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>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change was dependent on a different patch that
never actually made it into FFmpeg, and it actually
ended up breaking builds.
This reverts commit 70cf7bb958899c369f778e29b8bd4203723bed04.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When skipping over the extended header, take into account
that the size field has already been read. The extended header
also takes up space, so adjust total header length accordingly.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
* cus/stable:
ffplay: Copy audio side data too. This fixes handling of some rare nellymoser files that change the sample rate mid stream (sample file at: http://trac.videolan.org/vlc/ticket/5586)
Merged-by: Michael Niedermayer <michaelni@gmx.at>
this file uses the M_PI macro since
4e74187db2f5db52f88729efc662df9d6bc763e1, so include the correct header
directly.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
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>
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>
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>
