mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-10 03:19:40 +00:00
Code Issues 1 Projects Releases Wiki Activity
15745 commits 44 branches 1509 tags 459 MiB
Commit graph

38 commits

Author SHA1 Message Date
Sebastian Dröge
5b23cf694c amcaudiodec: Calculate number of samples per frame for MP3 and use that 
Some audio decoders (at least the MP3 decoder on MTK based devices) outputs
raw audio in batches of multiple audio frames. We need to handle that
properly, otherwise the base class will be kind of unhappy.
 2014-02-14 12:51:50 +01:00
Sebastian Dröge
aeeeedd7c7 amcaudiodec: Error out if we get an invalid raw audio buffer size 2014-02-14 12:51:50 +01:00
Sebastian Dröge
2476b926b1 amcaudiodec: Release buffer back to Android if allocating output fails 2014-02-14 12:51:50 +01:00
Sebastian Dröge
30d80bf658 androidmedia: Don't handle FLUSHING or NOT_LINKED as error 
And also just stop the task for FLUSHING.
 2014-01-25 17:41:18 +01:00
Andoni Morales Alastruey
6b49683447 androidmedia: fix access to invalid buffers in the decoding loop 
Flushing the decoder invalidates all buffers, so it should be done
after quiting the decoding loop. Otherwise we can jump into
"failed_release" and stop everything

https://bugzilla.gnome.org/show_bug.cgi?id=711156
 2013-10-30 16:24:17 +01:00
Sebastian Dröge
e66f097f37 androidmedia: Fix indention 2013-05-20 13:24:07 +02:00
Sebastian Dröge
a887497d18 amcaudiodec: Properly set output format instead of just setting caps 2013-04-16 16:23:46 +02:00
Sebastian Dröge
055154fe78 androidmedia: Set parent class in class_init 2013-04-16 16:17:50 +02:00
Sebastian Dröge
7832b9b43f androidmedia: Handle refcounts of pad templates correctly 
gst_pad_template_new() does not take ownership of caps anymore and
gst_element_class_add_pad_template() does take ownership of the
pad template now.
 2013-04-16 16:00:30 +02:00
Sebastian Dröge
fd8defc525 amcaudiodec: Use 1.0-style raw audio caps 2013-04-16 15:32:01 +02:00
Sebastian Dröge
75493a7731 androidmedia: Properly set subclasses metadata/pad templates in base_init 2013-04-16 15:31:34 +02:00
Sebastian Dröge
b2e5b9cb67 androidmedia: Update to new GLib thread API 2012-11-12 11:32:44 +01:00
Sebastian Dröge
079c68e4de androidmedia: Port to 1.0 2012-10-25 14:05:48 +02:00
Sebastian Dröge
31f0f163bd Try to handle format changes more gracefully 
And make stop() faster and more robust
 2012-10-15 16:28:43 +02:00
Sebastian Dröge
b0b642d8ab Add workaround for Google MP3 decoder outputting garbage in first output buffer 
And assume one decoded input frame per output buffer to fix timestamp
handling by the base class.
 2012-10-15 16:28:43 +02:00
Sebastian Dröge
2f3b2c586f Also add some more output format checks to the audio decoder 2012-10-15 16:28:42 +02:00
Sebastian Dröge
6ed3ea7cad Don't set timestamps, the baseclass handles this for us anyway 2012-10-15 16:28:42 +02:00
Sebastian Dröge
0b0255ed12 Merge structures into caps instead of appending them 2012-10-15 16:28:42 +02:00
Sebastian Dröge
9f7ed3f180 Use orc_memcpy() instead of memcpy() as it should be faster for larger memory areas 2012-10-15 16:28:42 +02:00
Sebastian Dröge
7a53743bfd Add support for audio/mpeg-L2 for the Nexus 7 2012-10-15 16:28:42 +02:00
Sebastian Dröge
cb6f703669 Only mpegversion=4 is supported for AAC 2012-10-15 16:28:41 +02:00
Sebastian Dröge
742816df69 Remove some obsolete FIXMEs 2012-10-15 16:28:41 +02:00
Sebastian Dröge
f839a615a0 Fix flushing logic to unbreak handling of flushing seeks 2012-10-15 16:28:41 +02:00
Sebastian Dröge
7dbef6415f Fix Vorbis decoding 2012-10-15 16:28:41 +02:00
Sebastian Dröge
562b033872 Wait at most 100ms for dequeueing a buffer, and retry after that 
Some codecs don't fail the dequeueing if the codec is flushed,
causing deadlocks when shutting down the element.
 2012-10-15 16:28:41 +02:00
Sebastian Dröge
9a034f2beb Release output buffers a bit earlier 2012-10-15 16:28:41 +02:00
Sebastian Dröge
4c3617f7f0 Keep another reference of the input buffer in handle_frame() 
It might be unreffed in the meantime by the other thread due
to finish_frame().
 2012-10-15 16:28:41 +02:00
Sebastian Dröge
613a75ad92 Pass -1 as number of frames to the base class 
This will automatically do the right thing if timestamps are good enough
 2012-10-15 16:28:41 +02:00
Sebastian Dröge
0143418e0c Use correct buffer variable 2012-10-15 16:28:41 +02:00
Sebastian Dröge
bb8e53d4b8 Always pass 1 as number of frames to finish_frame() 2012-10-15 16:28:41 +02:00
Sebastian Dröge
99e287b86b Correctly remember number of channels and samplerate 2012-10-15 16:28:40 +02:00
Sebastian Dröge
efac01d096 Don't unref input buffer from handle_frame(), it's owned by the baseclass 2012-10-15 16:28:40 +02:00
Sebastian Dröge
3c40a9ca56 Remove invalid GST_ERROR() call 2012-10-15 16:28:40 +02:00
Sebastian Dröge
b07cd212ac Only try to get the channel-mask if it is set 2012-10-15 16:28:40 +02:00
Sebastian Dröge
afbc23ceed Add support for FLAC 2012-10-15 16:28:40 +02:00
Sebastian Dröge
b64e57ce42 Fix src template caps 2012-10-15 16:28:40 +02:00
Sebastian Dröge
6cc535581f Fix plugin initialization 2012-10-15 16:28:40 +02:00
Sebastian Dröge
77fbaae250 Add support for audio decoders, completely untested so far 2012-10-15 16:28:40 +02:00