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>

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.

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().

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.

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

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

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

It's not supposed to be set by demuxers.

Set avg_frame_rate and r_frame_rate instead.

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.

Demuxers are not supposed to set it.

Demuxers are not supposed to set it.
Set stream timebase and framerates instead (this is a cfr container with
no timestamps).

Demuxers are not supposed to set it.
Set r_frame_rate and avg_frame_rate instead.

The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes"
come first, "Frames" behind.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>

libavformat/rmdec.c:383: warning: format ‘%d’ expects type ‘int’, but argument 7 has type ‘int64_t’

This makes the packetization spec compliant for cases where one single
GOB doesn't fit into an RTP packet.

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

Otherwise the values are overwritten.

This fixes cases where the user had specified one desired MTU
via an option, and the protocol indicates another one.

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