Commit Graph

8254 Commits

Author SHA1 Message Date
Diego Biurrun
ffae713a5b Fix a bunch of common typos. 2012-03-09 22:02:49 +01:00
Alex Converse
100c3fb2d1 mpegts: Always honor a registration descriptor if present and there is no other codec information. 2012-03-09 09:48:14 -08:00
Martin Storsjö
6294d708b8 rtsp: Only set the ttl parameter if the server actually gave a value
Passing ttl=0 to the rtp/udp url contexts makes packets never
leave the host machine.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:04:32 +02:00
Martin Storsjö
2bfd92b330 udp: Set ttl for read-write streams, too, not only for write-only ones
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:04:05 +02:00
Martin Storsjö
c700fdb00f udp: Only bind to the multicast address if in read-only mode
This fixes sending back RTCP RR packets if receiving RTP over
multicast.

If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.

This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:03:46 +02:00
Martin Storsjö
1b89bcdd7f udp: Clarify the comment about binding the multicast address
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:03:11 +02:00
Martin Storsjö
113d3e106d udp: Reorder comments
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:03:10 +02:00
Dale Curtis
ef0d779706 Fix uninitialized reads on malformed ogg files.
The ogg decoder wasn't padding the input buffer with the appropriate
FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in
various pieces of parsing code when they thought they had more data than
they actually did.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2012-03-08 11:52:15 -08:00
Martin Storsjö
94f1b11a6f rtpenc: Fix the AVRational used for av_rescale_q_rnd
The current one has a zero denominator - this is what was
intended in 14aecc50fa.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-08 01:15:28 +02:00
Martin Storsjö
a887c87c23 udp: Print an error message if bind fails
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-07 21:52:19 +02:00
Ronald S. Bultje
a93b572ae4 smacker: error out if palette copy-with-offset overruns palette size.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2012-03-07 09:35:03 -08:00
Carl Eugen Hoyos
a294a7a1b3 mov: Allow last chunk to have an arbitrary number of samples.
Fixes ticket #673.
(cherry picked from commit 8dcd2a41ec)

Signed-off-by: Alex Converse <alex.converse@gmail.com>
2012-03-06 15:25:34 -08:00
Reimar Döffinger
632eb1bbae cdxl demux: do not create packets with uninitialized data at EOF.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
2012-03-05 16:27:31 -05:00
Justin Ruggles
94cf64b81f cosmetics: reindent 2012-03-05 13:08:19 -05:00
Justin Ruggles
8c1d6ac66a avformat: do not require a pixel/sample format if there is no decoder
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.

Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
2012-03-05 13:08:18 -05:00
Justin Ruggles
a7fa75684d avformat: do not fill-in audio packet duration in compute_pkt_fields()
Use the estimated duration only to calculate missing timestamps if needed.
2012-03-05 13:08:18 -05:00
Justin Ruggles
6c65cf58fd lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
Also, do not give AVCodecContext.frame_size priority for muxing.

Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
             by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
                 using the packet size and average bit rate.
2012-03-05 13:08:18 -05:00
Justin Ruggles
f1e73100d9 siff: do not set AVCodecContext.frame_size
also, properly set AVCodecContext.bits_per_coded_sample, AVStreasm.start_time,
and AVPacket.duration.
2012-03-05 13:08:17 -05:00
Justin Ruggles
ec2e767bf3 amr demuxer: do not set AVCodecContext.frame_size.
it is not necessary.
2012-03-05 13:08:17 -05:00
Justin Ruggles
8d1a20aa7c aiffdec: do not set AVCodecContext.frame_size
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.

Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
2012-03-05 13:08:17 -05:00
Justin Ruggles
237a855caf mov: do not set AVCodecContext.frame_size
It is not necessary.
2012-03-05 13:08:17 -05:00
Justin Ruggles
9727264220 ape: do not set AVCodecContext.frame_size.
prevents lavf from setting incorrect packet durations.
2012-03-05 13:08:17 -05:00
Justin Ruggles
2dd18d4435 rdt: remove workaround for infinite loop with aac
avformat_find_stream_info() no longer hangs while waiting for AAC frame_size
2012-03-05 13:08:16 -05:00
Justin Ruggles
9c365fe8ae avformat: do not require frame_size in avformat_find_stream_info() for CELT
In Ogg/CELT, frame_size is found in the same place as the sample_rate and
channels, so we do not need to force the frame_size to be parsed.
2012-03-05 13:08:16 -05:00
Justin Ruggles
fbc8c59679 avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
It was only needed to avoid a bad time base (and thus non-monotone timestamps)
for stream copy to avi.
2012-03-05 13:08:16 -05:00
Justin Ruggles
84b6ae0808 avformat: do not require frame_size in avformat_find_stream_info() for AAC
We already will get the needed info because of CODEC_CAP_CHANNEL_CONF
2012-03-05 13:08:16 -05:00
Justin Ruggles
620b88a302 swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size
This way we can do stream copy without having the demuxer wait until
frame_size has been set.
2012-03-05 13:08:16 -05:00
Justin Ruggles
14aecc50fa rtpenc: use av_get_audio_frame_duration() for max_frames_per_packet
It is more reliable than AVCodecContext.frame_size for codecs with constant
packet duration.
2012-03-05 13:08:16 -05:00
Justin Ruggles
c019070fda riffenc: use av_get_audio_frame_duration()
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
2012-03-05 13:08:15 -05:00
Anton Khirnov
27c7ca9c12 lavf: deobfuscate read_frame_internal().
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.

The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.

compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
2012-03-05 18:47:05 +01:00
Anton Khirnov
dcee811505 lavf: make read_from_packet_buffer() more flexible.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.

Also move the function higher in the file, since it will be called from
read_frame_internal().
2012-03-05 18:44:45 +01:00
Anton Khirnov
52b0943f10 lavf: factorize freeing a packet buffer. 2012-03-05 18:44:30 +01:00
Diego Biurrun
0a41f47dc1 dv: Do not redundantly initialize struct members to zero. 2012-03-05 17:02:59 +01:00
Justin Ruggles
b7beabab4b tiertexseq: set correct block_align for audio 2012-03-03 17:03:27 -05:00
Justin Ruggles
f9cf91d822 tiertexseq: set audio stream start time to 0
Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.
2012-03-03 17:03:27 -05:00
Justin Ruggles
0883109b27 voc/avs: Do not change the sample rate mid-stream.
Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.
2012-03-03 17:03:27 -05:00
Justin Ruggles
4da374f8a9 segafilm: use the sample rate as the time base for audio streams 2012-03-03 17:03:27 -05:00
Justin Ruggles
ea289186f0 ea: fix audio pts
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
2012-03-03 17:03:27 -05:00
Justin Ruggles
01be6fa926 psx-str: fix audio pts
Each packet has 18 sectors with 224/channels samples in each sector.
2012-03-03 17:03:27 -05:00
Justin Ruggles
d0ab585074 vqf: set packet duration
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
2012-03-03 17:03:26 -05:00
Justin Ruggles
101c369b7c tta demuxer: set packet duration 2012-03-03 17:03:26 -05:00
Justin Ruggles
5a9b952201 thp: set audio packet durations 2012-03-03 16:58:45 -05:00
Justin Ruggles
5602a464c9 avcodec: add a Vorbis parser to get packet duration
This also allows for removing some of the Vorbis-related hacks.
2012-03-03 16:43:11 -05:00
Alex Converse
1aa708988a mpegts: Pad the packet buffer in handle_packet().
This allows it to be used with get_bits without the thread of overreads.

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2012-03-02 15:44:42 -08:00
Alex Converse
4df369692e mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2012-03-02 15:44:42 -08:00
Ronald S. Bultje
9c239f6026 matroska: check buffer size for RM-style byte reordering.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
2012-03-02 10:32:22 -08:00
Alex Converse
1697c29d75 rmdec: Honor .RMF tag size rather than assuming 18. 2012-03-02 09:31:32 -08:00
Anton Khirnov
56bf24ad78 r3d: don't set codec timebase.
It's not supposed to be set by demuxers.

Set avg_frame_rate and r_frame_rate instead.
2012-03-02 17:21:45 +01:00
Anton Khirnov
efec3bc65a electronicarts: set timebase for tgv video.
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
2012-03-02 11:11:38 +01:00
Anton Khirnov
e39400c3a8 electronicarts: parse the framerate for cmv video. 2012-03-02 11:11:38 +01:00