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* 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.
* qtdemux: don't leak atom data in case of a wrong fourccStefan Kost2009-03-311-1/+1
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* matroska: don't leak read data in demuxerStefan Kost2009-03-311-0/+2
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* udp: don't use protocol in debug message after freeingStefan Kost2009-03-312-2/+2
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* rtpmp4adepay: output should be framed alreadyTim-Philipp Müller2009-03-301-2/+2
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* flac: require a 'newer' flac and remove support for the legacy flac APITim-Philipp Müller2009-03-276-524/+9
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* rtspsrc: link to the on_npt_stop signal to EOSWim Taymans2009-03-271-0/+12
| | | | | Connect to the on_npt_stop signal of the session manager to schedule the EOS actions.
* qtdemux: some stream synchronization to aid seeking in unbalanced clipsMark Nauwelaerts2009-03-261-5/+80
| | | | | | | Some clips (trailers) may have (length-wise) unbalanced streams, which stalls the pipeline if seeking into that region. Additional stream synchronization can handle this, as well as sparse (subtitle) streams (at some later time ?)
* qtdemux: additional safety and sanity checks (push based mode)Mark Nauwelaerts2009-03-261-1/+20
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* videomixer: some more indent fixesWim Taymans2009-03-261-3/+0
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* videomixer: fix gst-indent screwupWim Taymans2009-03-261-50/+0
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* rtspsrc: better error message when the RTSP extension for Real streams is ↵Tim-Philipp Müller2009-03-253-3/+45
| | | | | | | | missing Try to post a decent error message when it looks like we're failing because the Real RTSP extension plugin is missing. Also add i18n bits for rtspsrc so our error messages get translated.
* i18n: make sure gettext gives us UTF-8 at all timesTim-Philipp Müller2009-03-252-0/+2
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* rtpmp4apay,rtpmp4depay: fix buffer leaks in AAC payloader and depayloaderTim-Philipp Müller2009-03-252-0/+6
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* rtpmp4apay: warn if input is unframedTim-Philipp Müller2009-03-251-2/+8
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