Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Having a somehow off seeking is better than having none at all.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

The asf_read_pts should read the bitstream directly.

The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

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

For the standardized 8 kHz sample rate, this works exactly the same.
For nonstandard sample rates, the different predefined G726
names (G726-16, G726-24, G726-32, G726-40) are interpreted as an
indication of the bits per coded sample, even though their
actual bitrates aren't what the name specifies.

This feels more sane than using free-form names for nonstandard
sample rate/bitrate combinations, e.g like G726-22, G726-33
for 11025 Hz.
Signed-off-by: Martin Storsjö <martin@martin.st>

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

This avoids overflow if frame_size is over 2147, since both
frame_size and AV_TIME_BASE are plain integers.
Signed-off-by: Martin Storsjö <martin@martin.st>

This prevents frame allocation overflows, and fixed
fate-h264-conformance-mr3_tandberg_b with 2 threads.

Signed-off-by: Mans Rullgard <mans@mansr.com>

This makes whitespace and register names consistent with
the style used in more recent code.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Although this adds a few lines, the macro calls are less convoluted.
Signed-off-by: Mans Rullgard <mans@mansr.com>

this removes 2 redundant tests for pcm in mkv.
we can add the coverage back in later as fate-lavf tests if needed.

this file uses the M_PI macro since
4e74187d, so include the correct header
directly.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>

This is needed for optimised transforms.
Signed-off-by: Mans Rullgard <mans@mansr.com>

Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>

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

This simplifies the open functions by avoiding one function
call that needs error checking, reducing the amount of
extra bulk code.
Signed-off-by: Martin Storsjö <martin@martin.st>

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

This prevents memory leaks if this function returns an error.
Signed-off-by: Martin Storsjö <martin@martin.st>

The private data pointer isn't a file handle, this protocol
doesn't have any file handle to return.
Signed-off-by: Martin Storsjö <martin@martin.st>

This should clean up leaked memory.
Signed-off-by: Martin Storsjö <martin@martin.st>

This string will be passed to ff_http_auth_create_response
even if no proxy is used, resulting in reading uninitialized
memory. The other auth string is always initialized by
av_url_split.
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

There's no reason to use two arrays for this.

Based off commit 2fea60c6
to FFmpeg by Michael Niedermayer.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>

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

Signed-off-by: Mans Rullgard <mans@mansr.com>

It's supposed to be called only from (de)muxers.

This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>

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

I.e. on streamcopy set output codec timebase from input stream timebase
(as opposed to input codec timebase). This should be more sane, because
since the stream is not decoded, the input codec tb has no relation to
the timestamps of the copied packets.

It's not supposed to be set outside of lavc.

It's not supposed to be set outside of lavc.
Set r_frame_rate instead.