT_DEVICE NULL
#define DEFAULT_DEVICE_NAME NULL
#ifdef HAVE_PULSE_1_0
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
#endif
enum
{
PROP_0,
@ -64,6 +73,10 @@ enum
PROP_CLIENT,
PROP_STREAM_PROPERTIES,
PROP_SOURCE_OUTPUT_INDEX,
#ifdef HAVE_PULSE_1_0
PROP_VOLUME,
PROP_MUTE,
#endif
PROP_LAST
};
@ -126,7 +139,8 @@ GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSrc, gst_pulsesrc);
G_DEFINE_TYPE_WITH_CODE (GstPulseSrc, gst_pulsesrc, GST_TYPE_AUDIO_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_MIXER, gst_pulsesrc_mixer_interface_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_PROPERTY_PROBE,
gst_pulsesrc_property_probe_interface_init));
gst_pulsesrc_property_probe_interface_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
static void
gst_pulsesrc_class_init (GstPulseSrcClass * klass)
@ -174,7 +188,7 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
clientname = gst_pulse_client_name ();
/**
* GstPulseSink:client
* GstPulseSrc:client
*
* The PulseAudio client name to use.
*
@ -230,6 +244,35 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
"Captures audio from a PulseAudio server", "Lennart Poettering");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&pad_template));
#ifdef HAVE_PULSE_1_0
/**
* GstPulseSrc:volume
*
* The volume of the record stream. Only works when using PulseAudio 1.0 or
* later.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class,
PROP_VOLUME, g_param_spec_double ("volume", "Volume",
"Linear volume of this stream, 1.0=100%",
0.0, MAX_VOLUME, DEFAULT_VOLUME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstPulseSrc:mute
*
* Whether the stream is muted or not. Only works when using PulseAudio 1.0
* or later.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class,
PROP_MUTE, g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream",
DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
}
static void
@ -254,6 +297,16 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
pulsesrc->paused = TRUE;
pulsesrc->in_read = FALSE;
#ifdef HAVE_PULSE_1_0
pulsesrc->volume = DEFAULT_VOLUME;
pulsesrc->volume_set = FALSE;
pulsesrc->mute = DEFAULT_MUTE;
pulsesrc->mute_set = FALSE;
pulsesrc->notify = 0;
#endif
pulsesrc->mixer = NULL;
pulsesrc->properties = NULL;
@ -290,7 +343,15 @@ gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
if (pulsesrc->context) {
pa_context_disconnect (pulsesrc->context);
/* Make sure we don't get any further callbacks */
pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
#ifdef HAVE_PULSE_1_0
pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
#endif
pa_context_unref (pulsesrc->context);
pulsesrc->context = NULL;
}
}
@ -408,6 +469,260 @@ no_mainloop:
}
}
#ifdef HAVE_PULSE_1_0
static void
gst_pulsesrc_source_output_info_cb (pa_context * c,
const pa_source_output_info * i, int eol, void *userdata)
{
GstPulseSrc *psrc;
psrc = GST_PULSESRC_CAST (userdata);
if (!i)
goto done;
/* If the index doesn't match our current stream,
* it implies we just recreated the stream (caps change)
*/
if (i->index == psrc->source_output_idx) {
psrc->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
psrc->mute = i->mute;
}
done:
pa_threaded_mainloop_signal (psrc->mainloop, 0);
}
static gdouble
gst_pulsesrc_get_stream_volume (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gdouble v;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(o = pa_context_get_source_output_info (pulsesrc->context,
pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
pulsesrc)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock;
}
unlock:
v = pulsesrc->volume;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
if (v > MAX_VOLUME) {
GST_WARNING_OBJECT (pulsesrc, "Clipped volume from %f to %f", v,
MAX_VOLUME);
v = MAX_VOLUME;
}
return v;
/* ERRORS */
no_mainloop:
{
v = pulsesrc->volume;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return v;
}
no_index:
{
v = pulsesrc->volume;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return v;
}
info_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_get_source_output_info() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static gboolean
gst_pulsesrc_get_stream_mute (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gboolean mute;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(o = pa_context_get_source_output_info (pulsesrc->context,
pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
pulsesrc)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock;
}
unlock:
mute = pulsesrc->mute;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return mute;
/* ERRORS */
no_mainloop:
{
mute = pulsesrc->mute;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return mute;
}
no_index:
{
mute = pulsesrc->mute;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return mute;
}
info_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_get_source_output_info() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesrc_set_stream_volume (GstPulseSrc * pulsesrc, gdouble volume)
{
pa_cvolume v;
pa_operation *o = NULL;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (!pulsesrc->source_output_idx)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
GST_DEBUG_OBJECT (pulsesrc, "setting volume to %f", volume);
gst_pulse_cvolume_from_linear (&v, pulsesrc->sample_spec.channels, volume);
if (!(o = pa_context_set_source_output_volume (pulsesrc->context,
pulsesrc->source_output_idx, &v, NULL, NULL)))
goto volume_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
pulsesrc->volume = volume;
pulsesrc->volume_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return;
}
no_index:
{
pulsesrc->volume = volume;
pulsesrc->volume_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return;
}
volume_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_set_source_output_volume() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesrc_set_stream_mute (GstPulseSrc * pulsesrc, gboolean mute)
{
pa_operation *o = NULL;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (!pulsesrc->source_output_idx)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
GST_DEBUG_OBJECT (pulsesrc, "setting mute state to %d", mute);
if (!(o = pa_context_set_source_output_mute (pulsesrc->context,
pulsesrc->source_output_idx, mute, NULL, NULL)))
goto mute_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
pulsesrc->mute = mute;
pulsesrc->mute_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return;
}
no_index:
{
pulsesrc->mute = mute;
pulsesrc->mute_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return;
}
mute_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_set_source_output_mute() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
#endif
static void
gst_pulsesrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
@ -444,6 +759,14 @@ gst_pulsesrc_set_property (GObject * object,
pa_proplist_free (pulsesrc->proplist);
pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
break;
#ifdef HAVE_PULSE_1_0
case PROP_VOLUME:
gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
break;
case PROP_MUTE:
gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -476,6 +799,14 @@ gst_pulsesrc_get_property (GObject * object,
case PROP_SOURCE_OUTPUT_INDEX:
g_value_set_uint (value, pulsesrc->source_output_idx);
break;
#ifdef HAVE_PULSE_1_0
case PROP_VOLUME:
g_value_set_double (value, gst_pulsesrc_get_stream_volume (pulsesrc));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_pulsesrc_get_stream_mute (pulsesrc));
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -569,6 +900,30 @@ gst_pulsesrc_stream_overflow_cb (pa_stream * s, void *userdata)
GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got overflow");
}
#ifdef HAVE_PULSE_1_0
static void
gst_pulsesrc_context_subscribe_cb (pa_context * c,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseSrc *psrc = GST_PULSESRC (userdata);
if (t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_CHANGE)
&& t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_NEW))
return;
if (idx != psrc->source_output_idx)
return;
/* Actually this event is also triggered when other properties of the stream
* change that are unrelated to the volume. However it is probably cheaper to
* signal the change here and check for the volume when the GObject property
* is read instead of querying it always. */
/* inform streaming thread to notify */
g_atomic_int_compare_and_exchange (&psrc->notify, 0, 1);
}
#endif
static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
@ -591,6 +946,10 @@ gst_pulsesrc_open (GstAudioSrc * asrc)
pa_context_set_state_callback (pulsesrc->context,
gst_pulsesrc_context_state_cb, pulsesrc);
#ifdef HAVE_PULSE_1_0
pa_context_set_subscribe_callback (pulsesrc->context,
gst_pulsesrc_context_subscribe_cb, pulsesrc);
#endif
GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
GST_STR_NULL (pulsesrc->server));
@ -672,6 +1031,13 @@ gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
pa_threaded_mainloop_lock (pulsesrc->mainloop);
pulsesrc->in_read = TRUE;
#ifdef HAVE_PULSE_1_0
if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
g_object_notify (G_OBJECT (pulsesrc), "volume");
g_object_notify (G_OBJECT (pulsesrc), "mute");
}
#endif
if (pulsesrc->paused)
goto was_paused;
@ -985,9 +1351,27 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
#ifdef HAVE_PULSE_1_0
pa_operation *o;
#endif
pa_threaded_mainloop_lock (pulsesrc->mainloop);
#ifdef HAVE_PULSE_1_0
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_operation_unref (o);
#endif
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
@ -1000,11 +1384,19 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED;
#ifdef HAVE_PULSE_1_0
if (pulsesrc->mute_set && pulsesrc->mute)
flags |= PA_STREAM_START_MUTED;
#endif
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED) < 0)
flags) < 0) {
goto connect_failed;
}
pulsesrc->corked = TRUE;
@ -1028,6 +1420,13 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
#ifdef HAVE_PULSE_1_0
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
#endif
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);

9
ext/pulse/pulsesrc.h
View file

@ -72,6 +72,15 @@ struct _GstPulseSrc
GstPulseMixerCtrl *mixer;
GstPulseProbe *probe;
#ifdef HAVE_PULSE_1_0
gdouble volume;
gboolean volume_set:1;
gboolean mute:1;
gboolean mute_set:1;
gint notify; /* atomic */
#endif
gboolean corked:1;
gboolean stream_connected:1;
gboolean operation_success:1;

54
gst/audioparsers/gstaacparse.c
View file

@ -69,16 +69,18 @@ GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
gboolean gst_aac_parse_start (GstBaseParse * parse);
gboolean gst_aac_parse_stop (GstBaseParse * parse);
static gboolean gst_aac_parse_start (GstBaseParse * parse);
static gboolean gst_aac_parse_stop (GstBaseParse * parse);
static gboolean gst_aac_parse_sink_setcaps (GstBaseParse * parse,
GstCaps * caps);
static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
GstCaps * filter);
gboolean gst_aac_parse_check_valid_frame (GstBaseParse * parse,
static gboolean gst_aac_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * size, gint * skipsize);
GstFlowReturn gst_aac_parse_parse_frame (GstBaseParse * parse,
static GstFlowReturn gst_aac_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
gboolean gst_aac_parse_convert (GstBaseParse * parse,
@ -131,6 +133,7 @@ gst_aac_parse_class_init (GstAacParseClass * klass)
parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_parse_frame);
parse_class->check_valid_frame =
GST_DEBUG_FUNCPTR (gst_aac_parse_check_valid_frame);
@ -568,7 +571,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
*
* Returns: TRUE if buffer contains a valid frame.
*/
gboolean
static gboolean
gst_aac_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
{
@ -643,7 +646,7 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
* Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed
* forward. Otherwise appropriate error is returned.
*/
GstFlowReturn
static GstFlowReturn
gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
GstAacParse *aacparse;
@ -700,7 +703,7 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
*
* Returns: TRUE if startup succeeded.
*/
gboolean
static gboolean
gst_aac_parse_start (GstBaseParse * parse)
{
GstAacParse *aacparse;
@ -721,9 +724,44 @@ gst_aac_parse_start (GstBaseParse * parse)
*
* Returns: TRUE is stopping succeeded.
*/
gboolean
static gboolean
gst_aac_parse_stop (GstBaseParse * parse)
{
GST_DEBUG ("stop");
return TRUE;
}
static GstCaps *
gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

41
gst/audioparsers/gstac3parse.c
View file

@ -165,6 +165,8 @@ static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static gboolean gst_ac3_parse_src_event (GstBaseParse * parse,
GstEvent * event);
static GstCaps *gst_ac3_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
#define gst_ac3_parse_parent_class parent_class
G_DEFINE_TYPE (GstAc3Parse, gst_ac3_parse, GST_TYPE_BASE_PARSE);
@ -195,8 +197,8 @@ gst_ac3_parse_class_init (GstAc3ParseClass * klass)
parse_class->check_valid_frame =
GST_DEBUG_FUNCPTR (gst_ac3_parse_check_valid_frame);
parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_parse_frame);
parse_class->src_event = GST_DEBUG_FUNCPTR (gst_ac3_parse_src_event);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_ac3_parse_get_sink_caps);
}
static void
@ -702,3 +704,40 @@ gst_ac3_parse_src_event (GstBaseParse * parse, GstEvent * event)
return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);
}
static GstCaps *
gst_ac3_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed and alignment field. We can convert
* between different alignments. */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
gst_structure_remove_field (s, "alignment");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

57
gst/audioparsers/gstamrparse.c
View file

@ -71,16 +71,18 @@ static const gint block_size_wb[16] =
#define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND)
#define AMR_MIME_HEADER_SIZE 9
gboolean gst_amr_parse_start (GstBaseParse * parse);
gboolean gst_amr_parse_stop (GstBaseParse * parse);
static gboolean gst_amr_parse_start (GstBaseParse * parse);
static gboolean gst_amr_parse_stop (GstBaseParse * parse);
static gboolean gst_amr_parse_sink_setcaps (GstBaseParse * parse,
GstCaps * caps);
static GstCaps *gst_amr_parse_sink_getcaps (GstBaseParse * parse,
GstCaps * filter);
gboolean gst_amr_parse_check_valid_frame (GstBaseParse * parse,
static gboolean gst_amr_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize);
GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
static GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
@ -112,6 +114,7 @@ gst_amr_parse_class_init (GstAmrParseClass * klass)
parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_setcaps);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_getcaps);
parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_amr_parse_parse_frame);
parse_class->check_valid_frame =
GST_DEBUG_FUNCPTR (gst_amr_parse_check_valid_frame);
@ -251,7 +254,7 @@ gst_amr_parse_parse_header (GstAmrParse * amrparse,
*
* Returns: TRUE if the given data contains valid frame.
*/
gboolean
static gboolean
gst_amr_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
{
@ -338,7 +341,7 @@ done:
*
* Returns: #GstFlowReturn defining the parsing status.
*/
GstFlowReturn
static GstFlowReturn
gst_amr_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
return GST_FLOW_OK;
@ -353,7 +356,7 @@ gst_amr_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
*
* Returns: TRUE on success.
*/
gboolean
static gboolean
gst_amr_parse_start (GstBaseParse * parse)
{
GstAmrParse *amrparse;
@ -374,7 +377,7 @@ gst_amr_parse_start (GstBaseParse * parse)
*
* Returns: TRUE on success.
*/
gboolean
static gboolean
gst_amr_parse_stop (GstBaseParse * parse)
{
GstAmrParse *amrparse;
@ -385,3 +388,41 @@ gst_amr_parse_stop (GstBaseParse * parse)
amrparse->header = 0;
return TRUE;
}
static GstCaps *
gst_amr_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Rename structure names */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
if (gst_structure_has_name (s, "audio/AMR"))
gst_structure_set_name (s, "audio/x-amr-nb-sh");
else
gst_structure_set_name (s, "audio/x-amr-wb-sh");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

38
gst/audioparsers/gstdcaparse.c
View file

@ -78,6 +78,8 @@ static gboolean gst_dca_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * size, gint * skipsize);
static GstFlowReturn gst_dca_parse_parse_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
#define gst_dca_parse_parent_class parent_class
G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE);
@ -99,6 +101,7 @@ gst_dca_parse_class_init (GstDcaParseClass * klass)
parse_class->check_valid_frame =
GST_DEBUG_FUNCPTR (gst_dca_parse_check_valid_frame);
parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_parse_frame);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
@ -464,3 +467,38 @@ broken_header:
return GST_FLOW_ERROR;
}
}
static GstCaps *
gst_dca_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

39
gst/audioparsers/gstflacparse.c
View file

@ -201,6 +201,8 @@ static GstFlowReturn gst_flac_parse_pre_push_frame (GstBaseParse * parse,
static gboolean gst_flac_parse_convert (GstBaseParse * parse,
GstFormat src_format, gint64 src_value, GstFormat dest_format,
gint64 * dest_value);
static GstCaps *gst_flac_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
#define gst_flac_parse_parent_class parent_class
G_DEFINE_TYPE (GstFlacParse, gst_flac_parse, GST_TYPE_BASE_PARSE);
@ -233,6 +235,8 @@ gst_flac_parse_class_init (GstFlacParseClass * klass)
baseparse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_flac_parse_pre_push_frame);
baseparse_class->convert = GST_DEBUG_FUNCPTR (gst_flac_parse_convert);
baseparse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_flac_parse_get_sink_caps);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
@ -1502,3 +1506,38 @@ gst_flac_parse_convert (GstBaseParse * parse,
return GST_BASE_PARSE_CLASS (parent_class)->convert (parse, src_format,
src_value, dest_format, dest_value);
}
static GstCaps *
gst_flac_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

39
gst/audioparsers/gstmpegaudioparse.c
View file

@ -100,6 +100,8 @@ static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
GstFormat src_format, gint64 src_value,
GstFormat dest_format, gint64 * dest_value);
static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
#define gst_mpeg_audio_parse_parent_class parent_class
G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
@ -161,6 +163,8 @@ gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
/* register tags */
#define GST_TAG_CRC "has-crc"
@ -1288,3 +1292,38 @@ gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
return GST_FLOW_OK;
}
static GstCaps *
gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the parsed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "parsed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

6
gst/effectv/gstedge.c
View file

@ -140,7 +140,7 @@ gst_edgetv_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
p = *src;
q = *(src - 4);
/* difference between the current pixel and right neighbor. */
/* difference between the current pixel and left neighbor. */
r = ((p & 0xff0000) - (q & 0xff0000)) >> 16;
g = ((p & 0xff00) - (q & 0xff00)) >> 8;
b = (p & 0xff) - (q & 0xff);
@ -199,8 +199,12 @@ gst_edgetv_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
dest[width + 3] = v3;
dest[width * 2] = v2;
dest[width * 2 + 1] = v2;
dest[width * 2 + 2] = 0;
dest[width * 2 + 3] = 0;
dest[width * 3] = v2;
dest[width * 3 + 1] = v2;
dest[width * 3 + 2] = 0;
dest[width * 3 + 3] = 0;
src += 4;
dest += 4;

3
gst/effectv/gstquark.c
View file

@ -328,7 +328,4 @@ gst_quarktv_init (GstQuarkTV * filter)
{
filter->planes = PLANES;
filter->current_plane = filter->planes - 1;
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

26
gst/effectv/gstradioac.c
View file

@ -131,14 +131,15 @@ enum
#define RATIO 0.95
static guint32 palettes[COLORS * PATTERN];
static gint swap_tab[] = { 2, 1, 0, 3 };
#define gst_radioactv_parent_class parent_class
G_DEFINE_TYPE (GstRadioacTV, gst_radioactv, GST_TYPE_VIDEO_FILTER);
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define CAPS_STR GST_VIDEO_CAPS_MAKE ("RGBx")
#define CAPS_STR GST_VIDEO_CAPS_MAKE ("{ RGBx, BGRx }")
#else
#define CAPS_STR GST_VIDEO_CAPS_MAKE ("xBGR")
#define CAPS_STR GST_VIDEO_CAPS_MAKE ("{ xBGR, xRGB }")
#endif
static GstStaticPadTemplate gst_radioactv_src_template =
@ -162,18 +163,20 @@ makePalette (void)
#define DELTA (255/(COLORS/2-1))
/* red, gree, blue */
for (i = 0; i < COLORS / 2; i++) {
palettes[i] = i * DELTA;
palettes[COLORS + i] = (i * DELTA) << 8;
palettes[COLORS * 2 + i] = (i * DELTA) << 16;
}
for (i = 0; i < COLORS / 2; i++) {
palettes[+i + COLORS / 2] = 255 | (i * DELTA) << 16 | (i * DELTA) << 8;
palettes[i + COLORS / 2] = 255 | (i * DELTA) << 16 | (i * DELTA) << 8;
palettes[COLORS + i + COLORS / 2] =
(255 << 8) | (i * DELTA) << 16 | i * DELTA;
palettes[COLORS * 2 + i + COLORS / 2] =
(255 << 16) | (i * DELTA) << 8 | i * DELTA;
}
/* white */
for (i = 0; i < COLORS; i++) {
palettes[COLORS * 3 + i] = (255 * i / COLORS) * 0x10101;
}
@ -347,7 +350,19 @@ gst_radioactv_transform (GstBaseTransform * trans, GstBuffer * in,
height = GST_VIDEO_FRAME_HEIGHT (&in_frame);
GST_OBJECT_LOCK (filter);
palette = &palettes[COLORS * filter->color];
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
if (GST_VIDEO_FRAME_FORMAT (&in_frame) == GST_VIDEO_FORMAT_RGBx) {
palette = &palettes[COLORS * filter->color];
} else {
palette = &palettes[COLORS * swap_tab[filter->color]];
}
#else
if (GST_VIDEO_FRAME_FORMAT (&in_frame) == GST_VIDEO_FORMAT_xBGR) {
palette = &palettes[COLORS * filter->color];
} else {
palette = &palettes[COLORS * swap_tab[filter->color]];
}
#endif
diff = filter->diff;
if (filter->mode == 3 && filter->trigger)
@ -643,7 +658,4 @@ gst_radioactv_init (GstRadioacTV * filter)
filter->color = DEFAULT_COLOR;
filter->interval = DEFAULT_INTERVAL;
filter->trigger = DEFAULT_TRIGGER;
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

93
gst/effectv/gstripple.c
View file

@ -311,9 +311,9 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
guint32 *src, *dest;
GstVideoFrame in_frame, out_frame;
gint x, y, i;
gint dx, dy;
gint dx, dy, o_dx;
gint h, v;
gint width, height;
gint m_w, m_h, v_w, v_h;
gint *p, *q, *r;
gint8 *vp;
GstClockTime timestamp, stream_time;
@ -344,22 +344,24 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
else
motiondetect (filter, src);
m_w = filter->map_w;
m_h = filter->map_h;
v_w = GST_VIDEO_FRAME_WIDTH (&in_frame);
v_h = GST_VIDEO_FRAME_HEIGHT (&in_frame);
/* simulate surface wave */
width = filter->map_w;
height = filter->map_h;
/* This function is called only 30 times per second. To increase a speed
* of wave, iterates this loop several times. */
for (i = loopnum; i > 0; i--) {
/* wave simulation */
p = filter->map1 + width + 1;
q = filter->map2 + width + 1;
r = filter->map3 + width + 1;
for (y = height - 2; y > 0; y--) {
for (x = width - 2; x > 0; x--) {
h = *(p - width - 1) + *(p - width + 1) + *(p + width - 1) + *(p +
width + 1)
+ *(p - width) + *(p - 1) + *(p + 1) + *(p + width) - (*p) * 9;
p = filter->map1 + m_w + 1;
q = filter->map2 + m_w + 1;
r = filter->map3 + m_w + 1;
for (y = m_h - 2; y > 0; y--) {
for (x = m_w - 2; x > 0; x--) {
h = *(p - m_w - 1) + *(p - m_w + 1) + *(p + m_w - 1) + *(p + m_w + 1)
+ *(p - m_w) + *(p - 1) + *(p + 1) + *(p + m_w) - (*p) * 9;
h = h >> 3;
v = *p - *q;
v += h - (v >> decay);
@ -374,11 +376,11 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
}
/* low pass filter */
p = filter->map3 + width + 1;
q = filter->map2 + width + 1;
for (y = height - 2; y > 0; y--) {
for (x = width - 2; x > 0; x--) {
h = *(p - width) + *(p - 1) + *(p + 1) + *(p + width) + (*p) * 60;
p = filter->map3 + m_w + 1;
q = filter->map2 + m_w + 1;
for (y = m_h - 2; y > 0; y--) {
for (x = m_w - 2; x > 0; x--) {
h = *(p - m_w) + *(p - 1) + *(p + 1) + *(p + m_w) + (*p) * 60;
*q = h >> 6;
p++;
q++;
@ -394,12 +396,12 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
vp = filter->vtable;
p = filter->map1;
for (y = height - 1; y > 0; y--) {
for (x = width - 1; x > 0; x--) {
for (y = m_h - 1; y > 0; y--) {
for (x = m_w - 1; x > 0; x--) {
/* difference of the height between two voxel. They are twiced to
* emphasise the wave. */
vp[0] = sqrtable[((p[0] - p[1]) >> (point - 1)) & 0xff];
vp[1] = sqrtable[((p[0] - p[width]) >> (point - 1)) & 0xff];
vp[1] = sqrtable[((p[0] - p[m_w]) >> (point - 1)) & 0xff];
p++;
vp += 2;
}
@ -407,48 +409,34 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
vp += 2;
}
width = GST_VIDEO_FRAME_WIDTH (&in_frame);
height = GST_VIDEO_FRAME_HEIGHT (&in_frame);
vp = filter->vtable;
/* draw refracted image. The vector table is stretched. */
for (y = 0; y < height; y += 2) {
for (x = 0; x < width; x += 2) {
for (y = 0; y < v_h; y += 2) {
for (x = 0; x < v_w; x += 2) {
h = (gint) vp[0];
v = (gint) vp[1];
dx = x + h;
dy = y + v;
if (dx < 0)
dx = 0;
if (dy < 0)
dy = 0;
if (dx >= width)
dx = width - 1;
if (dy >= height)
dy = height - 1;
dest[0] = src[dy * width + dx];
dx = CLAMP (dx, 0, (v_w - 1));
dy = CLAMP (dy, 0, (v_h - 1));
dest[0] = src[dy * v_w + dx];
i = dx;
o_dx = dx;
dx = x + 1 + (h + (gint) vp[2]) / 2;
if (dx < 0)
dx = 0;
if (dx >= width)
dx = width - 1;
dest[1] = src[dy * width + dx];
dx = CLAMP (dx, 0, (v_w - 1));
dest[1] = src[dy * v_w + dx];
dy = y + 1 + (v + (gint) vp[filter->map_w * 2 + 1]) / 2;
if (dy < 0)
dy = 0;
if (dy >= height)
dy = height - 1;
dest[width] = src[dy * width + i];
dy = y + 1 + (v + (gint) vp[m_w * 2 + 1]) / 2;
dy = CLAMP (dy, 0, (v_h - 1));
dest[v_w] = src[dy * v_w + o_dx];
dest[width + 1] = src[dy * width + dx];
dest[v_w + 1] = src[dy * v_w + dx];
dest += 2;
vp += 2;
}
dest += width;
dest += v_w;
vp += 2;
}
GST_OBJECT_UNLOCK (filter);
@ -493,9 +481,11 @@ gst_rippletv_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
filter->map_h = height / 2 + 1;
filter->map_w = width / 2 + 1;
/* we over allocate the buffers, as the render code does not handle clipping
* very well */
if (filter->map)
g_free (filter->map);
filter->map = g_new0 (gint, filter->map_h * filter->map_w * 3);
filter->map = g_new0 (gint, (1 + filter->map_h) * filter->map_w * 3);
filter->map1 = filter->map;
filter->map2 = filter->map + filter->map_w * filter->map_h;
@ -503,15 +493,15 @@ gst_rippletv_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
if (filter->vtable)
g_free (filter->vtable);
filter->vtable = g_new0 (gint8, filter->map_h * filter->map_w * 2);
filter->vtable = g_new0 (gint8, (1 + filter->map_h) * filter->map_w * 2);
if (filter->background)
g_free (filter->background);
filter->background = g_new0 (gint16, width * height);
filter->background = g_new0 (gint16, width * (height + 1));
if (filter->diff)
g_free (filter->diff);
filter->diff = g_new0 (guint8, width * height);
filter->diff = g_new0 (guint8, width * (height + 1));
GST_OBJECT_UNLOCK (filter);
return TRUE;
@ -650,6 +640,7 @@ gst_rippletv_init (GstRippleTV * filter)
{
filter->mode = DEFAULT_MODE;
/* FIXME: remove this when memory corruption after resizes are fixed */
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

3
gst/effectv/gstshagadelic.c
View file

@ -289,7 +289,4 @@ gst_shagadelictv_init (GstShagadelicTV * filter)
{
filter->ripple = NULL;
filter->spiral = NULL;
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

3
gst/effectv/gststreak.c
View file

@ -296,7 +296,4 @@ static void
gst_streaktv_init (GstStreakTV * filter)
{
filter->feedback = DEFAULT_FEEDBACK;
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

3
gst/effectv/gstvertigo.c
View file

@ -351,7 +351,4 @@ gst_vertigotv_init (GstVertigoTV * filter)
filter->phase = 0.0;
filter->phase_increment = 0.02;
filter->zoomrate = 1.01;
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
}

7
gst/effectv/gstwarp.c
View file

@ -133,11 +133,7 @@ initDistTable (GstWarpTV * filter, gint width, gint height)
{
gint32 halfw, halfh, *distptr;
gint x, y;
#ifdef PS2
float m;
#else
float m;
#endif
halfw = width >> 1;
halfh = height >> 1;
@ -304,6 +300,5 @@ gst_warptv_class_init (GstWarpTVClass * klass)
static void
gst_warptv_init (GstWarpTV * warptv)
{
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (warptv));
gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (warptv));
/* nothing to do */
}

14
gst/flv/gstflvdemux.c
View file

@ -781,10 +781,10 @@ gst_flv_demux_push_tags (GstFlvDemux * demux)
}
static void
gst_flv_demux_update_resync (GstFlvDemux * demux, guint32 pts, guint32 * last,
GstClockTime * offset)
gst_flv_demux_update_resync (GstFlvDemux * demux, guint32 pts, gboolean discont,
guint32 * last, GstClockTime * offset)
{
if (ABS (pts - *last) >= RESYNC_THRESHOLD) {
if (!discont && ABS (pts - *last) >= RESYNC_THRESHOLD) {
/* Theoretically, we should use substract the duration of the last buffer,
but this demuxer sends no durations on buffers, not sure if it cannot
know, or just does not care to calculate. */
@ -992,8 +992,8 @@ gst_flv_demux_parse_tag_audio (GstFlvDemux * demux, GstBuffer * buffer)
}
/* detect (and deem to be resyncs) large pts gaps */
gst_flv_demux_update_resync (demux, pts, &demux->last_audio_pts,
&demux->audio_time_offset);
gst_flv_demux_update_resync (demux, pts, demux->audio_need_discont,
&demux->last_audio_pts, &demux->audio_time_offset);
/* Fill buffer with data */
GST_BUFFER_TIMESTAMP (outbuf) = pts * GST_MSECOND + demux->audio_time_offset;
@ -1369,8 +1369,8 @@ gst_flv_demux_parse_tag_video (GstFlvDemux * demux, GstBuffer * buffer)
}
/* detect (and deem to be resyncs) large pts gaps */
gst_flv_demux_update_resync (demux, pts, &demux->last_video_pts,
&demux->video_time_offset);
gst_flv_demux_update_resync (demux, pts, demux->video_need_discont,
&demux->last_video_pts, &demux->video_time_offset);
/* Fill buffer with data */
GST_BUFFER_TIMESTAMP (outbuf) = pts * GST_MSECOND + demux->video_time_offset;

2
gst/isomp4/gstqtmuxmap.c
View file

@ -199,7 +199,7 @@ GstQTMuxFormatProp gst_qt_mux_format_list[] = {
"ismlmux",
"ISML",
"GstISMLMux",
GST_STATIC_CAPS ("video/quicktime, variant = (string) iso"),
GST_STATIC_CAPS ("video/quicktime, variant = (string) iso-fragmented"),
GST_STATIC_CAPS (MPEG4V_CAPS "; " H264_CAPS),
GST_STATIC_CAPS (MP3_CAPS "; " AAC_CAPS)
}

1
gst/multifile/gstmultifilesink.c
View file

@ -594,6 +594,7 @@ gst_multi_file_sink_ensure_max_files (GstMultiFileSink * multifilesink)
multifilesink->n_files >= multifilesink->max_files) {
filename = multifilesink->files->data;
g_remove (filename);
g_free (filename);
multifilesink->files = g_slist_delete_link (multifilesink->files,
multifilesink->files);
multifilesink->n_files -= 1;

3
gst/wavparse/gstwavparse.c
View file

@ -1164,7 +1164,8 @@ gst_wavparse_stream_headers (GstWavParse * wav)
if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
tag == GST_RIFF_TAG_LIST) {
tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
tag == GST_RIFF_TAG_IDVX) {
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
GST_FOURCC_ARGS (tag));
gst_buffer_unref (buf);

13
tests/check/elements/multifile.c
View file

@ -40,19 +40,19 @@ run_pipeline (GstElement * pipeline)
gst_element_set_state (pipeline, GST_STATE_NULL);
}
#if !GLIB_CHECK_VERSION(2,26,0)
static gchar *
g_mkdtemp (const gchar * template)
g_mkdtemp (gchar * template)
{
gchar *s;
gchar *tmpdir;
s = g_strdup (template);
tmpdir = mkdtemp (s);
tmpdir = mkdtemp (template);
if (tmpdir == NULL) {
g_free (s);
g_free (template);
}
return tmpdir;
}
#endif
GST_START_TEST (test_multifilesink_key_frame)
{
@ -93,7 +93,6 @@ GST_START_TEST (test_multifilesink_key_frame)
g_free (mfs_pattern);
g_free (my_tmpdir);
g_free (template);
}
GST_END_TEST;
@ -144,7 +143,6 @@ GST_START_TEST (test_multifilesink_max_files)
g_free (mfs_pattern);
g_free (my_tmpdir);
g_free (template);
}
GST_END_TEST;
@ -202,7 +200,6 @@ GST_START_TEST (test_multifilesrc)
g_free (mfs_pattern);
g_free (my_tmpdir);
g_free (template);
}
GST_END_TEST;