Commit graph

329 commits

Author SHA1 Message Date
Tim-Philipp Müller
8d1122013b audiodecoder: add _finish_subframe() method
This allows us to output audio samples without discarding
any input frames, which is useful for some formats/codecs
(e.g. the MonkeysAudio decoder implementation in ffmpeg
which will might return e.g. 16 output buffers for an
input buffer for certain files).

In the past decoder implementations just concatenated
the returned audio buffers until a full frame had been
decoded, but that's no longer possible to do efficiently
when the decoder returns audio samples in non-interleaved
layout.

Allowing subframes to be output before the entire input
frame is decoded can also be useful to decrease startup
latency/delay.

https://gitlab.freedesktop.org/gstreamer/gst-libav/issues/49
2019-03-05 19:49:13 +00:00
Matthew Waters
675415bf2e gl: try to use highp precision where supported
The use of mediump as a specifier in GLSL shaders will have limited
resolution and when used as texture coordinates may become inaccurate
over texture sizes of 1024.
2019-02-28 17:26:32 +11:00
Sebastian Dröge
f90dac8d48 rtsp-connection: Make use of new GstRTSPMessage API for directly storing a body buffer and add API for writing multiple messages
By doing so we can send a whole GstBufferList and each memory in the
contained buffers without copying into a single memory area and with a
single writev() call. This improves performance considerably for
high-packet-rate streams.

This depends on https://gitlab.gnome.org/GNOME/glib/merge_requests/333
to be efficient, otherwise each chunk of memory is a separate write()
call.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/370
2019-01-29 14:17:23 +02:00
Sebastian Dröge
b3c0d8b89b rtsp-message: Add support for storing GstBuffers directly as body payload of messages
This makes it unnecessary for callers to first merge together all
memories, and it allows API like GstRTSPConnection to write them out
without first copying all memories together or using writev()-style API
to write multiple memories out in one go.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/370
2019-01-29 14:17:23 +02:00
Tim-Philipp Müller
c48b3d15c8 docs: add new interlaced video API to docs 2019-01-06 16:32:34 +00:00
Tim-Philipp Müller
a9cf6f238f video: build GstVideoAggregator which was moved from -bad 2018-12-28 12:16:12 +01:00
Sebastian Dröge
c02d3b03c2 videotimecode: Add API for initializing from a GDateTime with validation
The old API would only assert or return an invalid timecode, the new API
returns a boolean or NULL. We can't change the existing API
unfortunately but can at least deprecate it.
2018-12-19 23:11:24 +00:00
Mathieu Duponchelle
1edb2c4242 audio-converter: add API to determine passthrough mode
audioconvert's passthrough status can no longer be determined
strictly from input / output caps equality, as a mix-matrix can
now be specified.

We now call gst_base_transform_set_passthrough dynamically, based
on the return from the new gst_audio_converter_is_passthrough()
API, which takes the mix matrix into account.
2018-12-17 14:23:49 +00:00
Sebastian Dröge
fd1a31ee11 video-anc: Add API for converting GstVideoCaptionType from/to GstCaps 2018-12-15 21:31:14 +00:00
Justin Kim
5303e2c32b rtcpbuffer: add support XR packet parsing
According to RFC3611, the extended report blocks in XR packet can
have variable length. To visit each block, the iterator should look
into block header. Once XR type is extracted, users can parse the
detailed information by given functions.

Loss/Duplicate RLE
The Loss RLE and the Duplicate RLE have same format so
they can share parsers. For unit test, randomly generated
pseudo packet is used.

Packet Receipt Times
The packet receipt times report block has a list of receipt
times which are in [begin_seq, end_seq).

Receiver Reference Time paser for XR packet
The receiver reference time has ntptime which is 64 bit type.

DLRR
The DLRR report block consists of sub-blocks which has ssrc, last RR,
and delay since last RR. The number of sub-blocks should be calculated
from block length.

Statistics Summary
The Statistics Summary report block provides fixed length
information.

VoIP Metrics
VoIP Metrics consists of several metrics even though they are in
a report block. Data retrieving functions are added per metrics.

https://bugzilla.gnome.org/show_bug.cgi?id=789822
2018-12-13 14:01:06 -05:00
Tomasz Andrzejak
e0268c02ab audiodecoder: add API for setting caps on the source pad
This patch adds API in the audio decoder base class for setting the arbitrary
caps on the source pad.  Previously only caps converted from audio info were
possible.  This is particularly useful when subclass wants to set caps features
for audio decoder producing metadata.
2018-11-21 10:11:40 +00:00
Sebastian Dröge
6e9c71e6c1 video-anc: Implement a VBI encoder
This allows writing out data from caption meta and similar to VBI
2018-11-12 14:09:28 +00:00
Stian Selnes
f766b85b96 rtpbasepayload: rtpbasedepayload: Add source-info property
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947
2018-10-10 14:38:01 -04:00
Sebastian Dröge
c5b78fdc93 gl: Rename GST_TYPE_GL_STEREO_DOWNMIX GType macro everywhere
The old one still exists but behind GST_DISABLE_DEPRECATED
2018-10-03 14:49:32 +03:00
Olivier Crête
20566a54e4 docs: libs: Add new symbols to section file 2018-07-16 17:09:02 -04:00
Mathieu Duponchelle
0304c63f00 rtspdefs: Add gst_rtsp_generate_digest_auth_response_from_md5
Passwords are usually not stored in clear text, usually
the A1 section of the response is stored as is in .htdigest
files.

https://bugzilla.gnome.org/show_bug.cgi?id=796636
2018-06-21 15:32:12 +02:00
Mathieu Duponchelle
a4a27fdca8 sdp: Add new constructor, sdp_message_from_text
Helper function for bindings, in python for example
users can now replace:

res, msg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(text.encode()), msg)

with:

res, msg = GstSdp.SDPMessage.new_from_text(text)

https://bugzilla.gnome.org/show_bug.cgi?id=796563
2018-06-11 20:21:08 +02:00
Matthew Waters
c4367b63d8 gl/format: add a function to retrieve if a format is supported 2018-05-05 21:24:25 +10:00
Hyunjun Ko
56ab7e0e1d dmabufallocator: adds gst_dmabuf_allocator_alloc_with_flags
If we can guarantee the lifetime of the fd is longer than
the memory, we can use DONT_CLOSE flag not to close when release.
But it's not provided in gstdmabuf yet while gstfdmemory does.

For example, in case of using VA-API or MSDK, we would need this api.
Otherwise we should call dup to duplicate the fd.

https://bugzilla.gnome.org/show_bug.cgi?id=794829
2018-04-26 16:40:54 -04:00
Edward Hervey
9dceb6ca52 video: Add support for VANC and Closed Caption
This commits add common elements for Ancillary Data and Closed
Caption support in GStreamer:

* A VBI (Video Blanking Interval) parser that supports detection
  and extraction of Ancillary data according to the SMPTE S291M
  specification. Currently supports the v210 and UYVY video
  formats.

* A new GstMeta for Closed Caption : GstVideoCaptionMeta. This
  supports the two types of CC : CEA-608 and CEA-708, along with
  the 4 different ways they can be transported (other systems
  are super-set of those).

https://bugzilla.gnome.org/show_bug.cgi?id=794901
2018-04-09 15:15:24 +02:00
Edward Hervey
10c161c7a7 docs/libs: The big spring cleanup
* Explicitely specify which headers aren't to be included in gtkdoc-scan
  This is essentially all the headers that are not installed and only
  for internal/local usage. This also includes the orc-generated headers.
* Remove all symbols/sections that are no longer present (due to accurately
  scanning only the headers we need).
* Add or expose sections which weren't previously exposed
* Make sure the "unified" library headers (ex: gst/video/video.h) are used
  everywhere applicable. Only use the specific headers where applicable
  (such as the GL-implementation-specific objects)
* Add all documentation which was not previously exposed in the right sections
* Update 'types' file to get as many runtime information as possible

This brings down the number of unused symbols to 15 (from over 300).
2018-04-02 08:53:28 +02:00
Tim-Philipp Müller
bef2d2b9e3 docs: libs: add another missing symbol 2018-03-11 22:45:32 +00:00
Tim-Philipp Müller
a23046fb2e docs: add video region of interest add/get parameter api to docs 2018-03-11 19:07:04 +00:00
Tim-Philipp Müller
98fc23062f docs: add GstPhysMemoryAllocator to docs 2018-03-08 01:01:53 +00:00
Mathieu Duponchelle
9cf4293bde audio-converter: add a convenience conversion method
This is useful from python bindings

https://bugzilla.gnome.org/show_bug.cgi?id=793492
2018-02-15 20:51:30 +01:00
Mathieu Duponchelle
9046e6001b AudioConverter: register as boxed type
https://bugzilla.gnome.org/show_bug.cgi?id=793492
2018-02-15 20:51:30 +01:00
Nicolas Dufresne
731f1ca63e doc: Include new GstVideoOverlay API 2018-02-14 14:07:54 -05:00
Tim-Philipp Müller
4984c84505 docs: add GstAudioAggregator to docs 2018-02-13 17:10:42 +00:00
Tim-Philipp Müller
666bbb1550 docs: add moved gl lib to documentation 2017-12-19 12:01:56 +00:00
Tim-Philipp Müller
b60cc0274c appsrc: add support for pushing buffer lists
And samples that carry buffer lists.

https://bugzilla.gnome.org/show_bug.cgi?id=752363
2017-12-09 11:09:16 +00:00
Mathieu Duponchelle
545e0b003b API: gst_discoverer_info_get_live
https://bugzilla.gnome.org/show_bug.cgi?id=783722
2017-10-11 19:47:19 +02:00
Mathieu Duponchelle
2a26baf4be API: gst_discoverer_audio_info_get_channel_mask
https://bugzilla.gnome.org/show_bug.cgi?id=783722
2017-10-11 19:46:29 +02:00
Sebastian Dröge
bf68e74403 audio: Add stream align API for getting timestamp at discont and number of samples since discont
https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:06:24 +03:00
Sebastian Dröge
ec1e20ffe5 audio: Add helper object for audio discontinuity detection and sample alignment
This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.

https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:06:05 +03:00
Mathieu Duponchelle
877d6faeea [API]: gst_audio_channel_mixer_new_with_matrix
+ Refactor previous constructor to call on that new constructor

+ Reimplement is_passthrough to strictly check whether the matrix
  is an identity matrix, comparing channel-masks was incorrect:
  the mixer may be remixing from a list of positions to the same
  list of positions, but ordered differently, and reciprocally,
  the mixer may be remixing from a list of positions to another
  list of positions identically ordered

+ Remove unused tmp field, must have been a refactoring leftover

https://bugzilla.gnome.org/show_bug.cgi?id=785471
2017-09-22 16:19:58 +02:00
Georg Lippitsch
b3df5786a9 videotimecode: Init from GDateTime
Add a function to init the time code from a GDateTime

https://bugzilla.gnome.org/show_bug.cgi?id=778702
2017-02-23 19:50:39 +02:00
Víctor Manuel Jáquez Leal
4fa6a2aba1 docs: update libs section
Include documented symbols that were not declared in section file.
2017-01-21 18:06:11 +01:00
Thibault Saunier
46b424a38b encoding-profile: Add a way to copy an encoding profile
It is often usefull to make sure that you get a full copy of a profile.
For example you want to let the user modify it in the user interface
but still keep an unchanged version for later use.

API:
  gst_encoding_profile_copy
2017-01-06 11:40:20 -03:00
Evan Nemerson
98064ed9bf audioringbuffer: add set_callback_full() for g-i
https://bugzilla.gnome.org/show_bug.cgi?id=678301
2016-12-22 15:34:58 +00:00
Sebastian Dröge
90b24d34b3 rtsp: Add gst_rtsp_message_parse_auth_credentials() to parse authentication credentials
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-21 09:39:21 +02:00
Sebastian Dröge
828c8604dd rtsp: Add gst_rtsp_generate_digest_auth_response() to calculate digest auth response
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-21 09:39:21 +02:00
Julien Isorce
3bf893e12a video: add gst_video_decoder_allocate_output_frame_with_params
It adds a third argument to pass GstBufferPoolAcquireParams
to gst_buffer_pool_acquire_buffer.

If a user subclasses GstBufferPoolAcquireParams, this allows to
pass an updated param to the underlying buffer pool at each
gst_video_decoder_allocate_output_frame_with_params call.

https://bugzilla.gnome.org/show_bug.cgi?id=773165
2016-11-04 16:18:13 +00:00
Julien Isorce
f5eb366335 allocators: define GST_CAPS_FEATURE_MEMORY_DMABUF
Adds "memory:DMABuf" caps feature. Since 1.11 tag.
Useful when the the dma-buf buffer cannot be mapped to CPU for r/w requests.
Example: protected content or platform constraints.

https://bugzilla.gnome.org/show_bug.cgi?id=759358
2016-11-03 13:19:12 -04:00
Nicolas Dufresne
c37b1e8c56 dmabuf: Make the allocator sub-classable
This should allos for cleaner code when implement such allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=768794
2016-11-03 13:19:12 -04:00
Sebastian Dröge
79809633de video-info: Add optional field-order caps field for interlaced-mode=interleaved
Usually this information is static for the whole stream, and various
container formats store this information inside the headers for the
whole stream.

Having it inside the caps for these cases simplifies code and makes it
possible to express these requirements more explicitly with the caps.

https://bugzilla.gnome.org/show_bug.cgi?id=771376
2016-11-01 20:40:07 +02:00
Xabier Rodriguez Calvar
0341f04ce1 videodirection: interface for rotation and flip
A GstVideoOrientationMethod enumeration is also provided for the
admitted property values.

https://bugzilla.gnome.org/show_bug.cgi?id=768687
2016-08-25 10:19:13 +03:00
Sebastian Dröge
7f7d667e0f videotimecode: Add to docs and exports list 2016-08-04 19:06:45 +03:00
Joan Pau Beltran
c6722c06a0 appsink: add _pull_sample/preroll() variants with timeout
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).

This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.

To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.

Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".

https://bugzilla.gnome.org/show_bug.cgi?id=768852
2016-07-18 16:55:16 +01:00
Sebastian Dröge
dc8120f298 appsrc: Add duration property for providing a duration in TIME format
https://bugzilla.gnome.org/show_bug.cgi?id=766229
2016-05-10 16:50:32 +03:00
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