- 08 Mar, 2017 2 commits
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Sebastian Dröge authored
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Sebastian Dröge authored
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- 03 Mar, 2017 3 commits
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Jan Schmidt authored
Ensure that reverse playback works and generates the range of timestamps (0-3s) we expect, in monotonically descending order.
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Jan Schmidt authored
Fix the check for whether the start time of the segment has been reached when playing in reverse. Otherwise, playback stops after reaching the start of any file part, instead of continuing until all parts within the segment have played
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Jan Schmidt authored
We parse the next moof in advance of having pushed all samples from the previous one in some cases, and we'll still need the crypto info from the previous fragment so keep around any unused crypto info entries when adding new ones
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- 02 Mar, 2017 2 commits
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- 01 Mar, 2017 1 commit
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- 28 Feb, 2017 5 commits
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Sebastian Dröge authored
CID 1398545
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Sebastian Dröge authored
CID 1363332
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Sebastian Dröge authored
CID 1212149
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Sebastian Dröge authored
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Edward Hervey authored
Needs to know where the gstapp headers are
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- 27 Feb, 2017 9 commits
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Sebastian Dröge authored
qtdemux.c: In function ‘qtdemux_parse_samples’: qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context] if (stream->samples_per_frame * stream->bytes_per_frame) { ~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
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Sebastian Dröge authored
gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’: gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size] memset (mp3parse->xing_seek_table_inverse, 0, 256); ^~~~~~ gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’: gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size] memset (mp3parse->xing_seek_table_inverse, 0, 256); ^~~~~~
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Sebastian Dröge authored
This prevents storing an infinite amount of e.g. comment headers if they come without a new initialization header in front of them. There can only be one header of each type.
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Sebastian Dröge authored
Check if encoding, payloading, depayloading and decoding works if the stream configuration (and thus the headers) change.
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Sebastian Dröge authored
If we also replace all headers when receiving any possibly following comments header, we would throw away the config header before being able to make use of it.
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George Kiagiadakis authored
A sparse stream's ending timestamp can be considerably smaller than the ending timestamps of the other streams, which can lead to skipping considerable time from the next part. https://bugzilla.gnome.org/show_bug.cgi?id=761086
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George Kiagiadakis authored
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- 26 Feb, 2017 1 commit
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Andrew authored
In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP timestamps is more than (3 * jbuf->clock_rate) we call rtp_jitter_buffer_reset_skew which resets pts to 0. So components down the pipeline (playes, mixers) just skip frames/samples until pts becomes equal to pts before gap. In version 1.10.2 and before this checking was bypassed for packets with "estimated dts", and gaps were handled correctly. https://bugzilla.gnome.org/show_bug.cgi?id=778341
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- 24 Feb, 2017 5 commits
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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- 22 Feb, 2017 2 commits
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Seungha Yang authored
Let libsoup handle redirection automatically. And then, to figure out redirection uri, extract it on "restarted" callback which will be fired before soup_session_send() is returned. https://bugzilla.gnome.org/show_bug.cgi?id=778428
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Nicolas Dufresne authored
Update the image size according the amount of data we are going to read/write. This workaround bugs in driver where the sizeimage provided by TRY/S_FMT represent the buffer length (maximum size) rather then the expected bytesused (buffer size). https://bugzilla.gnome.org/show_bug.cgi?id=775564
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- 21 Feb, 2017 2 commits
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Guillaume Desmottes authored
streamheader and codec_data buffers fields are only meant to be in the negotiated caps, not the template caps. Fixes false-positive leaks of those buffers detected by the leaks tracer, as template caps are static, and we decided to not include code in gstreamer core to handle this unusual case of template caps having buffers in them. https://bugzilla.gnome.org/show_bug.cgi?id=768762
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- 20 Feb, 2017 1 commit
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Jochen Henneberg authored
The payloader needs to reset and update the vorbis config data which is pushed on the network if it receives new headers, or at least, it may have to do so. Without this, the stream configuration could change without the payloader sending the new configuration to the other side.
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- 17 Feb, 2017 5 commits
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Jan Schmidt authored
This reverts commit 107902ec. This commit intended to ensure that keyframe seeks land at the start timestamp of a keyframe, rather than in the middle of one, but they cause trouble on files with sparse streams, or with JPEG 'cover art' tracks that have only one or a few JPEG samples with very long durations. That's still desirable for doing seamless cutting of videos, but needs a rethink for implementation. https://bugzilla.gnome.org/show_bug.cgi?id=778690
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- 16 Feb, 2017 1 commit
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Jan Schmidt authored
Add a new boolean surround-delay property that makes audioecho just apply a delay to certain channels to create a surround effect, rather than an echo on all channels. This is useful when upmixing from stereo - for example. Add a surround-mask property to control which channels are considered surround sound channels when adding a delay with surround-delay = true Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>
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- 14 Feb, 2017 1 commit
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Sebastian Dröge authored
This goes around the inefficient control message based filtering and does all the filtering kernel-side. Unfortunately this is Linux-only and there is no IPv6 variant of it (yet).
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