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* qtdemux: implement 3GPP (TS 26.244 V8.0.0) Asset metadata handling, Fixes ↵Marco Ballesio2009-04-155-2/+145
| | | | | | #132193 Implements 3gpp iso metadata tags which are different from mov udta atoms.
* debugutils: Use G_BEGIN_DECLS/G_END_DECLS.Peter Kjellerstedt2009-04-151-25/+17
| | | | | Use G_BEGIN_DECLS/G_END_DECLS to avoid gst-indent messing up the indentation due to extern "C" { }.
* debug: rename debug to debugutils to avoid clash with --disable-debug. Fixes ↵Stefan Kost2009-04-1525-4/+4
| | | | #562168
* debug: indent before renamingStefan Kost2009-04-158-71/+58
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* g726depay: add property for aal2 forceWim Taymans2009-04-151-0/+50
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* g726depay: implement RFC3551 packingWim Taymans2009-04-152-20/+154
| | | | | | | We implemented the AAL2 packing, add the encoding-name for those to the caps and a property to force AAL2 decoding (always TRUE for now). Implement RFC3551 unpacking for regular G726. See #567140.
* rtph263pay: fix buildWim Taymans2009-04-151-2/+0
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* h263pay: various fixesYouness Alaoui2009-04-141-20/+27
| | | | | | | Re-enable mode A support and a property to control it. Fix memory leak of GstRtpH263PayBoundry objects. Fix marker. Fixes #509311
* h263pay: Fix the payloaderJanin Kolenc2009-04-142-186/+1987
| | | | | Fix the H263 payloader to be more RFC 2190 compliant. See #509311
* avidemux: don't push EOS in streaming modeWim Taymans2009-04-141-4/+2
| | | | | | In streaming mode, avidemux is not supposed to send an EOS event downstream but it is supposed to return UNEXPECTED from the chain function instead so that upstream can do the right EOS handling.
* Add initial support for muxing/demuxing Speex audioSebastian Dröge2009-04-133-0/+134
| | | | | Note: This is not in the Matroska spec yet Fixes bug #578310.
* pulsesink: handle NULL timing infoWim Taymans2009-04-101-11/+15
| | | | Don't crash when the timing info is not yet available.
* pulse: make it work on 0.9.12Stefan Kost2009-04-102-12/+22
| | | | | | | | First we ignore request to fill the ringbuffer which are less then a segment. The small request where causing stutter. Then we disable flushing the stream when running against pa 0.9.12 as this triggers an assertiong in the sound server and terminates it. It does not happen with 0.9.10 and 0.9.14.
* pulsesink: handle server disconnect in get_timeWim Taymans2009-04-101-0/+11
| | | | | When the server is disconnected or when we are shut down, make our clock return an invalid time instead of erroring out.
* pulsesink: bps is signed int to avoid overflowWim Taymans2009-04-101-2/+1
| | | | | Keep bps as gint instead of guint because we will be doing signed math with it later on and we don't want weird results.
* avidemux: add convert query, fix duration queryLRN2009-04-101-1/+31
| | | | | | | | | Fix the duration query so that it also works with formats other than TIME, such as DEFAULT to get the number of frames. Add a convert function. Fixes #578052.
* pulsesink: check for a streamWim Taymans2009-04-091-19/+8
| | | | | Don't try to change the stream volume (and other things) when we don't have a stream yet. Just store the values for later.
* pulsesink: fix compilation for newer pulseaudioWim Taymans2009-04-091-2/+2
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* pulsesink: uncork fixes and use prebuf = 0Wim Taymans2009-04-091-77/+69
| | | | | | | We can use prebuf = 0 to instruct pulse to not pause the stream on underflows. This way we can remove the underflow callback. We however have to manually uncork the stream now when we have no available space in the buffer or when we are writing too far away from the current read_index.
* pulsesink: handle write errorsWim Taymans2009-04-091-3/+14
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* pulsesink: write silence on underflowWim Taymans2009-04-091-0/+38
| | | | | Start filling up the buffer with empty samples when an underflow happens. We need to do this to keep pulseaudio reporting the right time for us.
* pulsesink: handle pull-based schedulingWim Taymans2009-04-091-118/+11
| | | | | Use the default basesink methods for implementing pull based scheduling, it works fine for us.
* pulsesink: add beginnings of pull-based schedulingWim Taymans2009-04-091-19/+116
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* pulsesink: keep track of clock resetWim Taymans2009-04-091-41/+63
| | | | | | | when we switch streams, the clock will reset to 0. Make sure that the provided clock doesn't get stuck when this happens by keeping an initial offset. We also need to make sure that we subtract this offset in samples when writing to the ringbuffer.
* pulsesink: rewrite pulsesinkWim Taymans2009-04-092-770/+1269
| | | | | Derive from BaseAudioSink and implement our custom ringbuffer that maps to the internal pulseaudio ringbuffer.
* pulse: remove some stray debug linesWim Taymans2009-04-091-8/+0
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* jpegdec: use slightly more adaptive formula for QoSTim-Philipp Müller2009-04-092-6/+24
| | | | | Should work at least a tad better if the decoder can't keep up, and should also spread dropped frames a bit more evenly over time.
* wavparse: don't leak pad-templateStefan Kost2009-04-071-0/+1
| | | | gst_element_class_add_pad_template() does not take ownership.
* Automatic update of common submoduleFelipe Contreras2009-04-041-0/+0
| | | | From d0ea89e to b3941ea
* add pending_samples so that we only update segment's last stop after really ↵Thomas Vander Stichele2009-04-042-1/+9
| | | | sending the samples
* add debug and an assertThomas Vander Stichele2009-04-041-1/+7
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* add debuggingThomas Vander Stichele2009-04-041-0/+2
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* add a test to check that we get all decoded bytesThomas Vander Stichele2009-04-043-1/+206
| | | | | | | | | from a 10-buffer audiotestsrc flac, in the case of: - a full decode - a decode of a seek for the full file - a decode of a seek for a small part, smaller than the first buffer The test fails because flacdec drops the first outgoing buffer on a seek
* clipping should also work if it's done on the first buffer starting at 0Thomas Vander Stichele2009-04-041-1/+1
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* Automatic update of common submoduleEdward Hervey2009-04-041-0/+0
| | | | From f8b3d91 to d0ea89e
* Fix grammar.Zaheer Merali2009-04-031-1/+1
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* rtspsrc: allow http:// on the proxy settingWim Taymans2009-04-021-2/+6
| | | | | | Allow and ignore http:// at the start of the proxy setting, like souphttpsrc. Fixes #573173
* rtspsrc: don't leak the udpsrc padWim Taymans2009-04-021-8/+1
| | | | | Fix memory leak in rtspsrc because we didn't unref the udpsrc pad. See #577318
* rtptheorapay: fix length encoding in packed headers.Michael Smith2009-04-011-1/+4
| | | | As for vorbis payloader; this by inspection had the same bug.
* rtpvorbispay: in packed headers, properly flag multibyte lengths.Michael Smith2009-04-011-1/+4
| | | | | | In the sequence of header lengths, for headers >127 bytes, we use multiple bytes to encode the length. Bytes other than the last must have the top (flag) bit set.
* id3v2mux: write RVA2 frames containing peak/gain volume dataJonathan Matthew2009-04-022-8/+106
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* jpegdec: demote some log message from DEBUG to LOGTim-Philipp Müller2009-04-022-9/+12
| | | | And log decoder object.
* jpegdec: implement basic QoSTim-Philipp Müller2009-04-012-3/+129
| | | | Don't decode frames that are going to be too late anyway.
* rtspsrc: don't emit ugly warnings with older rtpjitterbuffer versionsTim-Philipp Müller2009-04-011-2/+7
| | | | | | The on-npt-stop signals was added only recently to rtpjitterbuffer in -bad, so check if the signal exists before g_signal_connect()ing to it, to avoid warnings.
* rtspsrc: add proxy supportWim Taymans2009-03-312-0/+89
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* matroska: don't leak serialized values when writing tagsStefan Kost2009-03-311-0/+1
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* matroska: don't alter passed data and especialy don't leak.Stefan Kost2009-03-311-3/+5
| | | | If we need different size, Make a copy, work with that and free it.
* goom: the structure is not fully initialized, but the copied.Stefan Kost2009-03-311-1/+2
| | | | Set to fully to 0 to avoid creep of uninitialized values.
* matroska: init endianess as such and signedness as boolean.Stefan Kost2009-03-311-2/+3
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* qtdemux: don't use ininitialized var in debug log statementStefan Kost2009-03-311-2/+2
| | | | Also make the log statement useful by printing the human readable format name.
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