(issue1210)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes trac #1045.

Thanks to Peter Ross for his help with this patch.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Bug introduced by libav in 1bb3990b56

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Fixes Ticket385

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This allows future assumptions to be made without affecting non-intra files.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The "ECs != 1 -> OP1a" assumption was wrong. Luckily, the file that triggered
that behavior had two ECs, not zero. Hence distinguishing between them is
simple in this case.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This fixes rare cases where OPAtom may be treated as OP1a, causing all essence
to be read into RAM.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

The current one has a zero denominator - this is what was
intended in 14aecc50fa.

Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Martin Storsjö <martin@martin.st>

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org

Fixes ticket #673.
(cherry picked from commit 8dcd2a41ec)

Signed-off-by: Alex Converse <alex.converse@gmail.com>

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>

All colorspaces are supported.

Renamed libutvideo.cpp to libutvideodec.cpp.

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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.

Use the estimated duration only to calculate missing timestamps if needed.

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.

also, properly set AVCodecContext.bits_per_coded_sample, AVStreasm.start_time,
and AVPacket.duration.

it is not necessary.

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.

It is not necessary.

prevents lavf from setting incorrect packet durations.

avformat_find_stream_info() no longer hangs while waiting for AAC frame_size

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.

It was only needed to avoid a bad time base (and thus non-monotone timestamps)
for stream copy to avi.

We already will get the needed info because of CODEC_CAP_CHANNEL_CONF

This way we can do stream copy without having the demuxer wait until
frame_size has been set.

It is more reliable than AVCodecContext.frame_size for codecs with constant
packet 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.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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.

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().

This fixes issues when the bitrate is variable or inaccurate but the
frame size has not been determined yet.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Update FATE test to reflect delayed video due to the file having audio-only
frames prior to the first frame with video.

Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.

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.

Each packet has 18 sectors with 224/channels samples in each sector.

Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.

This also allows for removing some of the Vorbis-related hacks.