- 05 Apr, 2017 1 commit
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Damien Riegel authored
This adds partial support for the RFC 4175 (raw video over RTP). The only supported formats are the YCbCr-4:2:2 8 bit because it's natively supported by FFmpeg with pixel format UYVY, and 10 bit which requires the vrawdepay codec to convert the payload in a format handled by FFmpeg. Signed-off-by:
Damien Riegel <damien.riegel@savoirfairelinux.com> Reviewed-by:
Rostislav Pehlivanov <atomnuker@gmail.com>
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- 05 Nov, 2016 1 commit
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Timur Aydin authored
When ffplay is used to play from the RTSP URL that serves 24 bit audio content, ffplay fails to recognize the audio codec format. The attached patch adds support for playing 24 bit audio content over RTSP by defining a dynamic payload handler for "L24". Signed-off-by:
Michael Niedermayer <michael@niedermayer.cc>
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- 22 Oct, 2016 1 commit
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Carl Eugen Hoyos authored
Add new mime types AAL2-G726 for g726 as suggested in rfc 3551. This patch will break interaction with applications that incorrectly use big-endian G.726 with mime type G726 but we know of at least one device (DVTel camera) that correctly implements the rfc, so do the same. Fixes ticket #5890.
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- 07 Jun, 2016 1 commit
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Diego Biurrun authored
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- 11 May, 2016 1 commit
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Martin Storsjö authored
It doesn't matter what the actual reason for not returning an AVPacket was - if we didn't return any packet and we have the next one queued, parse it immediately. (rtp_parse_queued_packet always consumes a queued packet if one exists, so there's no risk for infinite loops.) Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 04 May, 2016 1 commit
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Vittorio Giovara authored
Signed-off-by:
Diego Biurrun <diego@biurrun.de>
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- 26 Mar, 2016 1 commit
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Michael Niedermayer authored
Signed-off-by:
Michael Niedermayer <michael@niedermayer.cc>
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- 21 Mar, 2016 1 commit
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Thomas Volkert authored
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- 23 Feb, 2016 1 commit
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Anton Khirnov authored
Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
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- 19 Feb, 2016 1 commit
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Diego Biurrun authored
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- 16 Sep, 2015 3 commits
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Luca Barbato authored
And avoid a memory leak. Signed-off-by:
Luca Barbato <lu_zero@gentoo.org>
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Eloi BAIL authored
This commit print as AV_LOG_VERBOSE the jitter buffer size. It might be the default value or the value set by application. Signed-off-by:
Eloi BAIL <eloi.bail@savoirfairelinux.com> Signed-off-by:
Martin Storsjö <martin@martin.st>
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Eloi BAIL authored
This commit adds a warning trace when jitter buffer is full. It helps to understand leading decoding issues. Signed-off-by:
Eloi BAIL <eloi.bail@savoirfairelinux.com> Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 15 Sep, 2015 2 commits
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Eloi BAIL authored
This commit adds an error trace when jitter buffer is full. It helps to understand leading decoding issues. Signed-off-by:
Michael Niedermayer <michael@niedermayer.cc>
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Eloi BAIL authored
This commit print as AV_LOG_INFO the jitter buffer size. It might be the default value or the value set by application. Signed-off-by:
Eloi BAIL <eloi.bail@savoirfairelinux.com> Signed-off-by:
Michael Niedermayer <michael@niedermayer.cc>
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- 19 Apr, 2015 1 commit
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Vittorio Giovara authored
This applies to every library where performance is not critical.
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- 05 Mar, 2015 1 commit
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Thomas Volkert authored
The code was tested with live555 server. Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 01 Mar, 2015 1 commit
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Gilles Chanteperdrix authored
This makes more sense than mapping to AV_CODEC_ID_SUBRIP. Nothing indicates that a T.140 track contains subrip sub-titles. Signed-off-by:
Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org> Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 24 Feb, 2015 3 commits
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Martin Storsjö authored
This makes it clear that the individual parsing functions can't touch the parsed out value. Signed-off-by:
Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This avoids allocating space for a too large buffer for all the name strings. Signed-off-by:
Martin Storsjö <martin@martin.st>
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Gilles Chanteperdrix authored
Map this to AV_CODEC_ID_TEXT. Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 22 Feb, 2015 1 commit
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Thomas Volkert authored
(tested with live555 RTSP server) Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 21 Feb, 2015 4 commits
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Gilles Chanteperdrix authored
Signed-off-by:
Martin Storsjö <martin@martin.st>
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Gilles Chanteperdrix authored
Signed-off-by:
Martin Storsjö <martin@martin.st>
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Gilles Chanteperdrix authored
When receiving an RTCP packet, the difference between the last RTCP timestamp and the base timestamp may be negative. As these timestamps are of the uint32_t type, the result becomes a large integer. Cast the difference to int32_t to avoid this issue. The result of this issue is very large start times for RTSP streams, and difficulty to restart correctly after a pause. Signed-off-by:
Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org> Signed-off-by:
Martin Storsjö <martin@martin.st>
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Gilles Chanteperdrix authored
When receiving an RTCP packet, the difference between the last RTCP timestamp and the base timestamp may be negative. As these timestamps are of the uint32_t type, the result becomes a large integer. Cast the difference to int32_t to avoid this issue. The result of this issue is very large start times for RTSP streams, and difficulty to restart correctly after a pause. Signed-off-by:
Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org> Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 15 Feb, 2015 2 commits
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Thomas Volkert authored
Tested with live555 RTSP server Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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Thomas Volkert authored
Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 14 Feb, 2015 2 commits
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Gilles Chanteperdrix authored
Signed-off-by:
Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org> Reviewed-by:
Thomas Volkert <silvo@gmx.net> Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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Gilles Chanteperdrix authored
Reviewed-by:
Thomas Volkert <silvo@gmx.net> Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 12 Feb, 2015 1 commit
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Gilles Chanteperdrix authored
Reviewed-by:
Thomas Volkert <silvo@gmx.net> Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 23 Dec, 2014 1 commit
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Michael Niedermayer authored
Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 18 Dec, 2014 1 commit
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Thomas Volkert authored
The packetizer only supports splitting at GOB headers - if such aren't available frequently enough, it splits at any random byte offset (not at a macroblock boundary either, which would be allowed by the spec) and sends a payload header pretend that it starts with a GOB header. As long as a receiver doesn't try to handle such cases cleverly but just drops broken frames, this shouldn't matter too much in practice. Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 24 Oct, 2014 1 commit
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Martin Storsjö authored
The ones left using av_gettime are NTP timestamps (for RTCP, which is specified to send the actual current realtime clock in RTCP SR packets), and the NUT muxer timestamper, which is documented as using wallclock time. Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 24 Sep, 2014 1 commit
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Vittorio Giovara authored
The RFC spec draft only specifies the "H265" name - there is no specification saying how to interpret "HEVC" (if such a packet format is specified it could be an entirely different format). Since this is a very new standard (still a draft), there is little need for compatibility with existing, broken implementations. Therefore remove the extra alias, to avoid the risk of encouraging incorrect usage. Intentionally keeping the ff_hevc_dynamic_handler name for the handler, to use "hevc" consistently as name for the codec instead of "h265" within the library internals as long as there only is one single variant in actual use. Signed-off-by:
Martin Storsjö <martin@martin.st>
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- 03 Sep, 2014 1 commit
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Thomas Volkert authored
As specified in draft-ietf-payload-rtp-h265-06. Signed-off-by:
Luca Barbato <lu_zero@gentoo.org>
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- 26 Aug, 2014 1 commit
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ThomasVolkert authored
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- 24 Aug, 2014 1 commit
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ThomasVolkert authored
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- 09 Jul, 2014 1 commit
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Anton Khirnov authored
Use it for logging, instead of NULL or the stream codec context.
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- 08 Dec, 2013 1 commit
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Andrey Utkin authored
Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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