Commit graph

4698 commits

Author SHA1 Message Date
Tim-Philipp Müller
a62c7bd54c Fix use of undeclared core debug category symbols
libgstreamer currently exports some debug category
symbols GST_CAT_*, but those are not declared in any
public headers.

Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN()
to declare and use those, but that's just not right at
all, and it won't work on Windows with MSVC. Instead look
up the categories via the API.
2016-02-20 11:31:43 +00:00
Tim-Philipp Müller
ddfe7a2808 win32: remove outdated build cruft
This hasn't been touched for generations, doesn't work,
and is just causing confusion. We also don't want to
maintain these files manually.
2016-02-20 10:05:17 +00:00
Havard Graff
1cb19d1146 rtp: build audio library before rtp
Because audio-enumtypes.h needs to be available for
gstrtpbaseaudiopayload.c

https://bugzilla.gnome.org/show_bug.cgi?id=761949
2016-02-16 17:42:44 +02:00
Stian Selnes
85f297d648 videoencoder: Fix leak when pre_push does not return OK
https://bugzilla.gnome.org/show_bug.cgi?id=761951
2016-02-13 10:09:45 -03:00
Wim Taymans
5cef3f31ad audio-converter: make a copy if we can't write in unpack
If we don't have writable memory, make sure to make a copy of the input
samples into a temporary (writable) buffer, even if we are dealing with
a native intermediate format that we don't need to call the unpack
function for.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655
2016-02-10 12:51:23 +01:00
HoonHee Lee
dfa2f49523 audio/videodecoder: Minor cleanup of last commit
https://bugzilla.gnome.org/show_bug.cgi?id=761218
2016-01-28 13:21:49 +01:00
HoonHee Lee
15df3c812b audio/videodecoder: use gst_pad_peer_query_caps to make output caps
gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
In that case, use gst_pad_peer_query_caps() with template caps as filter
to have negotiated output caps properly before forwarding GAP event.

https://bugzilla.gnome.org/show_bug.cgi?id=761218
2016-01-28 11:34:22 +01:00
Aurélien Zanelli
43db5e5ebd video-format: fix GstVideoFormatInfo documentation warnings
Add missing ':' to tile_ws and tile_hs fields documentation to avoid
bad render of these two fields, mark reserved bytes as private to hide
field and avoid gtkdoc warning and add parameters description to
documented macro to avoid gtkdoc warnings.

https://bugzilla.gnome.org/show_bug.cgi?id=761132
2016-01-26 14:48:19 -05:00
Wim Taymans
03566e5002 audio-converter: add reset function 2016-01-26 17:19:34 +01:00
Wim Taymans
2d971df593 audio-converter: handle NULL input
Allow NULL as input to mean silence samples.
2016-01-26 17:19:34 +01:00
Wim Taymans
6050509b65 audio-converter: improve _update_config
Allow NULL config to keep the existing parameters.
Fix the docs.
2016-01-26 17:19:34 +01:00
Wim Taymans
0f757bc23c audio-converter: audio-converter: make some optimized functions
Make optimized functions for generic and passthrough conversion.
2016-01-26 17:19:34 +01:00
Wim Taymans
cde091ae81 audio-quantize: add _reset function
Add a reset function that clears any history.
2016-01-26 16:45:44 +01:00
Wim Taymans
3674742957 audio-converter: ensure correct alignment of samples
Make sure that the data we allocate for our temporary buffers is
properly aligned.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938
2016-01-21 16:27:50 +01:00
Wim Taymans
c99caa6b37 video-color: add Adobe RGB primaries and transfer function 2016-01-21 10:45:40 +01:00
Wim Taymans
773e2476e6 video-info: enfore RGB matrix for RGB formats
In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
RGB formats and warn when the GstVideoInfo colorimetry is wrong.
In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
for RGB formats and warn about inconsistent caps.

See https://bugzilla.gnome.org/show_bug.cgi?id=759624
2016-01-20 10:19:34 +01:00
Wim Taymans
83fe1c7705 video-converter: ignore matrix for RGB formats
For RGB formats, the matrix in the colorimetry (conversion from YUV to
RGB) is irrelevant and we should ignore it and assume the identity
transform for everything we do.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624
2016-01-20 10:02:20 +01:00
Thibault Saunier
1bf18f6f44 videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
It was never actually supported or used

https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-20 09:51:09 +02:00
Thibault Saunier
7d35a07f1a Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
This reverts commit 63517d0ed3.

It was wrong ref counting wise and we decided to deprecated DROPPED
return value

https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-20 09:50:17 +02:00
Arun Raghavan
91084f49c0 appsrc: Minor documentation cleanup 2016-01-19 12:56:20 +05:30
Evan Callaway
a1f789770d Add WAIT_ON_EOS flag to gstappsink.
If set, an appsink that receives an EOS will wait until all of its buffers have been processed before continuing.

https://bugzilla.gnome.org/show_bug.cgi?id=756187
2016-01-18 13:23:55 +02:00
Sebastian Dröge
761142e15a audioencoder: Add note to the documentation about various settings being reset before set_format()
It's quite unexpected behaviour that various subclass settings are just
reset before set_format(). Unfortunately changing this now has the risk
of breaking existing code but we should reconsider this for 2.0.
2016-01-16 11:05:13 +01:00
Thibault Saunier
63517d0ed3 videoencoder: Release video frame when ->handle return ERROR or DROPPED
https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-16 08:53:35 +01:00
Wim Taymans
1b412a523d audio-channel-mixer: round before truncating
Round the result before truncating for int channel mixing.
2016-01-12 15:56:36 +01:00
Wim Taymans
ef3844cf6f audio-converter: Avoid conversion when possible
When the input and output formats are the same and in a possible
intermediate format, avoid unpack and pack.
Never do passthrough channel mixing.
Only do dithering and noise shaping in S32 format
2016-01-12 15:27:16 +01:00
Wim Taymans
4d47d43a13 audio-channel-mixer: add more formats
Add support for float and int16 mixing
Remove in-place processing, this simplifies things as we won't be using it.
Don't do clipping for float audio formats
2016-01-12 11:43:20 +01:00
Wim Taymans
8a8b12189e audio-converter: improve processing loop
Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
2016-01-12 11:37:17 +01:00
Wim Taymans
85afad72ec audio-converter: small API tweaks
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
2016-01-08 17:34:50 +01:00
Wim Taymans
7f49b946cc audio-converter: prepare API for rate changes
Use the update function to update the sample rates along with the config
once we implement resampling.
2016-01-08 17:28:31 +01:00
Wim Taymans
980163457e audio-convert: simplify API
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
2016-01-08 17:19:58 +01:00
Sebastian Dröge
0da2709d0c audio/video: Use G_GNUC_INTERNAL for internal functions 2016-01-08 17:50:50 +02:00
Wim Taymans
40f4c5e352 audio: GstAudioChannelMix -> GstAudioChannelMixer
Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
looks better and to avoid a conflict with a library in -bad.
2016-01-08 16:41:17 +01:00
Aurélien Zanelli
c119715e25 videopool: store videoinfo after choosing the biggest buffer size
Otherwise, pool could be negotiated with a size which will be different
from the one used in allocation which is the GstVideoInfo.

https://bugzilla.gnome.org/show_bug.cgi?id=760222
2016-01-06 11:29:42 -05:00
Tim-Philipp Müller
7ffd9ce291 docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
gtk-doc can handle static inline functions just fine these days,
there's no need for this stuff any more.
2016-01-03 17:21:18 +00:00
Sebastian Dröge
4d1726fddd riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
Apparently this #define is unused.
2016-01-03 10:33:53 +02:00
Stefan Sauer
f4ae53144e riff-ids: remove trailing whitespace 2016-01-02 23:29:43 +01:00
Stefan Sauer
adb24a54ca riff-ids: fix two swapped ids
For these fourcc ids the name and value is swapped. This was causing a warning
when registering the avi ids.
2016-01-02 23:29:43 +01:00
Sebastian Dröge
81cfb23945 sdp: Also reorder SUBDIRS to try even harder to build the RTP library first 2015-12-31 20:43:28 +02:00
Sebastian Dröge
bbd82057ab sdp: The SDP library depends on the RTP library now and is not independent anymore
Fix up the build dependencies.
2015-12-31 20:41:38 +02:00
Hyunjun Ko
682b523652 sdp: add helper fuctions from/to sdp from/to caps
<gstsdpmessage.h>
GstCaps*       gst_sdp_media_get_caps_from_media   (const GstSDPMedia *media, gint pt);
GstSDPResult   gst_sdp_media_set_media_from_caps   (const GstCaps* caps, GstSDPMedia *media);
gchar *        gst_sdp_make_keymgmt                (const gchar *uri, const gchar *base64);
GstSDPResult   gst_sdp_message_attributes_to_caps  (GstSDPMessage *msg, GstCaps *caps);
GstSDPResult   gst_sdp_media_attributes_to_caps    (GstSDPMedia *media, GstCaps *caps);

<gstmikey.h>
GstMIKEYMessage * gst_mikey_message_new_from_caps  (GstCaps *caps);
gchar *           gst_mikey_message_base64_encode  (GstMIKEYMessage* msg);

https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:11:57 +02:00
Sebastian Dröge
43655580e7 encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE 2015-12-29 17:56:21 +02:00
Sebastian Dröge
f31240a765 encoding-profile: Don't use preset_name string after free
When we run the loop for another time and do not have a preset name, we would
try to print the preset name of a previous iteration that is already freed.

Also move some other variables into the block where they are actually used
to prevent similar mistakes in the future.

CID 1346536
2015-12-29 17:55:23 +02:00
Stefan Sauer
7bbfa39ada audioconvert: fix passthrough operation
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.

Fixes #759890
2015-12-29 14:40:32 +01:00
Reynaldo H. Verdejo Pinochet
e61f5b2138 videoblend: special case 1x1 src dims on increment computation
Fix crash with 1x1 overlay pixmap

https://bugzilla.gnome.org/show_bug.cgi?id=757290
2015-12-28 14:16:41 -08:00
Stefan Sauer
0bd3f818bb audio-converter: code cleanup
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
2015-12-27 19:25:20 +01:00
Sebastian Dröge
3459bd6854 audio: Fix some documentation warnings
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
2015-12-26 09:43:56 +01:00
Sebastian Dröge
3ba59f0b62 videoaffinetransformmeta: Add (transfer none) annotation for return value 2015-12-26 09:43:51 +01:00
Sebastian Dröge
7fddeaa878 pbutils: Link to libgstbase for bytewriter and adapter 2015-12-24 13:59:52 +01:00
Thibault Saunier
512ac3ea72 encodebin: Implement an encoding profile serialization format
https://bugzilla.gnome.org/show_bug.cgi?id=759356
2015-12-24 09:52:53 +01:00
Kazunori Kobayashi
d43f1b2a5a appsrc: Clear is_eos flag when receiving the flush-stop event
The EOS event can be propagated to the downstream elements when
is_eos flag remains set even after leaving the flushing state.
This fix allows this element to normally restart the streaming
after receiving the flush event by clearing the is_eos flag.

https://bugzilla.gnome.org/show_bug.cgi?id=759110
2015-12-19 11:35:39 +01:00
Wim Taymans
08734e7598 audio-converter: rework the main processing loop
Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
2015-12-16 11:13:15 +01:00
Xavier Claessens
429860e51f base: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-14 13:39:43 -05:00
Sebastian Dröge
b0c834df1b rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes 2015-12-14 19:03:33 +01:00
Evan Callaway
5ac65d9e3a rtspconnection: Use relative URI for non-proxy tunneled requests
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.

https://bugzilla.gnome.org/show_bug.cgi?id=758922
2015-12-14 18:21:10 +01:00
Evan Callaway
65c7bd7a2c rtspconnection: Support authentication during tunneling setup
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled.  The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.

The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:00:45 +01:00
Sebastian Dröge
d6be67265f rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
CID 1139615
2015-12-14 13:11:21 +01:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
2015-12-14 09:16:08 +01:00
Wim Taymans
aec17c63fd audio-converter: improve API for non-interleaved formats
Make it possible to pass an array of sample blocks when dealing with
non-interleaved formats.
2015-12-14 09:16:08 +01:00
Luis de Bethencourt
055ed65d92 riff: add FourCC aliases
Support media using the aliases defined in http://www.fourcc.org/ that are
exact duplicates of already known codes.
2015-12-12 20:22:44 +00:00
Luis de Bethencourt
98e93ec5ee riff: use defined FourCC
Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h,
like gst_riff_create_audio_caps() does.
2015-12-12 20:22:09 +00:00
Julien Isorce
4f396ae61c videodecoder: add some debug around pool negotiation
It lets us know easily which pool is activated or
inactivated during the negotiation.

https://bugzilla.gnome.org/show_bug.cgi?id=720597
2015-12-11 14:55:46 +00:00
Song Bing
a476145797 video/convertframe: Add crop meta support via videocrop
https://bugzilla.gnome.org/show_bug.cgi?id=759329
2015-12-11 16:14:59 +02:00
Tim-Philipp Müller
f0db396e63 rtpbasedepay: when setting discont flag make sure rtpbuffer is current
Depayloaders will look at rtpbuffer->buffer for the discont flag.
When we set the discont flag on a buffer in the rtp base depayloader
and we have to make the buffer writable, make sure the rtpbuffer
actually contains the newly-flagged buffer, not the original input
buffer. This was introduced with the addition of the process_rtp_packet
vfunc, but would only trigger if the input buffer wasn't flagged
already and was not writable already.
2015-12-11 11:06:35 +00:00
Tim-Philipp Müller
86350ff8b7 rtpbasedepay: fix possible refcounting issue when detecting a discont
When we detect a discont and the input buffer isn't already flagged
as discont, handle_buffer() does a gst_buffer_make_writable() on the
input buffer in order to set the flag. This assumed it had ownership
of the input buffer though, which it didn't. This would still work
fine in most scenarios, but could lead to crashes or mini object
unref criticals in some cases when a discont is detected, e.g. when
using pcapparse in front of a depayloader. This problem was
introduced in bc14cdf529.
2015-12-11 10:38:14 +00:00
Wim Taymans
5e55968546 audio-convert: improve converter API
Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
2015-12-09 17:16:26 +01:00
Philippe Normand
872f40d7d9 appsrc: duration query support based on the size property
https://bugzilla.gnome.org/show_bug.cgi?id=759126
2015-12-08 12:42:46 +02:00
Edward Hervey
d34aaf9e9b videodecoder: Avoid pushing buffers before segment start
In the case where the stream doesn't have a framerate set and the frames
don't have a duration set, we still want to use the clipping path to
make sure we don't push buffers outside of the segment.

The problem was the previous iteration was setting a duration of 2s, which
meant that any buffer which was less than 2s before the segment start would
end up getting pushed.

Instead, use a saner 40ms (25fps single frame duration) to figure out whether
the frame could be within the segment or not
2015-12-03 16:42:50 +01:00
Reynaldo H. Verdejo Pinochet
4ed7b0a0e6 Drop usage of deprecated g-ir-scanner --strip-prefix flag 2015-12-02 20:19:43 -08:00
Edward Hervey
f9b9472ad4 discoverer: Also consider XSUB as a subtitle format 2015-12-02 16:37:50 +01:00
Edward Hervey
817c780380 pbutils: Add description for XSUB subpicture format 2015-12-02 16:37:50 +01:00
Edward Hervey
27f2328348 riff: 'DXSA' is the same as 'DXSB'
Which is subpicture/x-xsub
2015-12-02 16:37:50 +01:00
Evan Callaway
e47643122c rtspconnection: Update capitalization of x-sessioncookie
Some servers incorrectly parse header names with strict case-sensitivity.  For
compatibility with these systems change X-Sessioncookie to x-sessioncookie.

https://bugzilla.gnome.org/show_bug.cgi?id=758921
2015-12-02 16:29:53 +02:00
Sebastian Dröge
2f3eb47a95 audiobasesrc: Post latency message on the bus after set_caps()
The latency is only known once the caps are known, and might change
whenever the caps are changing.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
2015-12-01 19:58:25 +02:00
Michael Olbrich
43155807cd audiobasesink: Post latency message on the bus after set_caps()
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.

Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
2015-12-01 19:58:25 +02:00
Luis de Bethencourt
2a70c86e85 codec-utils: accept wrong version field in OpusHead header
Some Opus files found on the wild have 0 in the version field of the
OpusHead header, instead of the correct value of 1. The files still
play, don't make this error fatal.

https://bugzilla.gnome.org/show_bug.cgi?id=758754
2015-12-01 15:47:35 +00:00
William Manley
aae0dc37c9 allocators: add debug category for fd memory and allocator
Debugging can now be viewed by setting GST_DEBUG=fdmemory:9

https://bugzilla.gnome.org/show_bug.cgi?id=758744
2015-11-27 15:33:47 +00:00
Ravi Kiran K N
df5725e683 id3v2frames: Handle private frames
Handle PRIV ID3 tag having owner information (string)
and binary data, add to tag messages list.

https://bugzilla.gnome.org/show_bug.cgi?id=730926
2015-11-20 20:20:18 +00:00
Tim-Philipp Müller
93a92d7f70 tags: id3: make sure to register private-id3v2-frame tag before using it 2015-11-20 19:15:22 +00:00
Ognyan Tonchev
7a702df863 rtspconnection: Add support for parsing custom headers
https://bugzilla.gnome.org/show_bug.cgi?id=758235
2015-11-18 00:15:32 +00:00
Reynaldo H. Verdejo Pinochet
0c95b0a738 Remove unnecessary NULL checks before g_free()
g_free() is NULL-safe
2015-11-17 14:50:27 -08:00
Luis de Bethencourt
09c881ee14 codec-utils: guint8 can't hold value over 255
channels is a guint8, so the max value is 255 and checking if it value is
> 256 will never be false.

CID 1338687, CID 1338688
2015-11-12 14:39:22 +00:00
Luis de Bethencourt
df16e8dd5a audio-converter: remove unneeded check for unsigned < 0
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.

CID 1338689
2015-11-12 14:18:30 +00:00
Vineeth TM
3f099e3c29 pbutils:encoding-target: Fix string memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=757926
2015-11-11 15:40:52 +01:00
Vineeth TM
b61e1465b7 audio-quantize: Fix dither_buffer memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=757928
2015-11-11 15:01:08 +01:00
Matthew Waters
0b98ed32ce videometa: add GstVideoAffineTransformationMeta
Adds a simple 4x4 affine transformations meta for passing arbitrary
transformations on buffers.

Based on patch by Matthieu Bouron

https://bugzilla.gnome.org/show_bug.cgi?id=731791
2015-11-11 00:19:25 +11:00
Wim Taymans
ff6d1a2a25 audio-converter: add output size argument
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
2015-11-10 09:53:59 +01:00
Thibault Saunier
629b63d1f2 discoverer: Check API arguments and assert if needed 2015-11-07 00:46:47 +01:00
Wim Taymans
30977cf1a5 audio-converter: require interleaved samples and no resampling
We can't yet do resampling or anything other than interleaved audio.
2015-11-06 18:00:41 +01:00
Wim Taymans
7abed02858 audio: update ORC dist files 2015-11-06 17:54:21 +01:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00
Wim Taymans
59db8ce542 audio-quantize: update docs
Update docs
Add another flag for the quantizer
2015-11-06 13:02:19 +01:00
Wim Taymans
dfbeb78342 audio: update orc files 2015-11-06 12:37:14 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.

API: gst_audio_channel_get_fallback_mask()
2015-11-05 12:50:18 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
2015-11-05 10:52:53 +01:00
Arnaud Vrac
dfe250d17d video: blend using OVER operation
Also support all premultiplied/non-premultiplied source/destination
configurations

https://bugzilla.gnome.org/show_bug.cgi?id=681447
2015-11-04 21:58:32 +01:00
Sebastian Dröge
bcd7b2fff2 codec-utils: Add utilities for Opus caps and the OpusHead header
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2015-11-03 20:35:33 +02:00
Sebastian Dröge
35ea6fdddf audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Tim-Philipp Müller
1f2fdd3789 audio: update disted orc backup files 2015-11-03 16:38:09 +00:00
Luis de Bethencourt
94a7f9fc4e audioclock: use GST_STIME_FORMAT for GstClockTimeDiff
GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
2015-11-03 14:08:29 +00:00
Luis de Bethencourt
227f1d1e0f videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-11-03 13:44:39 +00:00
Wim Taymans
801f7ca464 audio-format: add TRUNCATE_RANGE flag
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
2015-11-03 12:12:08 +01:00
Wim Taymans
914aa4aed1 audiopack: improve pack functions
Avoid shifts by using convh functions.
2015-11-03 12:12:08 +01:00
Luis de Bethencourt
fe62e797d5 audiobasesink: use GST_STIME_ARGS for GstClockTimeDiff
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
2015-11-02 17:35:20 +00:00
Sebastian Dröge
443171bb4c audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro 2015-11-02 17:35:45 +02:00
Luis de Bethencourt
b81b3f07ec videodecoder: subtract time difference with GST_CLOCK_DIFF
To ensure the subtraction of two GstClockTime values (which are guint64)
can be negative. Use GST_CLOCK_DIFF which returns a gint64.

CID 1338049
2015-11-02 12:09:45 +00:00
Thibault Saunier
a7123ebb58 encoding-profile: Do not force user to provide an encoding profile name
And use the profile called `default` if none provided.
2015-11-02 11:35:55 +01:00
Thibault Saunier
83fa06aab5 encoding-target: Do not unconditionally break when searching for a target
Otherwise the loop is useless!

Fixes CID 1338051
2015-11-02 11:31:34 +01:00
Sebastian Dröge
736a27fe1e audiofilter: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Sebastian Dröge
1da79c76a7 videodecoder: Print another time difference as a signed integer instead of a huge unsigned one 2015-10-29 16:52:49 +02:00
Sebastian Dröge
f17758d9e3 videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-10-29 16:01:26 +02:00
Thibault Saunier
2e20f3ba4f encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files 2015-10-28 16:07:44 +01:00
Thibault Saunier
96f69fa998 encoding-target: Allow having encoding target without a category set
There was already some code to handle that, but the support was not
complete in those code paths.
2015-10-28 16:07:44 +01:00
Thibault Saunier
0256381f6f encoding-target: Create directory before trying to save encoding targets 2015-10-28 16:07:44 +01:00
Thibault Saunier
db272cf9cb encoding-profile: Allow specifying the target category in the serialized encoding target 2015-10-28 16:07:44 +01:00
Pavel Bludov
8051434c03 video: overlay-composition: fix rectangle and composition cast macros
Closing parenthesis was missing in two cases.

https://bugzilla.gnome.org/show_bug.cgi?id=756893
2015-10-22 09:24:01 +01:00
Sebastian Dröge
5bdeed32bc rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1 2015-10-20 12:10:52 +03:00
eunhae choi
e98b96247f audiobasesink: fix issue about eos handling during flushing
If the flush-start is arrived during _eos_wait() in basesink,
the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
To resolve the overwritten issue,
the subclass doing the _eos_wait() call should return the right value.
If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
of the following state changing from PAUSED to PLAYING in basesink.

https://bugzilla.gnome.org/show_bug.cgi?id=754980
2015-10-19 12:12:12 -03:00
Sebastian Dröge
df0d5aa4d5 video: Add out annotations to the out parameters of gst_video_calculate_display_ratio()
https://bugzilla.gnome.org/show_bug.cgi?id=754567
2015-10-17 20:37:22 +03:00
Stian Selnes
0a668c1866 rtpbuffer: Add map flag to skip padding
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.

When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.

https://bugzilla.gnome.org/show_bug.cgi?id=752705
2015-10-15 22:42:37 +03:00
Sebastian Dröge
79d4765525 Revert "rtpbuffer: increase logging level when map fails"
This reverts commit e3c8a82017.

It causes too much noise in the logs.
2015-10-15 22:40:50 +03:00
Miguel París Díaz
e3c8a82017 rtpbuffer: increase logging level when map fails
https://bugzilla.gnome.org/show_bug.cgi?id=756641
2015-10-15 16:48:08 +03:00
Sebastian Dröge
b60ab758e4 Revert "audioencoder: timestamp headers same as first buffer and use duration 0"
This reverts commit dd4d6d9ed5.

It breaks ogg muxing and the vorbisenc unit test.
2015-10-12 14:02:58 +03:00
Havard Graff
dd4d6d9ed5 audioencoder: timestamp headers same as first buffer and use duration 0
https://bugzilla.gnome.org/show_bug.cgi?id=754224
2015-10-11 11:04:53 +01:00
Sebastian Dröge
4115814f36 Update GLib dependency to 2.40.0 2015-10-02 22:19:52 +03:00
Sebastian Rasmussen
042e71a117 rtpbasepayload: Implement video SDP attributes
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472
2015-10-02 17:44:14 +03:00
Vineeth TM
8e5f7f27f5 gstreamer: base: Fix memory leaks when context parse fails.
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.

And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.

https://bugzilla.gnome.org/show_bug.cgi?id=753852
2015-10-02 17:26:27 +03:00
Luis de Bethencourt
e90464d2f9 visual: make private all variable subclasses don't need
Subclasses don't need access to all variables. Making them private.

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:05:58 +01:00
Luis de Bethencourt
8ae0fd3990 visual: merge audiovisalizer base classes
Move the audiovisualizer base class to pbutils, so it can be used by plugins
from other modules

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:05:58 +01:00
Perry Hung
c5c2466d36 app: pass PKG_CONFIG_PATH for gir files for libgstapp as well
gir include search directories should respect PKG_CONFIG_PATH,
just like we do everywhere else. Makes g-i pick up the right
paths when using ./configure --with-pkg-config-path=

https://bugzilla.gnome.org/show_bug.cgi?id=755494
2015-09-29 13:10:35 +01:00
Sebastian Dröge
e49c492a53 Release 1.6.0 2015-09-25 23:15:20 +02:00
Sebastian Dröge
8239da2311 video-dither: Use saturated add when adding ordered dither for > 8 bit targets
Otherwise our 16 bit integers are going to overflow in intermediate
calculations, causing video to become mostly black.

https://bugzilla.gnome.org/show_bug.cgi?id=755392
2015-09-24 23:01:58 +02:00
Sebastian Dröge
1fb85733f5 video-frame: Fix gst_video_frame_copy() for formats with pstride==0
v210, UYVP and IYU1 are complex formats for which pixel stride does not really
have a meaning. If we copy width*pstride bytes per line, it's not going to do
the right thing. As a fallback, copy stride bytes per line. This might copy
uninitialized bytes at the end of each line, but at least copies the frame.

https://bugzilla.gnome.org/show_bug.cgi?id=755392
2015-09-24 17:39:43 +02:00
Aurélien Zanelli
94e0279c44 fdmemory: remove 'allow-none' annotation in gst_fd_allocator_alloc() doc
gst_fd_allocator_alloc() ensure that passed allocator is a fd memory
allocator, so that we can't pass NULL allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=754833
2015-09-23 10:01:41 -04:00
Aurélien Zanelli
4559feb69e dmabuf: remove 'allow-none' annotation in gst_dmabuf_allocator_alloc() doc
gst_dmabuf_allocator_alloc() ensure that passed allocator is a DMABuf
allocator, so that we can't pass NULL allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=754833
2015-09-23 10:01:41 -04:00
Christophe Fergeau
f096f1f9d3 appsink: Fix 'steaming' typo in API doc
There are several occurrences of 'steaming' where 'streaming' was meant.
2015-09-18 17:48:49 +02:00
Vineeth T M
060f0c21f2 audiosink, multisocketsink: Fix error leak during failures
https://bugzilla.gnome.org/show_bug.cgi?id=755143
2015-09-17 11:59:35 +02:00
Tim-Philipp Müller
a7bcdc3608 appsink: minor docs fix 2015-09-15 16:32:45 +01:00
Tim-Philipp Müller
8b96b52a62 videopool: ensure allocation alignment is consistent with video alignment requirements
Make sure GstAllocationParams alignment is not less than
any alignment requirement specified via GstVideoAlignment.

https://bugzilla.gnome.org/show_bug.cgi?id=754120
2015-09-15 10:20:37 +01:00
Thiago Santos
df931fb0b8 video-format: add missing alpha flag for some formats
Some formats didn't have the alpha flag marked, use the correct
macro so they get it right.

https://bugzilla.gnome.org/show_bug.cgi?id=754808
2015-09-10 07:28:21 -03:00
Sebastian Dröge
8613525301 appsrc: Always take the mutex before flushing the queue
Otherwise the application might push new buffers into the queue while we're
flushing, potentially causing the GQueue data structure to become inconsistent
and causing crashes soon after.

https://bugzilla.gnome.org/show_bug.cgi?id=754597
2015-09-09 12:33:02 +03:00
Vikram Fugro
bbe967a278 appsrc: retain the latest caps in queue when flushing
- Retain the latest caps in the internal queue, when
  flushing.
- Add a unit test case for the same.

https://bugzilla.gnome.org/show_bug.cgi?id=754597
2015-09-09 12:27:40 +03:00
Tim-Philipp Müller
86a1092e59 video: improve GstVideoFormatUnpack docs
https://bugzilla.gnome.org/show_bug.cgi?id=754650
2015-09-07 00:19:09 +01:00
Kouhei Sutou
3e41e784be libs: Fix build error on MinGW where "%ll" is not available
"ll" isn't available on MinGW. We can use G_GINT64_MODIFIER for portable
64bit size data modifier.

https://bugzilla.gnome.org/show_bug.cgi?id=754630
2015-09-06 18:32:16 +03:00
Havard Graff
181845ec1b libs: build rtp after audio
The dependency setup does not seem to work for all systems,
causing the build to fail with:

gstrtpbaseaudiopayload.c:65:0:
fatal error: gst/audio/audio-enumtypes.h: No such file or directory

My setup:
gcc (Ubuntu 4.8.4-2ubuntu1~14.04) 4.8.4
autoconf (GNU Autoconf) 2.69
automake (GNU automake) 1.14.1
libtool (GNU libtool) 2.4.2

https://bugzilla.gnome.org/show_bug.cgi?id=754344
2015-09-04 09:53:42 +01:00
Tim-Philipp Müller
c8ce5765a0 libs: rtp is no longer an independent subdir
https://bugzilla.gnome.org/show_bug.cgi?id=754344
2015-09-04 09:53:42 +01:00
Philippe Normand
9361f7c576 appsrc: remove duplicate get_size vfunc assignment 2015-08-31 12:11:10 +03:00
George Kiagiadakis
7860060536 allocators: include gstfdmemory.h in the main library header, allocators.h 2015-08-29 21:38:52 +02:00
Tim-Philipp Müller
7dac2e1eb1 audiobasesink: fix misleading error message debug detail
https://bugzilla.gnome.org/show_bug.cgi?id=754260
2015-08-29 10:44:28 +01:00
Carlos Rafael Giani
c95d809a96 audiobasesink: Fix incorrect/missing custom slaving method documentation
https://bugzilla.gnome.org/show_bug.cgi?id=754199
2015-08-28 10:13:44 +03:00
Sebastian Dröge
64301f1005 video(en|de)coder: Return TRUE when we consumed a tag event without creating a new event
Fixes spurious flow errors that especially break gst-validate.
2015-08-18 16:28:42 +03:00
Sebastian Dröge
6cda5d3494 audiodecoder: If there are no tags, don't try to do event handling on a NULL event
Fixes some crashes.
2015-08-18 16:01:28 +03:00
Sebastian Dröge
2de91c32e4 audioencoder: If there are no tags, don't try to do event handling on a NULL event
Fixes some crashes.
2015-08-18 15:58:57 +03:00
Tim-Philipp Müller
c21a972421 videoencoder: fix tag handling
Merge upstream tags with encoder tags and update whenever
any of those changes.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-18 12:44:04 +01:00
Tim-Philipp Müller
4c00709e22 audioencoder: fix tag handling
Merge upstream tags with encoder tags and update whenever
any of those changes.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-18 11:46:21 +01:00
Vineeth TM
fd7724b6e1 pbutils: discoverer: Set GError when NULL info is being returned.
When discovering the URI, if info is NULL, then instead of just returning NULL,
set the GError, so the error can be printed and notified.

https://bugzilla.gnome.org/show_bug.cgi?id=753701
2015-08-18 12:13:41 +03:00
Tim-Philipp Müller
135cad5b6e audiodecoder: try harder to avoid sending unnecessary tag updates 2015-08-16 18:28:09 +01:00
Tim-Philipp Müller
30b7cc6670 videodecoder: fix tag handling
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.

So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.

This slightly changes the behaviour of gst_video_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.

If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 18:23:07 +01:00
Tim-Philipp Müller
5ccc8432e0 audiodecoder: fix tag handling
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.

So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.

This slightly changes the behaviour of gst_audio_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.

If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 16:34:31 +01:00
Thiago Santos
5a0bee3f13 audioencoder: add src and sink query methods
Allows subclasses to do their own handling of GstQuery and still
chain up to the parent class to handle the ones that they don't want
to handle
2015-08-16 08:14:35 -03:00
Thiago Santos
d19f347223 Revert "audiodecoder: expose default query handling function"
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer

This reverts commit 8a64592481.
2015-08-15 13:44:35 -03:00
Thiago Santos
c63db0e104 Revert "videodecoder: expose default query handling function"
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer

This reverts commit ea9b6a7e3c.
2015-08-15 13:44:35 -03:00
Thiago Santos
33b1da7b80 audiodecoder: add option to use default pad accept-caps handling
Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary

API: gst_audio_decoder_set_use_default_pad_acceptcaps
2015-08-15 13:44:34 -03:00
Thiago Santos
4982188cf7 videodecoder: add option to use default pad accept-caps handling
Add gst_video_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary

API: gst_video_decoder_set_use_default_pad_acceptcaps
2015-08-15 13:44:34 -03:00
Jan Schmidt
95eb641821 rtpbasedepayload: Make stats creation threadsafe, fix a CRITICAL
Use the object lock to protect the internal segment when updating
against access from getting the stats property.

Fix a critical in gst-inspect or when retrieving the stats
before any segment has arrived by checking whether the
segment has been initted..
2015-08-15 23:37:26 +10:00
George Kiagiadakis
897371ac4f appsink: unref the preroll buffer and cleanup the segments on stop()
Just for consistency. No need to keep data around.
2015-08-14 19:27:07 +02:00
George Kiagiadakis
c3e4d8ca6f appsink: do not update preroll_caps unless the sink is prerolling
Just for consistency with the preroll_segment
2015-08-14 19:27:07 +02:00
George Kiagiadakis
c411819452 appsink: put the correct segment in the preroll sample
last_segment is only being updated in dequeue_buffer(),
which is only called from _pull_sample(). _pull_preroll()
simply re-uses an old or dummy segment while the actual
one sits and waits in the queue.

https://bugzilla.gnome.org/show_bug.cgi?id=751147
2015-08-14 19:27:06 +02:00
Thiago Santos
ea9b6a7e3c videodecoder: expose default query handling function
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.

API: gst_video_decoder_sink_query_default

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:41:58 -03:00
Thiago Santos
8a64592481 audiodecoder: expose default query handling function
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.

API: gst_audio_decoder_sink_query_default

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:41:58 -03:00
Sebastian Dröge
8dc4bfdd0e rtp: Depend on the audio library 2015-08-11 14:10:57 +02:00
Sebastian Dröge
a0f1b964f1 rtpbaseaudiopayload: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the audio tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2015-08-11 12:46:36 +02:00
Nicolas Dufresne
6ddab6918d basedepayloader: Don't re-timestamp with running-time
There was a confusion, six depayloaders where passing through the
timestamp while the base class was re-timestamping to running
time. This inconstancy has been unnoticed has in most use cases
the incoming segment is [0, inifnity] in which case timestamps are
the same as running time. With DTS/PTS shifting added (to avoid
negative values) and pcapparse sending a different segment this
started being an issue.

https://bugzilla.gnome.org/show_bug.cgi?id=753037
2015-08-10 13:26:20 -04:00
Thiago Santos
d259a233a5 videoencoder: remove empty line to make g-i-scanner happy
gstvideoencoder.h:228: Warning: GstVideo: "@transform_meta"
parameter unexpected at this location:
 * @transform_meta: Optional. Transform the metadata on ...
2015-08-10 10:40:19 -03:00
Thiago Santos
a1c3276159 videodecoder: documentation cleanup
Remove some whitespace and break lines longer than 80 columns
2015-08-10 10:40:07 -03:00
Thiago Santos
e59d1308cc videodecoder: push pending events before gap
Push all pending events before pushing the gap. This ensures the
segment is pushed before the gap so it can be properly translated
to the running time

Includes unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=753360
2015-08-10 00:21:37 -03:00
Brian Peters
1273848bf5 rtpbuffer: avoid accessing NULL buffer even more
Previous commit was incompletely applied.

https://bugzilla.gnome.org/show_bug.cgi?id=753001
2015-07-30 15:16:57 +01:00
Brian Peters
04c69952c0 rtp: buffer: don't access NULL buffer pointer
unmap will set rtpbuffer->buffer to NULL, so we need to
save the pointer to access it while the RTP buffer is
unmapped.

https://bugzilla.gnome.org/show_bug.cgi?id=753001
2015-07-30 14:30:44 +01:00
Tim-Philipp Müller
232bdf1711 rtpbasedepayload: fix leaks in error code paths
This was introduced when reshuffling the buffer unmaps
in commit bc14cdf529
rtp: rtpbasedepayload: add process_rtp_packet() vfunc

Fixes make check-valgrind.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-30 12:50:56 +01:00
Nicolas Dufresne
7c638e06ff depayloader: Use input segment start
When there is no clock_base provided, the start position is
set to 0 instead of the original segment start value. This
would break synchronization if start was not 0.

https://bugzilla.gnome.org/show_bug.cgi?id=752228
2015-07-18 15:40:26 -04:00
Wim Taymans
14083178b8 video: improve logging
Add logging categories for most video objects.
Remove some useless debug lines in video-info and videotestsrc.
Add a performance debug line in the video scaler.
2015-07-15 12:47:42 +02:00
Olivier Crête
dec3fb3723 rtsp: Include generated enum types in gstrtsptransport.h
GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not
including the generated file makes older gst-p-good fail to build,
so it constitues an API break.
2015-07-14 17:18:12 -04:00
Tim-Philipp Müller
bc14cdf529 rtp: rtpbasedepayload: add process_rtp_packet() vfunc
Add process_rtp_packet() vfunc that works just like the
existing process() vfunc only that it takes the GstRTPBuffer
that the base class has already mapped (with MAP_READ),
which means that the subclass doesn't have to map it again,
which allows more performant processing of input buffers
for most RTP depayloaders.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:29:29 +01:00
Nicolas Dufresne
8be88f0364 doc/build: Fix doc typos
This minor update should workaround a build system bug. While the
makefile has been updated to generate more enum type, there is nothing
that updates the header and would lead to the generated code to be
produced again. This minor doc fix should ensure no one get a build with
missing symbols.
2015-07-09 10:47:20 -04:00
Tim-Philipp Müller
3d7a92b452 rtpbasedepayload: fix typo in comment 2015-07-07 19:56:52 +01:00
Tim-Philipp Müller
fbf2773b2e rtpbasepayload: fix possible segment event leak
Need to clear it when shutting down, not when starting up.
Fixes leak in rtp-payloading unit test.
2015-07-07 15:05:59 +01:00
Hyunjun Ko
9dc463346e video/audio meta: transform_func: return FALSE if not supported or failed
https://bugzilla.gnome.org/show_bug.cgi?id=751778
2015-07-07 16:41:23 +03:00
Sebastian Dröge
a1f8fb32cb video: Add some more GTypes for enums 2015-07-07 14:32:56 +03:00
Tobias Mueller
190c7f1928 GstVideoScaler: Initialised scaling functions to get rid of compiler messages
E.g.

video-scaler.c: In function 'gst_video_scaler_horizontal':
video-scaler.c:1332:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
   func (scale, src, dest, dest_offset, width, n_elems);
   ^

video-scaler.c: In function 'gst_video_scaler_vertical':
video-scaler.c:1373:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
   func (scale, src_lines, dest, dest_offset, width, n_elems);
   ^

GCC's analyses seem to be correct, for the simple fact that if you pass
get_functions a known format, but no hscale or vscale, it'll return
True without having done anything.
Some callers check for the scale values to be not NULL, but then
hscale->resampler.max_taps could return 0.
A different approach to the one presented in this patch is to check
for those max_taps, too, before calling get_functions.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=752051
2015-07-07 13:06:34 +02:00
Sebastian Dröge
b495dfa5e6 {audio,video}info: Add GST_TYPE_{AUDIO,VIDEO}_INFO macros 2015-07-06 12:53:15 +03:00
Marcin Kolny
95fa0c58a2 video-info: implement GstVideoInfo as boxed type
GstVideoInfo usually is created on the stack, but boxed type can be useful
for bindings.

https://bugzilla.gnome.org/show_bug.cgi?id=752011
2015-07-06 12:52:51 +03:00
Stian Selnes
1586981b1b rtcpbuffer: Fix validation of packets with padding
The padding (if any) is included in the length of the last packet, see
RFC 3550.

Section 6.4.1:
   padding (P): 1 bit
      If the padding bit is set, this individual RTCP packet contains
      some additional padding octets at the end which are not part of
      the control information but are included in the length field. The
      last octet of the padding is a count of how many padding octets
      should be ignored, including itself (it will be a multiple of
      four).

Section A.2:
   *  The padding bit (P) should be zero for the first packet of a
      compound RTCP packet because padding should only be applied, if it
      is needed, to the last packet.

   *  The length fields of the individual RTCP packets must add up to
      the overall length of the compound RTCP packet as received.

https://bugzilla.gnome.org/show_bug.cgi?id=751883
2015-07-06 12:06:47 +03:00
Stian Selnes
008a228865 videodecoder: Fix setting default pixel-aspect-ratio
It's needed to check if pixel-aspect-ratio exists before fixating.
It does not exist if input caps is not set yet and allowed caps
does not contain pixel-aspect-ratio (e.g. when using GST_VIDEO_CAPS_MAKE)

https://bugzilla.gnome.org/show_bug.cgi?id=751932
2015-07-06 11:59:38 +03:00
Sebastian Dröge
56add20dc7 audioencoder: Don't try to get buffers from an empty adapter 2015-07-02 13:16:15 +02:00
Sebastian Dröge
f5eebb27a2 {audio,video}{en,de}oder: Also copy POOL metas and make sure to copy over metas when creating subbuffers
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
2015-07-01 10:58:07 +02:00
Sebastian Dröge
844fa28c67 audiodecoder: Add transform_meta() vfunc with default implementation
The default implementation copies all metadata without tags, and metadata
with only the audio tag. Same behaviour as in GstAudioFilter.

https://bugzilla.gnome.org/show_bug.cgi?id=742385
2015-06-30 10:40:10 +02:00
Sebastian Dröge
cc1719130b audioencoder: Add transform_meta() vfunc with default implementation
The default implementation copies all metadata without tags, and metadata
with only the audio tag. Same behaviour as in GstAudioFilter.

https://bugzilla.gnome.org/show_bug.cgi?id=742385
2015-06-30 10:40:10 +02:00
Sebastian Dröge
d2cc76b228 videodecoder: Add transform_meta() vfunc with default implementation
The default implementation copies all metadata without tags, and metadata
with only the video tag. Same behaviour as in GstVideoFilter.

This currently does not work if the ::parse() vfunc is implemented as all
metas are getting lost inside GstAdapter.

https://bugzilla.gnome.org/show_bug.cgi?id=742385
2015-06-30 10:40:09 +02:00