mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-10 11:29:55 +00:00
Code Issues 1 Projects Releases Wiki Activity

a52dec: Check that the a52_state is correctly initialized

Browse source
This commit is contained in:
Julien Isorce 2011-12-13 14:52:26 +01:00 committed by Sebastian Dröge
parent da9f20eceb
commit cdf8d0f6b2
1 changed files with 23 additions and 11 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

34
ext/a52dec/gsta52dec.c
View file

@ -21,7 +21,7 @@
* SECTION:element-a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
*
* <refsect2>
* <title>Example launch line</title>
* |[
@ -171,7 +171,7 @@ gst_a52dec_class_init (GstA52DecClass * klass)
*
* Force a particular output channel configuration from the decoder. By default,
* the channel downmix (if any) is chosen automatically based on the downstream
* capabilities of the pipeline.
* capabilities of the pipeline.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
@ -229,6 +229,9 @@ gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
a52dec->state = NULL;
a52dec->samples = NULL;
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
}
@ -390,6 +393,7 @@ gst_a52dec_push (GstA52Dec * a52dec,
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
@ -566,8 +570,8 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
}
/* If we haven't had an explicit number of channels chosen through properties
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* downstream element such as audioconvert.
*/
if (a52dec->request_channels != A52_CHANNEL) {
@ -591,13 +595,13 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
A52_3F2R | A52_LFE,
};
/* Prefer the original number of channels, but fixate to something
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
flags ? gst_a52dec_channels (flags, NULL) : 6);
gst_structure_get_int (structure, "channels", &channels);
if (channels <= 6)
if (gst_structure_get_int (structure, "channels", &channels)
&& channels <= 6)
flags = a52_channels[channels - 1];
else
flags = a52_channels[5];
@ -781,10 +785,10 @@ gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
if (!a52dec->sent_segment) {
GstSegment segment;
/* Create a basic segment. Usually, we'll get a new-segment sent by
/* Create a basic segment. Usually, we'll get a new-segment sent by
* another element that will know more information (a demuxer). If we're
* just looking at a raw AC3 stream, we won't - so we need to send one
* here, but we don't know much info, so just send a minimal TIME
* here, but we don't know much info, so just send a minimal TIME
* new-segment event
*/
gst_segment_init (&segment, GST_FORMAT_TIME);
@ -870,6 +874,12 @@ gst_a52dec_change_state (GstElement * element, GstStateChange transition)
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
a52dec->state = a52_init (klass->a52_cpuflags);
if (!a52dec->state) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("Failed to initialize a52 state"));
ret = GST_STATE_CHANGE_FAILURE;
}
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
@ -905,8 +915,10 @@ gst_a52dec_change_state (GstElement * element, GstStateChange transition)
clear_queued (a52dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
a52_free (a52dec->state);
a52dec->state = NULL;
if (a52dec->state) {
a52_free (a52dec->state);
a52dec->state = NULL;
}
break;
default:
break;