Commit graph

15229 commits

Author SHA1 Message Date
Vincent Penquerc'h d1ecd3cfa7 subparse: fix build with GCC 4.6.3
gstsubparse.c: In function ‘parse_subrip’:
gstsubparse.c:988:7: error: ignoring return value of ‘strtol’, declared with attribute warn_unused_result [-Werror=unused-result]
cc1: all warnings being treated as errors

https://bugzilla.gnome.org/show_bug.cgi?id=765042
2016-04-15 13:33:41 +01:00
Josep Torra 62655029c8 .gitignore: add test-resample binary 2016-04-15 13:08:38 +02:00
Aleix Conchillo Flaqué 62e0e74759 mikey: allow passing srtp or srtcp to create mikey message
Current implementation requires all srtp and srtcp parameters to be
given in the caps. MIKEY uses only one algorithm for encryption and one
for authentication so we now allow passing srtp or srtcp parameters. If
both are given srtp parametres will be preferred.

https://bugzilla.gnome.org/show_bug.cgi?id=765027
2016-04-15 12:33:43 +02:00
Julien Isorce cf63a900a9 Automatic update of common submodule
From 6f2d209 to ac2f647
2016-04-14 10:00:06 +01:00
Sebastian Dröge a82ef8983e videometa: Initialize all fields of all metas with default values
The metas are not allocated with all fields initialized to zeroes.

https://bugzilla.gnome.org/show_bug.cgi?id=764902
2016-04-13 10:10:51 +03:00
Arjen Veenhuizen c5ed98a35b videometa: Explicitly initialize GstVideoCropMeta on init
It is not allocated with all fields initialized to 0.

https://bugzilla.gnome.org/show_bug.cgi?id=764902
2016-04-13 10:10:46 +03:00
Guillaume Desmottes 6e7fa6659f alsa: properly convert position-less channels from ALSA
The only way for ALSA to expose a position-less multi channels is to
return an array full of SND_CHMAP_MONO. Converting this to a
GST_AUDIO_CHANNEL_POSITION_MONO array would be invalid as
GST_AUDIO_CHANNEL_POSITION_MONO is meant to be used only with one
channel.

Fix this by using GST_AUDIO_CHANNEL_POSITION_NONE which is meant to be
used for position-less channels.

https://bugzilla.gnome.org/show_bug.cgi?id=763799
2016-04-12 14:48:30 -04:00
Guillaume Desmottes 7c5dfd713c audioringbuffer: don't attempt to reorder position-less channels
As said in its doc GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used
for "position-less channels, e.g. from a sound card that records 1024
channels; mutually exclusive with any other channel position".

But at the moment using such positions would raise a
'g_return_if_reached' warning as gst_audio_get_channel_reorder_map()
would reject it.

Fix this by preventing any attempt to reorder in such case as that's not
what we want anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=763799
2016-04-12 14:48:30 -04:00
Guillaume Desmottes 1c56cfa144 audio: add debug output if channels mapping does not match
https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Guillaume Desmottes eef7312169 alsa: add some debugging output to alsa_detect_channels_mapping()
https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Guillaume Desmottes 3cb08304da gst-audio: add gst_audio_channel_positions_to_string()
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Guillaume Desmottes d9268a50f1 alsa: factor out alsa_detect_channels_mapping()
This code was duplicated in alsasrc and alsasink.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:34:13 -04:00
Guillaume Desmottes 592b87a463 alsa: coding style fix
Was using tabs instead of spaces.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:34:13 -04:00
Vivia Nikolaidou a0cf3b4262 fdmemory, rtpbasedepayload: Ran gst-indent
https://bugzilla.gnome.org/show_bug.cgi?id=764948
2016-04-12 17:34:18 +03:00
Vivia Nikolaidou 9d96959dde decodebin: Rename misleading variable is_parser_converter into is_parser
In that place, the variable isn't checking whether the element is a
converter, only if it is a parser.

https://bugzilla.gnome.org/show_bug.cgi?id=764948
2016-04-12 17:34:18 +03:00
Fabrice Bellet bfcd0737b7 audio: Fix a race with the audioringbuffer thread
There is a small window of time where the audio ringbuffer thread
can access the parent thread variable, before it's initialized
by the parent thread. The patch replaces this variable use by
g_thread_self().

https://bugzilla.gnome.org/show_bug.cgi?id=764865
2016-04-11 21:43:13 +10:00
Tim-Philipp Müller f05ea1e6a6 tests: libscpp: test RTP/RTCP buffer init macros with C++ compiler 2016-04-06 17:57:28 +01:00
Jan Schmidt 2dd399356f subtitleoverlay: Don't complain when stream-start is the first event.
When blocking the subtitle pad, it's expected that stream-start
is the first event, and that it can precede caps arriving on the
peer pad - in fact the caps can only have arrived on the peer
pad when it was pre-primed with sticky events previously.

Instead, just pass the stream-start and don't block, because
stream-start is sticky anyway.
2016-04-06 21:03:19 +10:00
Jan Schmidt c27df799bf subparse: WebVTT Cue identifiers are optional
Don't require a cue identifier preceding the time range line
when parsing WebVTT. We could also store the CueID, but it's
not using anywhere, so just ignore it for now.
2016-04-06 21:00:10 +10:00
Sebastian Dröge 4a9be62ec6 win32: Add new libgstaudio symbols 2016-04-05 14:26:55 +03:00
Víctor Manuel Jáquez Leal 37c4915109 libs: audio: split allocation query caps and pad caps
Since the allocation query caps contains memory size and the pad's caps
contains the display size, an audio encoder or decoder might need to allocate
a different buffer size than the size negotiated in the caps.

This patch splits this logic distinction for audiodecoder and audioencoder.

Thus the user, if needs a different allocation caps, should set it through
gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
vmethod. Otherwise the allocation_caps will be the same as the caps in the
src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=764421
2016-04-05 11:37:15 +02:00
Víctor Manuel Jáquez Leal b4a695cd11 libs: video: split allocation query caos and pad caps
Since the allocation query caps contains memory size and the pad's caps
contains the display size, a video encoder or decoder might need to allocate
a different frame size than the size negotiated in the caps.

This patch splits this logic distinction for videodecoder and videoencoder.

The user if needs a different allocation caps, should set the allocation_caps
in the GstVideoCodecState before calling negotiate() vmethod. Otherwise the
allocation_caps will be the same as the caps set in the src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=764421
2016-04-05 11:32:50 +02:00
Víctor Manuel Jáquez Leal 052fe11949 audioencoder: fix gtk-doc comment format 2016-04-04 17:12:16 +02:00
Mikhail Fludkov 7a206336dd rtpbasedepayload: look at ssrc before sequence numbers
Doing so prevents us dropping buffers in the rare, but possible, situations,
when the stream changes SSRC and new sequence numbers does not differ
much from the last sequence number from previous SSRC. For example:
ssrc - 0xaaaa 101,102,103,104 ssrc - 0xbbbb 102, 103, 104, 105...
In the scenario above we don't want to drop the first 3 packets of
0xbbbb stream.

https://bugzilla.gnome.org/show_bug.cgi?id=764459
2016-04-03 11:49:16 +03:00
Sebastian Dröge 6788003912 videorate: Don't fill up the segment with duplicate buffers if drop_only==TRUE 2016-04-03 11:40:50 +03:00
Sebastian Dröge eda44c640e videorate: Remove dead code
We never get into this code path at all if drop_only==TRUE.
2016-04-03 11:38:28 +03:00
Frédéric Bertolus 2626c02149 videorate: avoid useless buffer copy in drop-only mode
Make writable the buffer before pushing it lead to a buffer copy. It's
because a reference is keep for the previous buffer.
The previous buffer reference is only need to duplicate the buffer. In
drop-only mode, the previous buffer is release just after pushing the
buffer so a copy is done but it's useless.

https://bugzilla.gnome.org/show_bug.cgi?id=764319
2016-04-03 11:37:52 +03:00
Tim-Philipp Müller 9e311960cd video: fix example code in gst_video_frame_map() docs
GST_VIDEO_FRAME_PLANE_PSTRIDE() does not exist.

https://bugzilla.gnome.org/show_bug.cgi?id=764414
2016-04-02 15:19:44 +01:00
Tim-Philipp Müller 2102fdc983 discoverer: copy over result and seekable fields when copying a discoverer info
The function gst_discoverer_info_copy doesn't copy the data members seekable
and result of the source GstDiscovererInfo.

In the case of copying a GstDiscovererInfo for later use, the seekbale will be
undefined, which in practice usually will be false, even though the seekable of
the original GstDiscovererInfo is true.

https://bugzilla.gnome.org/show_bug.cgi?id=762710
2016-04-02 10:09:46 +01:00
Nicolas Dufresne a7809ecc8f video-format: Fix macro documentation
The parameter type was wrongly documenting that a GstVideoInfo structure
pointer was needed, while it needs a GstVideoFormatInfo structure
pointer.

https://bugzilla.gnome.org/show_bug.cgi?id=764414
2016-03-31 13:32:32 -04:00
Tim-Philipp Müller 778589cd5b test: fix indentation 2016-03-30 22:41:54 +01:00
Tim-Philipp Müller 3ab183c758 rtp: rtcpbuffer: fix indentation
https://bugzilla.gnome.org/show_bug.cgi?id=761944
2016-03-30 22:41:54 +01:00
Tim-Philipp Müller 5377088f1e rtp: rtpcbuffer: fix Since markers
https://bugzilla.gnome.org/show_bug.cgi?id=761944
2016-03-30 22:41:54 +01:00
Alessandro Decina 74efde50ad audio-resampler: disable neon on arm64
Fix the build on arm64 by using HAVE_ARM_NEON instead of __ARM_NEON__.
2016-03-30 11:16:49 +11:00
Jan Schmidt 1851777b94 subparse: Add more parsing guards
Insert extra checks for the validity of the incoming
data when parsing subrip/webvtt content and debug log
output for invalid content.

Should fix Coverity warnings.
2016-03-29 22:19:47 +11:00
Luis de Bethencourt 01778c5ac9 subparse: add missing break between formats
A break is missing at the end of case GST_SUB_PARSE_FORMAT_LRC or it will
fallthrough to WebVTT. This fixes commit fd2a14144a.
2016-03-29 11:27:40 +02:00
Sebastian Dröge 0582d5a1bc audio-resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x) in more places 2016-03-29 12:11:48 +03:00
Sreerenj Balachandran 1f03a7e41e win32: Update exports for new video formats
Update win32 exports for P010_10BE and P010_10LE
video formats.
2016-03-29 11:28:09 +03:00
Scott D Phillips 079ceb894c video: add P010 format support
P010 is a YUV420 format with an interleaved U-V plane and 2-bytes per
component with the the color value stored in the 10 most significant
bits.

https://bugzilla.gnome.org/show_bug.cgi?id=761607
---
Changes since v2:
- Set bits=16 in DPTH10_10_10_HI
Changes since v1:
- Fixed x-offset calculation in uv.
- Added 6-bit shifts to FormatInfo.
2016-03-29 11:16:42 +03:00
Sebastian Dröge 38a5a3614e resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x)
The latter is only available on x86-64 for some reason.
2016-03-29 10:15:07 +03:00
Edward Hervey de2ded9557 audio: Fix distcheck
Don't forget to dist the needed files (which don't need to be installed)
2016-03-29 08:22:29 +02:00
Wim Taymans 19f7d9ca46 audio-resampler: estimate memory usage in auto mode
Estimate the memory usage and use this to decide between full or
interpolated filter.
2016-03-28 15:37:36 +02:00
Wim Taymans f8e4c801eb audioresample: remove last ORC remains 2016-03-28 13:25:55 +02:00
Wim Taymans 984ee8a3f6 audio-resampler: small optimizations 2016-03-28 13:25:55 +02:00
Wim Taymans cf9059f070 audio-resampler: improve non-interleaved flags
Make it possible to have different interleaving on input and output
because we can quite trivially do that.
2016-03-28 13:25:55 +02:00
Wim Taymans 33855f0fe1 audio-resampler: unroll some more loops
Unroll some loops.
2016-03-28 13:25:55 +02:00
Wim Taymans 90a41b81dc audio-resampler: keep precision
Transpose and add before applying the cubic interpolation to avoid
overflows when using full precision.
2016-03-28 13:25:55 +02:00
Wim Taymans cc9d8594fe audio-resampler: small cleanups 2016-03-28 13:25:55 +02:00
Wim Taymans e209c0d565 audio-resampler: optimize no resampling
Switch to the faster nearest resample method when are doing no rate
conversion.
2016-03-28 13:25:54 +02:00
Wim Taymans f692d5e459 audio-resampler: add VARIABLE_RATE flag
Add a VARIABLE rate flag that selects an interpolating filter.
Move some function setup code in the _new function.
2016-03-28 13:25:54 +02:00