Commit Graph

41 Commits

Author SHA1 Message Date
Michael Niedermayer
9d76cf0b18 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtpdec: Templatize the code for different g726 bitrate variants
  rv40: move loop filter to rv34dsp context
  lavf: make av_set_pts_info private.
  rtpdec: Add support for G726 audio
  rtpdec: Add an init function that can do custom codec context initialization
  avconv: make copy_tb on by default.
  matroskadec: don't set codec timebase.
  rmdec: don't set codec timebase.
  avconv: compute next_pts from input packet duration when possible.
  lavf: estimate frame duration from r_frame_rate.
  avconv: update InputStream.pts in the streamcopy case.

Conflicts:
	avconv.c
	libavdevice/alsa-audio-dec.c
	libavdevice/bktr.c
	libavdevice/fbdev.c
	libavdevice/libdc1394.c
	libavdevice/oss_audio.c
	libavdevice/v4l.c
	libavdevice/v4l2.c
	libavdevice/vfwcap.c
	libavdevice/x11grab.c
	libavformat/au.c
	libavformat/eacdata.c
	libavformat/flvdec.c
	libavformat/mpegts.c
	libavformat/mxfenc.c
	libavformat/rtpdec_g726.c
	libavformat/wtv.c
	libavformat/xmv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-01 02:54:24 +01:00
Martin Storsjö
2583660664 rtpdec: Add an init function that can do custom codec context initialization
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-30 17:32:18 +02:00
Michael Niedermayer
e161b079de Merge remote-tracking branch 'qatar/master'
* qatar/master: (22 commits)
  configure: add check for w32threads to enable it automatically
  rtmp: do not hardcode invoke numbers
  cinepack: return non-generic errors
  fate-lavf-ts: use -mpegts_transport_stream_id option.
  Add an APIchanges entry and a minor bump for avio changes.
  avio: Mark the old interrupt callback mechanism as deprecated
  avplay: Set the new interrupt callback
  avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
  cinepak: remove redundant coordinate checks
  cinepak: check strip_size
  cinepak, simplify, use AV_RB24()
  cinepak: simplify, use FFMIN()
  cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
  applehttp: Fix seeking in streams not starting at DTS=0
  http: Don't use the normal http proxy mechanism for https
  tls: Handle connection via a http proxy
  http: Reorder two code blocks
  http: Add a new protocol for opening connections via http proxies
  http: Split out the non-chunked buffer reading part from http_read
  segafilm: add support for raw videos
  ...

Conflicts:
	avconv.c
	configure
	doc/APIchanges
	libavcodec/cinepak.c
	libavformat/applehttp.c
	libavformat/version.h
	tests/lavf-regression.sh
	tests/ref/lavf/ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-19 02:00:06 +01:00
John Brooks
12348ca25e rtpdec: unwrap RTP timestamps for PTS calculation
The timestamp field in RTPDemuxContext was unused before this.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-18 10:31:17 +02:00
Michael Niedermayer
f884ef00de Merge remote-tracking branch 'qatar/master'
* qatar/master: (31 commits)
  tiffenc: initialize forgotten avctx.
  avplay: free the active audio packet at exit.
  avplay: free rdft data used for spectrogram analysis.
  log.h: make AVClass a named struct
  fix ac3 encoder documentation
  vc1: more prettyprinting cosmetics
  vc1: prettyprint some tables
  vc1: K&R formatting cosmetics
  AVOptions: bump minor and add APIchanges entry.
  cmdutils/avtools: simplify show_help() by using av_opt_child_class_next()
  AVOptions: rename FF_OPT_TYPE_* => AV_OPT_TYPE_*
  Remove all uses of deprecated AVOptions API.
  AVOptions: add av_opt_next, deprecate av_next_option.
  AVOptions: add functions for evaluating option strings.
  AVOptions: split get_number().
  AVOptions: add av_opt_get*, deprecate av_get*.
  AVOptions: add av_opt_set*().
  AVOptions: add new API for enumerating children.
  rv34: move inverse transform functions to DSP context
  flvenc: Write the right metadata entry count
  ...

Conflicts:
	avconv.c
	cmdutils.c
	doc/APIchanges
	ffplay.c
	ffprobe.c
	libavcodec/ac3dec.c
	libavcodec/h264.c
	libavcodec/libvpxenc.c
	libavcodec/libx264.c
	libavcodec/mpeg12enc.c
	libavcodec/options.c
	libavdevice/libdc1394.c
	libavdevice/v4l2.c
	libavfilter/vf_drawtext.c
	libavformat/flvdec.c
	libavformat/mpegtsenc.c
	libavformat/options.c
	libavutil/avutil.h
	libavutil/opt.c
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-13 06:00:03 +02:00
Martin Storsjö
bfc6db4477 rtpdec: Add ff_ prefix to all nonstatic symbols
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:12 +03:00
Michael Niedermayer
7b376b398a Merge remote branch 'qatar/master'
* qatar/master:
  Handle unicode file names on windows
  rtp: Rename the open/close functions to alloc/free
  Lowercase all ff* program names.
  Refer to ff* tools by their lowercase names.
NOT Pulled  Replace more FFmpeg instances by Libav or ffmpeg.
  Replace `` by $() syntax in shell scripts.
  patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled  Remove stray libavcore and _g binary references.
  vorbis: Rename decoder/encoder files to follow general file naming scheme.
  aacenc: Fix whitespace after last commit.
  cook: Fix small typo in av_log_ask_for_sample message.
  aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
  Remove RDFT dependency from AAC decoder.
  Add some debug log messages to AAC extradata
  Fix mov debug (u)int64_t format strings.
  bswap: use native types for av_bwap16().
  doc: FLV muxing is supported.
  applehttp: Handle AES-128 encrypted streams
  Add a protocol handler for AES CBC decryption with PKCS7 padding
  doc: Mention that DragonFly BSD requires __BSD_VISIBLE set

Conflicts:
	ffplay.c
	ffprobe.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-24 03:41:22 +02:00
Martin Storsjö
9261e6cf3f rtp: Rename the open/close functions to alloc/free
This avoids clashes if we internally want to override the global
open function.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-04-24 00:05:37 +03:00
Michael Niedermayer
d4d09329ee Merge remote branch 'qatar/master'
* qatar/master:
  lavf: bump minor and add an APIChanges entry for avformat cleanup
  lavf: get rid of ffm-specific stuff in avformat.h
Not pulled:  avio: deprecate av_protocol_next().
  avio: add a function for iterating though protocol names.
  lavf: rename a parameter of av_sdp_create from buff->buf
  lavf: rename avf_sdp_create to av_sdp_create.
  lavf: make av_guess_image2_codec internal
  avio: make URLProtocol internal.
  avio: make URLContext internal.
  lavf: mark av_pkt_dump(_log) for remove on $next+1 bump.
  lavf: use designated initializers for all protocols
  applehttp: don't use deprecated url_ functions.
  avio: move two ff_udp_* functions from avio_internal to url.h
  asfdec: remove a forgotten declaration of nonexistent function
  avio: deprecate the typedef for URLInterruptCB

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-09 03:17:07 +02:00
Anton Khirnov
c486dade95 avio: make URLContext internal. 2011-04-08 15:59:19 +02:00
Mans Rullgard
2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Diego Elio Pettenò
53d3f66bae Make RTPFirstDynamicPayloadHandler static to rtpdec.c
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 119cc033fc)
2011-01-26 03:43:30 +01:00
Diego Elio Pettenò
119cc033fc Make RTPFirstDynamicPayloadHandler static to rtpdec.c
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-25 01:45:34 +00:00
Martin Storsjö
3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-01 22:27:16 +00:00
Martin Storsjö
1e515c4280 rtpdec: Add functions for finding depacketizers by name or payload id
Originally committed as revision 25891 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:39:50 +00:00
Martin Storsjö
2beb910d79 rtpdec: Allow dynamic payload handlers to handle static payload types, too
Originally committed as revision 25890 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:38:55 +00:00
Aurelien Jacobs
a178edf298 drop rtp_get_file_handles() which is not part of public API and not used anymore
Originally committed as revision 25556 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-23 16:21:49 +00:00
Aurelien Jacobs
fb660f4015 drop rtp_get_local_port() which is not part of public API and not used anymore
Originally committed as revision 25555 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-23 16:20:56 +00:00
Martin Storsjö
58ee09911e rtpdec: Reorder received RTP packets according to the seq number
Reordering is enabled only when receiving over UDP.

Originally committed as revision 25294 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:50:24 +00:00
Martin Storsjö
ad4ad27fb6 rtsp/rtpdec: Allow rtp_parse_packet to take ownership of the packet buffer
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.

Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:43:27 +00:00
Josh Allmann
186f1ec5f4 Add rtp_get_rtcp_file_handle function
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 24929 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 17:32:59 +00:00
Martin Storsjö
1537817e38 Move SPACE_CHARS back to libavformat/internal.h
It will be used by other parts of lavf now. This reverts svn rev 23846.

Originally committed as revision 24265 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-16 14:12:52 +00:00
Martin Storsjö
d74c6145fb rtpdec: Allow depacketizers to specify that pkt->pts should be left as AV_NOPTS_VALUE
Originally committed as revision 24234 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-14 12:26:16 +00:00
Josh Allmann
016bc031eb rtpdec: Add generic function for iterating over FMTP configuration lines
This will be used for cleaning up code that is common among RTP depacketizers.

Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-28 11:24:12 +00:00
Josh Allmann
8b114d85ba rtpdec: Move space_chars from avformat/internal to rtpdec
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23846 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-28 11:21:34 +00:00
Josh Allmann
ca937a5508 RTSP, rtpdec: Move RTPPayloadData into rtpdec_mpeg4 and remove all references to rtp_payload_data in rtpdec and rtsp
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23772 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-25 08:02:50 +00:00
Martin Storsjö
5948f82227 Reset RTCP timestamps after seeking, add range start offset to the packets timestamps
If these aren't reset, the timestamps make a huge jump when the next RTCP
is received.

Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-20 07:38:52 +00:00
Martin Storsjö
2cab6b48ad Revert svn rev 21857, readd first_rtcp_ntp_time in RTPDemuxContext
In order to sync RTP streams that get their initial RTCP timestamp at
different times, propagate the NTP timestamp of the first RTCP packet
to all other streams.

This makes the timestamps of returned packets start at (near) zero instead
of at any random offset.

Originally committed as revision 22917 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-20 07:34:28 +00:00
Luca Barbato
9094d867aa Make rtp protocol obey rfc3550
Originally committed as revision 22906 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-19 11:40:45 +00:00
Martin Storsjö
0e4b185a8d Fix leaks in the AAC RTP depacketizer
Originally committed as revision 22804 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-05 17:25:39 +00:00
Ronald S. Bultje
1da1e1933f Add void to a function prototype that takes no arguments. Fixes a potential
compiler warning.

Originally committed as revision 22803 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-05 16:13:51 +00:00
Stefano Sabatini
72415b2adb Define AVMediaType enum, and use it instead of enum CodecType, which
is deprecated and will be dropped at the next major bump.

Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 23:30:55 +00:00
Ronald S. Bultje
3307e6ea86 Prefix non-static RTSP functions with ff_.
Originally committed as revision 21974 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-02-23 00:35:50 +00:00
Ronald S. Bultje
fc78b0cb7e Remove first_rtcp_ntp_time. This is used to prevent overflow of the timestamp,
but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.

See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.

Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-02-16 23:00:03 +00:00
Martin Storsjö
9c8fa20d7e When using RTP-over-UDP, send dummy packets during stream setup, similar to
what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.

Patch by Martin Storsjö <$firstname at $firstname dot st>.

Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-02-16 22:50:50 +00:00
Ronald S. Bultje
f0a8039464 Add url_get_file_handle(), which is used to get the file descriptor
associated with the I/O handle (e.g. the fd returned by open()). See
"[RFC] rtsp.c EOF support" thread.

There were previously some URI-specific implementations of the same idea,
e.g. rtp_get_file_handles() and udp_get_file_handle(). All of these are
deprecated by this patch and will be removed at the next major API bump.

Originally committed as revision 17779 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-03-03 17:04:51 +00:00
Ronald S. Bultje
144ae29dde Implement marker bit, which is used for several RTP payloads currently
under review. See "[FFmpeg-devel] RTP mark bit not passed to parse_packet"
thread on mailinglist.

Originally committed as revision 17616 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-26 14:24:50 +00:00
Ronald S. Bultje
a68d44ed6f Don't (ab)use PKT_FLAG_* in RTP code, since the two have virtually nothing
in common except for this one value. Change was requested by Luca in the
"[FFmpeg-devel] RTP mark bit not passed to parse_packet" thread.

Originally committed as revision 17615 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-26 14:23:05 +00:00
Luca Abeni
e5f483c604 Move to rtp.h the prototypes of all the functions defined in rtp.c
Originally committed as revision 17389 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-17 08:09:34 +00:00
Luca Abeni
18c7b354c5 Move RTP_MAX_PACKET_LENGTH to rtpdec.h: it is not used in the muxer anymore
Originally committed as revision 17023 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-06 20:42:21 +00:00
Luca Abeni
302879cb36 Split rtp.h in rtp.h, rtpdec.h, and rtpenc.h
Originally committed as revision 17016 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-06 10:35:52 +00:00