Commit graph

15227 commits

Author SHA1 Message Date
Arun Raghavan
91084f49c0 appsrc: Minor documentation cleanup 2016-01-19 12:56:20 +05:30
Tim-Philipp Müller
4f5a263381 tools: gst-play: allow setting of flags in serialized foo+bar format
https://bugzilla.gnome.org/show_bug.cgi?id=751901
2016-01-18 19:21:52 +00:00
Hugues Fruchet
1c72c6ddce tools: gst-play: add command line options for verbose output and playbin flags
https://bugzilla.gnome.org/show_bug.cgi?id=751901
2016-01-18 19:21:52 +00:00
Sebastian Dröge
385ee7c5d8 win32: Update exports 2016-01-18 15:51:16 +02:00
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
Mathieu Duponchelle
2717f4a86f streamsynchronizer: Ignore flushing streams [..]
[..] when resetting group start time. In GES, we are usually connected
to the streamsynchronizer on one audio and one video pad.

When seeking the timeline, both nlecompositions often output their flush_start
before any of them has output its flush_stop.

The current code, when receiving the first flush stop was using the
running time of the start of the second composition, which could
be pretty much anything, and means nothing at that point.

This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
both when setting flushing and when checking it.

https://bugzilla.gnome.org/show_bug.cgi?id=750013
2016-01-16 11:05:13 +01:00
Sebastian Dröge
fccf83e69f playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
Otherwise a decoder supporting GL memory will think that all downstream can
support GL memory because of seeing its own template caps.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-16 11:05:13 +01:00
Sebastian Dröge
9713ab06cd Revert "playbin: only add the template caps when the result is empty"
This reverts commit 023af2d3b1.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
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
Edward Hervey
62053852de playsink: Properly mark pending blocked pads
When blocking input pads, we also need to properly set the appropriate
pending flag.

Without this, when switching stream types after initial configuration
(like going from Audio+Video to Audio+Video+Sub) playsink would never
wait for *all* input streams to be blocked (it would just wait for the
new input pad (text in this case) to be blocked).

Since the reconfiguration might introduce unlinking/relinking of elements,
we need to ensure that *ALL* input streams are blocked.

Failure to do so would result in having some input streams pushing data
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
(returning GST_FLOW_NOT_LINKED).

A later optimization could involve only blocking the input pads that
might be involved in reconfiguration. But better be safe than sorry for
now :)
2016-01-15 10:05:58 +01:00
Nirbheek Chauhan
bc90841c7d gst-device-monitor: Use g_printerr instead of g_error
g_error is meant to be used for programmer errors (causes an abort),
not for expected runtime errors.
2016-01-14 17:51:13 +00:00
Thiago Santos
0d18717912 subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
Subset check verifies also that all required fields are present
and is mostly commonly used when checking if an element accepts
a certain caps
2016-01-13 16:32:25 -03:00
Thiago Santos
81c52aaa16 playbin: use subset check instead of intersect
Elements usually require that all fields on their caps are present
on the fixed caps they receive. Using intersection won't verify it,
resort to using is_subset() checks.

https://bugzilla.gnome.org/show_bug.cgi?id=760477
2016-01-13 15:29:17 -03:00
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
Thiago Santos
20f6af651b subtitleoverlay: replace accept-caps with caps query
Those accept caps are actually checking if downstream supports
some particular caps to check if it need to negotiate a different
format. Checking only the next element with accept-caps is not enough
to guarantee that it is supported.

Using a caps query makes it obtain the supported caps for downstream
as a whole instead of only the next element.
2016-01-11 18:35:29 -03:00
Sebastian Dröge
2f7cd8608a audio: Update exported symbols list 2016-01-08 21:27:16 +02:00
Thiago Santos
5ef0a09794 videorate: replace accept-caps with a caps query
accept-caps is only a shallow check, it needs to know
whether downstream as a whole accepts the framerate
2016-01-08 15:05:38 -03:00
Tim-Philipp Müller
56be0653e0 docs: fix up for GstAudioChannelMix rename as well 2016-01-08 16:37:25 +00:00
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
Sebastian Dröge
844aa3e6a9 playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
accept-caps is only for one element, caps query is recursive. Fixes playback
with totem and other situations.

https://bugzilla.gnome.org/show_bug.cgi?id=760234
2016-01-08 16:32:32 +02:00
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
Aurélien Zanelli
9b9f913809 videotestsrc: add missing break in set_property switch case
To avoid future issue when adding new properties.

https://bugzilla.gnome.org/show_bug.cgi?id=760204
2016-01-06 13:21:06 +02:00
Koop Mast
17b16e7a83 tests: audioconvert: fix test compilation with clang
With clang 3.7.1 on FreeBSD:
elements/audioconvert.c:650:12: error: shifting a negative signed value is
      undefined [-Werror,-Wshift-negative-value]
      (-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
       ~~~ ^

https://bugzilla.gnome.org/show_bug.cgi?id=760134
2016-01-06 01:13:32 +00:00
Tim-Philipp Müller
e5fb7275c2 tests: fix indentation of various unit tests 2016-01-06 01:12:13 +00:00
Tim-Philipp Müller
63d7a2a89a docs: add new audio API 2016-01-05 22:52:34 +00:00
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
eb09889176 audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.

CID 1346530 and 1346529
2015-12-29 18:14:54 +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
898940a37f audioconvert: add a test for gap handling 2015-12-29 14:40:32 +01:00
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
Tim-Philipp Müller
f2ecf85103 tools: gst-device-monitor: print uint properties in both decimal and hex
Some values are easier to read and make sense of in hex.

https://bugzilla.gnome.org//show_bug.cgi?id=759780
2015-12-29 11:29:31 +00:00
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
Sebastian Dröge
0416f121f2 typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
We would otherwise read beyond the array bounds and crash every now and then.
This was introduced with 5640ba17c8.

https://bugzilla.gnome.org/show_bug.cgi?id=759910
2015-12-28 13:51:02 +02:00
Stefan Sauer
267e7ba1d9 tests: remove commented code from audioconvert test
This is just what we have in gst_check_buffer_data().
2015-12-27 19:41:43 +01:00
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