Commit Graph

8634 Commits

Author SHA1 Message Date
Anton Khirnov
782e64fbe1 wv,mpc8: don't return apetag data in packets. 2012-07-31 20:53:54 +02:00
Samuel Pitoiset
9c9c21eaa1 rtmp: do not warn about receiving metadata packets
They are managed in get_packet()

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2012-07-31 14:52:25 +02:00
Diego Biurrun
0177b7d23a Improve descriptiveness of a number of codec and container long names 2012-07-30 20:46:55 +02:00
Diego Biurrun
6774247a9d avformat: Drop pointless "format" from container long names 2012-07-30 13:59:06 +02:00
Anton Khirnov
29d70274ec wv: K&R formatting cosmetics 2012-07-30 00:42:20 +02:00
Luca Barbato
41f43202cf flvdec: remove spurious use of stream id
We match streams by codec id now.
2012-07-29 17:18:03 +02:00
Anton Khirnov
aba232cfa9 lavf: deprecate r_frame_rate.
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.

Replace it with the average framerate where it makes sense.

FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.

In some other tests lavf starts making up frame durations from different
frame.
2012-07-29 08:06:30 +02:00
Anton Khirnov
f66eeff1c8 lavf: round estimated average fps to a "standard" fps. 2012-07-29 08:05:46 +02:00
Anton Khirnov
fe1c1198e6 lavf: use dts difference instead of AVPacket.duration in find_stream_info()
AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.

The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
2012-07-29 08:04:42 +02:00
Luca Barbato
681ed00099 avf: introduce nobuffer option
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.

An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.

Additional changes by Josh Allmann <joshua.allmann@gmail.com>

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-07-29 07:58:00 +02:00
Anton Khirnov
c1d865d563 wv: return meaningful error codes. 2012-07-28 14:37:16 +02:00
Anton Khirnov
ccc10acb5b wv: return AVERROR_EOF on EOF, not EIO. 2012-07-28 14:37:12 +02:00
Anton Khirnov
f73e3938ac mp3dec: forward errors for av_get_packet().
Don't invent a bogus EIO error.

The code now doesn't check for ret == 0, but that check is redundant,
av_get_packet() never returns 0.
2012-07-28 14:37:00 +02:00
Anton Khirnov
67b1156fe8 mp3dec: remove a pointless local variable. 2012-07-28 14:36:55 +02:00
Anton Khirnov
61f8bb74f3 mp3dec: remove commented out cruft. 2012-07-28 14:36:51 +02:00
Anton Khirnov
efd34918ba lavf: remove commented out cruft in avformat_find_stream_info() 2012-07-28 10:00:38 +02:00
Anton Khirnov
c4ef6a3e4b Add missing libavutil/time.h includes. 2012-07-28 09:02:07 +02:00
Martin Storsjö
8ebacfb598 hls: Proceed to the next segment at any error code
Previously, we returned any error code except AVERROR_EOF to the
caller - only if AVERROR_EOF or 0 was returned, we proceeded to
the next segment.

With some setups of web servers, using Connection: close in https
and GnuTLS, we don't get a clean error code at the end of segments.
In those cases, just proceed to the next segment.

Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-28 01:21:32 +03:00
Martin Storsjö
41ecbbc7aa tls: Return AVERROR_EOF if the TLS_read/write functions return 0
OpenSSL returns 0 when the peer has closed the connection. GnuTLS
doesn't return that though, but returns
GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed
without a clean close notify packet.

Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-28 01:21:16 +03:00
Samuel Pitoiset
8ea1459bc3 rtmp: Check the buffer length of ping packets
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-26 22:57:15 +03:00
Samuel Pitoiset
e49e6b6451 rtmp: Allow having more unknown data at the end of a chunk size packet without failing
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-26 22:56:54 +03:00
Samuel Pitoiset
2357f60687 rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-26 22:56:54 +03:00
Antti Seppälä
5423e908c9 Support urlencoded http authentication credentials
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.

Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-26 00:18:32 +03:00
Samuel Pitoiset
abf77a247b rtmp: Return an error when the client bandwidth is incorrect
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 23:57:01 +03:00
Samuel Pitoiset
be8f949219 rtmp: Return proper error code in handle_server_bw
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 23:57:00 +03:00
Samuel Pitoiset
088a82bb33 rtmp: Return proper error code in handle_client_bw
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 23:57:00 +03:00
Samuel Pitoiset
e7ea6883bf rtmp: Return proper error codes in handle_chunk_size
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 23:57:00 +03:00
Samuel Pitoiset
6d1c9945dd rtmp: Factorize the code by adding handle_invoke
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 21:08:24 +03:00
Samuel Pitoiset
7be2a7d8ff rtmp: Factorize the code by adding handle_chunk_size
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 21:08:22 +03:00
Samuel Pitoiset
0ffd5161c4 rtmp: Factorize the code by adding handle_ping
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 21:08:20 +03:00
Samuel Pitoiset
912ecc9a19 rtmp: Factorize the code by adding handle_client_bw
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 21:08:18 +03:00
Samuel Pitoiset
9b498148ca rtmp: Factorize the code by adding handle_server_bw
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 21:08:09 +03:00
Samuel Pitoiset
758377a2b7 rtmp: Add a new option 'rtmp_pageurl'
This option specifies the URL of the web page in which the media
was embedded.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 20:33:13 +03:00
Samuel Pitoiset
63ffa154e9 rtmp: Make the description of the rtmp_tcurl option more generic
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 20:33:10 +03:00
Jordi Ortiz
ecfff0e992 sctp: add port missing error message
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 20:31:16 +03:00
Jordi Ortiz
f9a9a14862 tcp: add port missing error message
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-25 20:30:39 +03:00
Martin Storsjö
6a433fdba8 rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
Our implementation of RTMPE is heavily based on librtmp.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-24 18:13:26 +03:00
Samuel Pitoiset
f7bfb126cd rtmp: Move the CONFIG_ condition into the if conditions
This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-24 15:54:10 +03:00
Diego Biurrun
6b80142144 build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
The ffrtmpcrypt protocol depends on external libraries, which are
also required to compile the header file.
2012-07-24 00:21:39 +02:00
Adriano Pallavicino
999c63e4ca rtp: Only choose static payload types if the sample rate and channels are right
If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.

G722 is a special case where the normal rules don't apply.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-24 00:42:58 +03:00
Clément Bœsch
61884b9d1b wav: init st to NULL to avoid a false-positive warning.
If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.

Fixes warning:
    libavformat/wav.c: In function ‘wav_read_header’:
    libavformat/wav.c:499:44: warning: ‘st’ may be used uninitialized in this function [-Wmaybe-uninitialized]

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-07-23 11:34:08 -04:00
Samuel Pitoiset
08cd95e8a3 RTMPTE protocol support
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-23 16:32:09 +03:00
Samuel Pitoiset
acd554c103 RTMPE protocol support
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-23 16:32:07 +03:00
Samuel Pitoiset
0e31088b6c rtmp: Add ff_rtmp_calc_digest_pos()
This function is used for calculating digest position for RTMP handshake
packets.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-23 16:32:05 +03:00
Samuel Pitoiset
3505d5574e rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-23 16:32:03 +03:00
Ronald S. Bultje
fd48721849 lavf: use conditional notation for default codec in muxer declarations.
This removes the use of macro nesting in these code constructs, which
makes it easier to parse in pre-processors.
2012-07-22 16:10:21 -07:00
Anton Khirnov
721113bed2 matroskadec: return more correct error code on read error. 2012-07-22 09:14:05 +02:00
Kostya Shishkov
1470ce21ce Bump libavcodec and libavformat minor versions for G.723.1 decoder and demuxer 2012-07-22 08:43:12 +02:00
Mohamed Naufal Basheer
55c3a4f617 G.723.1 demuxer and decoder
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
2012-07-22 07:58:54 +02:00
Ronald S. Bultje
5354a904fe rtsp: remove terminal comma in FF_RTP_FLAG_OPTS macro.
This makes usage of the macro look more natural when
used with array entries.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2012-07-21 15:35:01 -04:00