Thibault Saunier [Tue, 26 Jan 2016 18:23:04 +0000 (19:23 +0100)]
encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
Some encoders can update the stream header through time (for example
vp8 might do that) but it does not strictly changes the output format.
Aurélien Zanelli [Tue, 26 Jan 2016 13:09:42 +0000 (14:09 +0100)]
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
Wim Taymans [Tue, 26 Jan 2016 15:56:57 +0000 (16:56 +0100)]
audio-converter: add reset function
Wim Taymans [Tue, 26 Jan 2016 15:36:41 +0000 (16:36 +0100)]
audio-converter: handle NULL input
Allow NULL as input to mean silence samples.
Wim Taymans [Tue, 26 Jan 2016 16:16:52 +0000 (17:16 +0100)]
audio-converter: improve _update_config
Allow NULL config to keep the existing parameters.
Fix the docs.
Wim Taymans [Tue, 26 Jan 2016 16:14:20 +0000 (17:14 +0100)]
audio-converter: audio-converter: make some optimized functions
Make optimized functions for generic and passthrough conversion.
Wim Taymans [Tue, 26 Jan 2016 15:34:35 +0000 (16:34 +0100)]
audio-quantize: add _reset function
Add a reset function that clears any history.
Tim-Philipp Müller [Mon, 25 Jan 2016 17:40:23 +0000 (17:40 +0000)]
build: remove nonsensical check for freetype
The examples need Gtk+, nothing uses freetype directly.
Tim-Philipp Müller [Mon, 25 Jan 2016 16:22:17 +0000 (16:22 +0000)]
tests: libvisual: make run faster
Reduce resolution, which shouldn't make any difference
to what's tested here. Makes test finish in less than
half the time it took before (8s vs. 21s).
Arun Raghavan [Mon, 25 Jan 2016 13:00:30 +0000 (18:30 +0530)]
alsa: Trivial doc update
alsasink now does more than just raw audio.
Sebastian Dröge [Thu, 21 Jan 2016 16:30:40 +0000 (18:30 +0200)]
decodebin: Correctly expose pads from elements that have directly exposable pads
analyze_new_pad() can return a new decode chain, which might have a new
GstDecodePad in the end. We should use those two for expose_pad() and not the
original ones that were passed to analyze_new_pad().
This fails when having a demuxer element that has raw pads immediately or
if a decoder with raw caps is after an adaptive demuxer.
https://bugzilla.gnome.org/show_bug.cgi?id=760949
Wim Taymans [Thu, 21 Jan 2016 15:08:46 +0000 (16:08 +0100)]
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
Wim Taymans [Thu, 21 Jan 2016 09:45:40 +0000 (10:45 +0100)]
video-color: add Adobe RGB primaries and transfer function
Wim Taymans [Wed, 20 Jan 2016 09:19:34 +0000 (10:19 +0100)]
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
Wim Taymans [Wed, 20 Jan 2016 09:02:20 +0000 (10:02 +0100)]
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
Thibault Saunier [Tue, 19 Jan 2016 22:26:57 +0000 (23:26 +0100)]
videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
It was never actually supported or used
https://bugzilla.gnome.org/show_bug.cgi?id=760666
Thibault Saunier [Tue, 19 Jan 2016 22:22:35 +0000 (23:22 +0100)]
Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
This reverts commit
63517d0ed348784cce4ab4b295c2c0f1b78baa81.
It was wrong ref counting wise and we decided to deprecated DROPPED
return value
https://bugzilla.gnome.org/show_bug.cgi?id=760666
Vineeth TM [Mon, 18 Jan 2016 02:40:36 +0000 (11:40 +0900)]
tests:audioconvert: Fix integer overflow build error
value of 32768L << 16 and 1L << 31 is
2147483648
but it exceeds the positive range of int which is
2147483647
resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
https://bugzilla.gnome.org/show_bug.cgi?id=760769
Arun Raghavan [Tue, 19 Jan 2016 07:09:22 +0000 (12:39 +0530)]
appsrc: Minor documentation cleanup
Tim-Philipp Müller [Thu, 14 Jan 2016 23:14:27 +0000 (23:14 +0000)]
tools: gst-play: allow setting of flags in serialized foo+bar format
https://bugzilla.gnome.org/show_bug.cgi?id=751901
Hugues Fruchet [Thu, 2 Jul 2015 15:58:00 +0000 (17:58 +0200)]
tools: gst-play: add command line options for verbose output and playbin flags
https://bugzilla.gnome.org/show_bug.cgi?id=751901
Sebastian Dröge [Mon, 18 Jan 2016 13:51:16 +0000 (15:51 +0200)]
win32: Update exports
Evan Callaway [Thu, 15 Oct 2015 14:38:16 +0000 (10:38 -0400)]
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
Sebastian Dröge [Sat, 16 Jan 2016 09:17:50 +0000 (10:17 +0100)]
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.
Mathieu Duponchelle [Sat, 9 Jan 2016 03:35:23 +0000 (04:35 +0100)]
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
Sebastian Dröge [Fri, 8 Jan 2016 16:53:52 +0000 (18:53 +0200)]
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
Sebastian Dröge [Fri, 8 Jan 2016 16:37:16 +0000 (18:37 +0200)]
Revert "playbin: only add the template caps when the result is empty"
This reverts commit
023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
https://bugzilla.gnome.org/show_bug.cgi?id=758212
Thibault Saunier [Fri, 15 Jan 2016 13:35:22 +0000 (13:35 +0000)]
videoencoder: Release video frame when ->handle return ERROR or DROPPED
https://bugzilla.gnome.org/show_bug.cgi?id=760666
Edward Hervey [Fri, 15 Jan 2016 08:50:29 +0000 (09:50 +0100)]
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 :)
Nirbheek Chauhan [Wed, 6 Jan 2016 04:42:43 +0000 (10:12 +0530)]
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.
Thiago Santos [Wed, 13 Jan 2016 19:32:25 +0000 (16:32 -0300)]
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
Thiago Santos [Tue, 12 Jan 2016 14:31:50 +0000 (11:31 -0300)]
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
Wim Taymans [Tue, 12 Jan 2016 14:56:36 +0000 (15:56 +0100)]
audio-channel-mixer: round before truncating
Round the result before truncating for int channel mixing.
Wim Taymans [Tue, 12 Jan 2016 14:27:16 +0000 (15:27 +0100)]
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
Wim Taymans [Tue, 12 Jan 2016 10:43:20 +0000 (11:43 +0100)]
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
Wim Taymans [Tue, 12 Jan 2016 10:37:17 +0000 (11:37 +0100)]
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.
Thiago Santos [Mon, 11 Jan 2016 21:24:48 +0000 (18:24 -0300)]
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.
Sebastian Dröge [Fri, 8 Jan 2016 19:27:16 +0000 (21:27 +0200)]
audio: Update exported symbols list
Thiago Santos [Fri, 8 Jan 2016 18:05:38 +0000 (15:05 -0300)]
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
Tim-Philipp Müller [Fri, 8 Jan 2016 16:08:47 +0000 (16:08 +0000)]
docs: fix up for GstAudioChannelMix rename as well
Wim Taymans [Fri, 8 Jan 2016 16:34:50 +0000 (17:34 +0100)]
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'
Wim Taymans [Fri, 8 Jan 2016 16:28:31 +0000 (17:28 +0100)]
audio-converter: prepare API for rate changes
Use the update function to update the sample rates along with the config
once we implement resampling.
Wim Taymans [Fri, 8 Jan 2016 16:17:44 +0000 (17:17 +0100)]
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.
Sebastian Dröge [Fri, 8 Jan 2016 15:50:21 +0000 (17:50 +0200)]
audio/video: Use G_GNUC_INTERNAL for internal functions
Wim Taymans [Fri, 8 Jan 2016 15:22:25 +0000 (16:22 +0100)]
audio: GstAudioChannelMix -> GstAudioChannelMixer
Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
looks better and to avoid a conflict with a library in -bad.
Sebastian Dröge [Thu, 7 Jan 2016 13:24:25 +0000 (15:24 +0200)]
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
Aurélien Zanelli [Wed, 6 Jan 2016 14:49:59 +0000 (15:49 +0100)]
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
Aurélien Zanelli [Wed, 6 Jan 2016 11:14:39 +0000 (12:14 +0100)]
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
Koop Mast [Wed, 6 Jan 2016 01:04:31 +0000 (01:04 +0000)]
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
Tim-Philipp Müller [Wed, 6 Jan 2016 01:06:10 +0000 (01:06 +0000)]
tests: fix indentation of various unit tests
Tim-Philipp Müller [Tue, 5 Jan 2016 22:52:34 +0000 (22:52 +0000)]
docs: add new audio API
Tim-Philipp Müller [Sun, 3 Jan 2016 17:21:18 +0000 (17:21 +0000)]
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.
Sebastian Dröge [Sun, 3 Jan 2016 08:33:53 +0000 (10:33 +0200)]
riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
Apparently this #define is unused.
Stefan Sauer [Sat, 2 Jan 2016 22:29:22 +0000 (23:29 +0100)]
riff-ids: remove trailing whitespace
Stefan Sauer [Sat, 2 Jan 2016 22:27:44 +0000 (23:27 +0100)]
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.
Sebastian Dröge [Thu, 31 Dec 2015 18:43:28 +0000 (20:43 +0200)]
sdp: Also reorder SUBDIRS to try even harder to build the RTP library first
Sebastian Dröge [Thu, 31 Dec 2015 18:41:38 +0000 (20:41 +0200)]
sdp: The SDP library depends on the RTP library now and is not independent anymore
Fix up the build dependencies.
Hyunjun Ko [Wed, 7 Oct 2015 09:50:18 +0000 (18:50 +0900)]
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
Sebastian Dröge [Tue, 29 Dec 2015 16:14:54 +0000 (18:14 +0200)]
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
Sebastian Dröge [Tue, 29 Dec 2015 15:56:21 +0000 (17:56 +0200)]
encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE
Sebastian Dröge [Tue, 29 Dec 2015 15:54:44 +0000 (17:54 +0200)]
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
Stefan Sauer [Tue, 29 Dec 2015 13:40:04 +0000 (14:40 +0100)]
audioconvert: add a test for gap handling
Stefan Sauer [Tue, 29 Dec 2015 13:23:59 +0000 (14:23 +0100)]
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
Tim-Philipp Müller [Tue, 29 Dec 2015 11:29:31 +0000 (11:29 +0000)]
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
Reynaldo H. Verdejo Pinochet [Thu, 12 Nov 2015 22:01:03 +0000 (14:01 -0800)]
videoblend: special case 1x1 src dims on increment computation
Fix crash with 1x1 overlay pixmap
https://bugzilla.gnome.org/show_bug.cgi?id=757290
Sebastian Dröge [Mon, 28 Dec 2015 10:28:26 +0000 (12:28 +0200)]
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
5640ba17c8db80976b7718904e4024dcfe9ee1a0.
https://bugzilla.gnome.org/show_bug.cgi?id=759910
Stefan Sauer [Sun, 27 Dec 2015 18:41:43 +0000 (19:41 +0100)]
tests: remove commented code from audioconvert test
This is just what we have in gst_check_buffer_data().
Stefan Sauer [Sun, 27 Dec 2015 18:25:20 +0000 (19:25 +0100)]
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.
Tim-Philipp Müller [Sat, 26 Dec 2015 11:34:47 +0000 (11:34 +0000)]
tools: gst-device-monitor: print non-string device properties too
Sebastian Dröge [Sat, 26 Dec 2015 08:43:56 +0000 (09:43 +0100)]
audio: Fix some documentation warnings
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
Sebastian Dröge [Sat, 26 Dec 2015 08:43:51 +0000 (09:43 +0100)]
videoaffinetransformmeta: Add (transfer none) annotation for return value
Sebastian Dröge [Fri, 25 Dec 2015 10:34:10 +0000 (11:34 +0100)]
playsink: Don't leak audio/video filters due to floating references weirdness
The filters' floating references are sinked during set_property() already,
which means that GstBin takes a new reference when adding the filter to it.
Get rid of the additional reference after adding the filter to the bin.
Sebastian Dröge [Fri, 25 Dec 2015 09:36:44 +0000 (10:36 +0100)]
playsink: Allow reuse of audio/video filters by unparenting them from their bins
And also recreate the chains if the filter is changing.
Sebastian Dröge [Fri, 25 Dec 2015 09:28:02 +0000 (10:28 +0100)]
playsink: Don't leak audio/video filters when using non-raw media
Sebastian Dröge [Thu, 24 Dec 2015 14:27:43 +0000 (15:27 +0100)]
Back to development
Sebastian Dröge [Thu, 24 Dec 2015 12:59:52 +0000 (13:59 +0100)]
pbutils: Link to libgstbase for bytewriter and adapter
Sebastian Dröge [Thu, 24 Dec 2015 12:59:15 +0000 (13:59 +0100)]
Release 1.7.1
Sebastian Dröge [Thu, 24 Dec 2015 12:10:08 +0000 (13:10 +0100)]
Update .po files
Sebastian Dröge [Thu, 24 Dec 2015 11:22:04 +0000 (12:22 +0100)]
po: Update translations
Thibault Saunier [Fri, 11 Dec 2015 14:38:00 +0000 (15:38 +0100)]
encodebin: Implement an encoding profile serialization format
https://bugzilla.gnome.org/show_bug.cgi?id=759356
Koop Mast [Sun, 20 Dec 2015 23:43:49 +0000 (00:43 +0100)]
configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.
https://bugzilla.gnome.org/show_bug.cgi?id=759713
Kazunori Kobayashi [Thu, 3 Dec 2015 02:53:05 +0000 (11:53 +0900)]
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
Thiago Santos [Wed, 16 Dec 2015 21:11:05 +0000 (18:11 -0300)]
examples: playback-test: remove unused variables
audiosink and videosink string variables are unused
Matthew Waters [Sun, 29 Nov 2015 23:28:55 +0000 (10:28 +1100)]
playbin: only add the template caps when the result is empty
Unconditionally adding the template caps when proxying the caps query will play
havoc with decoders that attempt to choose an output format based on some caps
features. Creating a sink that does not include those caps features and a
decoder/parser/etc that preferentially chooses some specific caps feature when
available, will always return the decoder/parser/etc template caps and choose a
feature that downstream will be unable to support.
Fix by limiting the addition of the template caps to when the result is actually
empty.
https://bugzilla.gnome.org/show_bug.cgi?id=758212
Sebastian Dröge [Thu, 17 Dec 2015 12:39:01 +0000 (13:39 +0100)]
configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0
It's meant to be used for external plugins that can then all be disabled via
--disable-external. gio-unix-2.0 however is just an optional dependency for
the TCP unit test.
Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part
there needs to be an AM_CONDITIONAL for the feature with FALSE.
Sebastian Dröge [Wed, 16 Dec 2015 16:07:54 +0000 (17:07 +0100)]
Revert "decodebin2: fix deadlock on chain shutdown"
This reverts commit
77dc09c3a9a5e5e371e189f39b5557db440a8dc9.
It can cause the FLUSH_START/STOP events to go to the sink elements, which
then causes state changes and various other problems. We shouldn't really
flush downstream here, the idea is to do *draining*.
Apart from that the testcase for the original bug here works without this
commit now.
Luis de Bethencourt [Wed, 16 Dec 2015 11:12:00 +0000 (11:12 +0000)]
multifdsink: fix typo in GST_WARNING_OBJECT
This should make easier to parse the debug logs.
s/fnctl/fcntl
Vincent Penquerc'h [Thu, 10 Apr 2014 14:36:15 +0000 (15:36 +0100)]
videorate: remove dead code
Since the loops increasing count from 0 are always run at least
once (if count < 1), count will always be at least one when
compared to the drop/dup conditions.
Coverity 1139674
Wim Taymans [Wed, 16 Dec 2015 09:45:48 +0000 (10:45 +0100)]
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.
Wim Taymans [Wed, 16 Dec 2015 09:44:16 +0000 (10:44 +0100)]
audioconvert: clear convert object
Sebastian Dröge [Wed, 16 Dec 2015 08:35:38 +0000 (09:35 +0100)]
docs: update to git
Nicolas Dufresne [Mon, 14 Dec 2015 18:59:02 +0000 (13:59 -0500)]
Revert "alsasrc: Disable HW timestamp"
This reverts commit
3642e9a3913a35c00f379034780c27298d09929c.
Xavier Claessens [Tue, 10 Nov 2015 17:54:23 +0000 (12:54 -0500)]
base: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
Nicolas Dufresne [Thu, 24 Sep 2015 22:26:51 +0000 (18:26 -0400)]
alsasrc: Disable HW timestamp
This is a workaround for broken pulse module.
Sebastian Dröge [Mon, 14 Dec 2015 18:03:33 +0000 (19:03 +0100)]
rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes
Evan Callaway [Mon, 14 Dec 2015 15:57:19 +0000 (10:57 -0500)]
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
Evan Callaway [Mon, 14 Dec 2015 14:10:16 +0000 (09:10 -0500)]
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
Sebastian Dröge [Mon, 14 Dec 2015 12:11:21 +0000 (13:11 +0100)]
rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
CID 1139615
Wim Taymans [Thu, 10 Dec 2015 16:46:26 +0000 (17:46 +0100)]
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.
Wim Taymans [Thu, 10 Dec 2015 15:26:40 +0000 (16:26 +0100)]
audio-converter: improve API for non-interleaved formats
Make it possible to pass an array of sample blocks when dealing with
non-interleaved formats.