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4831 commits

Author SHA1 Message Date
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
Wim Taymans
7bb149dcc1 audio-resampler: more neon optimizations 2016-03-28 13:25:54 +02:00
Wim Taymans
6dd5e5259f audio-resampler: avoid overflow in cubic interpolation 
Shift out an extra bit to have some more headroom when doing cubic
interpolation.
 2016-03-28 13:25:54 +02:00
Wim Taymans
61460fdfad audio-resampler: overread only 8 taps 
We only need 8 taps of zeroes as headroom for the SIMD optimized
functions.
 2016-03-28 13:25:54 +02:00
Wim Taymans
4772ebbddf audio-converter: use helper to check intermediate format 2016-03-28 13:25:54 +02:00
Wim Taymans
00e5a8bab8 audio-resampler: fix phase 2016-03-28 13:25:54 +02:00
Wim Taymans
9182ea17b5 audio-resampler: fix neon assembler 2016-03-28 13:25:53 +02:00
Wim Taymans
027165621b audio-resampler: avoid some format conversion 
Store the filter in the desired sample format so that we can simply do a
linear or cubic interpolation to get the new filter instead of having to
go through gdouble and then convert.
 2016-03-28 13:25:53 +02:00
Wim Taymans
2c33c2134c audio-resampler: fix neon linear float interpolation 2016-03-28 13:25:53 +02:00
Wim Taymans
d969a7a9d8 audio-resampler: reorder filter coefficients for more speed 
Reorder the filter coefficients to make it easier to use SIMD for
interpolation.
Fix orc flags a little.
Add specialized nearest resampling function.
 2016-03-28 13:25:53 +02:00
Wim Taymans
107f53ea0a audio-resampler: remove stereo optimizations 
The stereo optimizations don't give enough benefit.
Rename none to full to make it clear that we use a full filter instead
of an interpolated one
 2016-03-28 13:25:53 +02:00
Wim Taymans
b820074a49 audio-resample: remove neon double stubs 
NEON does not have double types.
 2016-03-28 13:25:53 +02:00
Wim Taymans
6f9237dfb5 audio-resampler: add more neon optimizations 2016-03-28 13:25:53 +02:00
Wim Taymans
307f360cca audio-resampler: add more neon optimizations 2016-03-28 13:25:53 +02:00
Wim Taymans
d5abdd83c9 audio-resampler: add neon optimizations 
Unroll some more loops in the fallback code that seems to work fine
for ARM.
Add some simple ARM optimizations taken from speex.
 2016-03-28 13:25:53 +02:00
Wim Taymans
25d81ffb55 audio-resampler: give better hints about the precision 
Give better hints to the compiler about the precision we expect from
the multiplications.
 2016-03-28 13:25:53 +02:00
Wim Taymans
ea497b509f audio-resample: small optimizations 
Remove some inline functions that are called in the slow path.
Unroll C fallback functions a little.
 2016-03-28 13:25:52 +02:00
Wim Taymans
167a415717 audio-resampler: Use n_phases when calculating taps offset 
Tweak linear interpolation oversampling.
Clear filter cache on rate changes when using a full filter.
 2016-03-28 13:25:52 +02:00
Wim Taymans
524ea147cc audio-resampler: improve filter construction 
Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
 2016-03-28 13:25:52 +02:00
Wim Taymans
0f3ff9177f audio-resampler: avoid overflow in fraction calculation 2016-03-28 13:25:52 +02:00
Wim Taymans
651ae201bc audio-resampler: increase precision 2016-03-28 13:25:52 +02:00
Wim Taymans
4cb52f1831 audio-resampler: add more optimizations 2016-03-28 13:25:52 +02:00
Wim Taymans
bdf194a09a audio-resample: fix taps conversion 
We do taps conversion in place so make sure we don't overwrite the
input with temporary data.
Optimize some more gint16 functions.
 2016-03-28 13:25:52 +02:00
Wim Taymans
f6e0481ab5 audio-resampler: Improve taps memory layout 
Rearrange the oversampled taps in memory to make it easier to use
SIMD instructions on them. this simplifies some sse code.
Add some more optimizations
 2016-03-28 13:25:52 +02:00
Wim Taymans
e9fc039bb1 audio-resampler: add cubic interpolation 2016-03-28 13:25:52 +02:00
Wim Taymans
58dcd0587d audio-resampler: add more functions 
Use some macros to generate more functions
 2016-03-28 13:25:51 +02:00
Wim Taymans
e02af5c534 audio-resampler: add linear interpolation method 
Make more functions into macros.
Add linear interpolation of filter coefficients.
 2016-03-28 13:25:51 +02:00
Wim Taymans
05d238def9 audio-resampler: add max-phase-error config 2016-03-28 13:25:51 +02:00
Wim Taymans
13e5b986cd audio-resampler: improve tap calculation 
Return the taps from make_taps, this makes it possible to not actually
have to cache the taps when we want to.
Fix overflow in phase calculation.
 2016-03-28 13:25:51 +02:00
Wim Taymans
6397db74cd audio-resampler: fix guint -> gint 2016-03-28 13:25:51 +02:00
Wim Taymans
45574ba4f4 audio-resampler: improve phase error 
Accept a phase error of maximum 10%, which turns out to be inaudible.
 2016-03-28 13:25:51 +02:00
Wim Taymans
b0b3350717 audio-resampler: improve phase calculation 
Also calculate the GCD with the current phase so that we can accurately
represent the current phase with the new resample rates.
 2016-03-28 13:25:51 +02:00
Wim Taymans
bbdb447b2b audio-resampler: fix history after buffer resize 
When we resize the temp buffer, move the history in its new place.
 2016-03-28 13:25:51 +02:00
Wim Taymans
ed747492ef audio-resampler: add reset function 
Add a function to reset the audio-resampler.
Use new function in audio-converter
Use the new functions in gstaudioresample and fixup drain functions.
 2016-03-28 13:25:51 +02:00
Wim Taymans
ea469ad9a8 audio-resampler: Small fixes 
Fix the phase.
Reset the new sample buffer with 0.
Move samples around when we change the filter size.
 2016-03-28 13:25:51 +02:00
Wim Taymans
a489f9ddb3 audio-resampler: Rework make_taps 
Make it return a pointer to the generated taps. That way we can later
decide to actually cache it or not.
 2016-03-28 13:25:51 +02:00
Wim Taymans
05eb109c0d audio-resampler: handle filter length changes 
Update the buffer with history samples when the filter length changes
because of an update of the parameters or sample rates.
 2016-03-28 13:25:51 +02:00
Wim Taymans
8dfb3ffb99 audio-resampler: fix samples_avail 
We only know the taps after we calculate them.
 2016-03-28 13:25:51 +02:00
Wim Taymans
c8fc9d88a7 audio-resampler: work on dynamically changing the samplerate 
Calculate the new phase for the new sample rate.
Fix some docs.
 2016-03-28 13:25:51 +02:00
Wim Taymans
4e48867097 audio-resampler: small cleanups 2016-03-28 13:25:51 +02:00
Wim Taymans
85c77659b9 audio-resampler: add fallback to mono function 
Remove stereo implementations. Implement fall back to mono functions
when the stereo function is missing.
 2016-03-28 13:25:50 +02:00
Wim Taymans
2555317a71 audio-resampler: add float stereo SSE function 2016-03-28 13:25:50 +02:00
Wim Taymans
e74c207433 audio-resampler: Fix compilation of intrinsics 
Only compile intrinsics when we are building for the selected
architecture.
Add sse4.1 optimized int32 resampler code.
 2016-03-28 13:25:50 +02:00
Wim Taymans
98bd349b88 audioconvert: only resample on supported formats 2016-03-28 13:25:50 +02:00
Wim Taymans
d348fbb9b9 audio-converter: make some optimized functions 
Make an optimized function that just calls the resampler when possible.
Optimize the resampler transform_size function a little.
 2016-03-28 13:25:50 +02:00
Wim Taymans
23531bdc93 audio-resampler: remove mirror function 
We don't need to mirror the input, just assume 0 samples.
Always move the processed samples to the start of the buffer.
Add some G_LIKELY
 2016-03-28 13:25:50 +02:00
Wim Taymans
6f685410b1 audio-resampler: also enable sse when sse2 is available 2016-03-28 13:25:50 +02:00
Wim Taymans
71871c5048 audio-resampler: optimizations 
Improve int16 resampling by using pmaddwd
Use intrinsics to scale and pack int16 samples
Align the coefficients so that we can use aligned loads
Add padding to taps and samples so that we don't have to use partial
loads for the remainder of the loops.
Remove copy_n, we can reuse the plain copy function with some new
parameters.
Align and pad the sample array.
 2016-03-28 13:25:50 +02:00
Wim Taymans
f55a67ca7c audio-resampler: make pluggable optimized functions 
Add support for x86 specialized functions and select them at runtime.
 2016-03-28 13:25:50 +02:00
Wim Taymans
819c4c26c7 audio-resampler: combine functions 2016-03-28 13:25:50 +02:00
Wim Taymans
de37491662 audio-converter: simplify API 
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
 2016-03-28 13:25:50 +02:00
Wim Taymans
1d9a793545 audio-converter: more work on resampling 
- Fix the resampler in the audio converter
- fix memory leaks
 2016-03-28 13:13:59 +02:00
Wim Taymans
75d668e152 audio-converter: add resampler 
Add a resampler to the processing chain when needed.
port the audio resampler to the new audioconverter library
 2016-03-28 13:13:59 +02:00
Tim-Philipp Müller
f4fb623aba audiodecoder: avoid unnecessary gst_pad_has_current_caps() checks 
No need to do this for each input buffer, we have the input caps
stored somewhere already.

https://bugzilla.gnome.org/show_bug.cgi?id=763337
 2016-03-24 14:49:12 +02:00
Jimmy Ohn
65f721b326 codec-utils: Add utilities for AAC and the AACHead header 
Add utilities about the channels and sample rate for AAC.

https://bugzilla.gnome.org/show_bug.cgi?id=749110
 2016-03-24 14:27:21 +02:00
Vineeth TM
44b70ca3a1 base: use new gst_element_class_add_static_pad_template() 
https://bugzilla.gnome.org/show_bug.cgi?id=763075
 2016-03-24 14:25:41 +02:00
Stian Selnes
93196092d1 rtcpbuffer: Add API for APP packets 
https://bugzilla.gnome.org/show_bug.cgi?id=761944
 2016-03-24 14:24:11 +02:00
Haakon Sporsheim
d8e9a711a0 rtcpbuffer: Add profile-specific extension API. 
https://bugzilla.gnome.org/show_bug.cgi?id=761950
 2016-03-24 14:22:54 +02:00
Víctor Manuel Jáquez Leal
327d617d39 install-plugins: update documentation 
Use gst-inspect-1.0 instead of gst-inspect-0.10

https://bugzilla.gnome.org/show_bug.cgi?id=763316
 2016-03-24 10:29:27 +02:00
Wim Taymans
c0ef1ea553 audio-channel-mixer: improve non-interleaved flags 
Make separate flags for non-interleaved input and output because the
channel mixer should be able to convert between the two layouts in the
future.
 2016-03-04 17:17:33 +01:00
Tim-Philipp Müller
b424712644 video: update disted orc backup file 
https://bugzilla.gnome.org/show_bug.cgi?id=761851
 2016-02-27 00:13:03 +00:00
Göran Jönsson
babcf4d30c video-converter: add direct UYVY to GRAY8 conversion function 
https://bugzilla.gnome.org/show_bug.cgi?id=761851
 2016-02-26 23:56:51 +00:00
Lim Siew Hoon
5ae5f9055f pbutils: docs: Remove the empty lines in between <refsect2> and </refsect2> 
They are converted into <para></para> by gtk-doc...

https://bugzilla.gnome.org/show_bug.cgi?id=762674
 2016-02-26 14:46:44 +02:00
Sebastian Dröge
87f380867b audiovisualizer: Let GstAudioVisualizerShader enum GType be autogenerated by glib-mkenums 
That happens automatically already anyway.
 2016-02-25 20:40:02 +02:00
Tim-Philipp Müller
c64dae3e1d video: flesh out docs for gst_video_frame_map() 2016-02-25 17:46:31 +00:00
Luis de Bethencourt
e23b12b1c9 visual: correct type name 
Base class type name should not reference libvisual since not all child
elements use this. This was an oversight when merging audiovisualizers into
a common base class.
 2016-02-25 10:51:50 +00:00
Wim Taymans
221e661f07 audio-quantize: fix feedback dither 
Make sure we allocated enough extra space in the error buffer to
store the feedback error.
 2016-02-24 14:57:31 +01:00
Wim Taymans
3e8cf31a96 audio-converter: perform dithering on the current format 
Use the current (intermediate) format to decide how to set up dithering
instead of the input format.
 2016-02-24 14:57:31 +01:00
Sebastian Dröge
2c29f09da8 rtpbasepayload: Handle gst_pad_get_current_caps() returning NULL gracefully 2016-02-23 18:23:45 +02:00
Adam Miartus
2bab28f494 tag: id3v2: read conductor tag 
ID3v2 features the TPE3 info frame, which contains information
about the conductor.

https://bugzilla.gnome.org/show_bug.cgi?id=762451
 2016-02-22 21:16:54 +00:00
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