* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Janne Grunau [Fri, 21 Oct 2011 15:25:30 +0000 (17:25 +0200)]
latmdec: fix audio specific config parsing
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
swresample_test.c:123:21: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement]
swresample_test.c:127:25: warning: ISO C90 forbids mixed declarations and code [-Wdeclaration-after-statement]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/swscale.c:2744:40: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2745:41: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2746:41: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/swscale.c:2747:78: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libswscale/x86/swscale_mmx.c:131:36: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/x86/swscale_mmx.c:132:37: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
libswscale/x86/swscale_mmx.c:133:74: warning: to be safe all intermediate pointers in cast from ‘int16_t **’ to ‘const int16_t **’ must be ‘const’ qualified [-Wcast-qual]
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Justin Ruggles [Mon, 14 Nov 2011 21:07:03 +0000 (16:07 -0500)]
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Add AV_NUM_DATA_POINTERS to simplify the bump transition.
This will allow for supporting more planar audio channels without having to
allocate separate pointer arrays.
Mans Rullgard [Tue, 29 Nov 2011 18:38:02 +0000 (18:38 +0000)]
Do not memcpy raw video frames when using null muxer
Commit 035af99 made avconv always call an encoder when using the
null muxer. While useful for 2-pass encodes, it inadvertently
caused an extra memcpy of raw frames when decoding only.
This hack restores the old behaviour when only decoding while
allowing use of the null muxer with encoded streams as well.
Martin Storsjö [Fri, 2 Dec 2011 09:06:18 +0000 (11:06 +0200)]
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
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.
* 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
...
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>
Martin Storsjö [Thu, 1 Dec 2011 08:45:26 +0000 (10:45 +0200)]
rtpdec: Interpret the different G726 names as bits_per_coded_sample
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.
* 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)
libx264: Implement rgb24 support through a seperate AVCodec.
This avoids people mistakely encoding in a way that many players dont support.
Fixes Ticket658
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Marton Balint <cus@passwd.hu>
Reimar Döffinger [Sat, 26 Mar 2011 15:25:10 +0000 (16:25 +0100)]
Add PlayStation Portable PMP format demuxer
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.
Martin Storsjö [Thu, 1 Dec 2011 09:24:23 +0000 (11:24 +0200)]
http: Make sure proxyauth is initialized
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.
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Miroslav Slugeň [Mon, 7 Nov 2011 11:13:55 +0000 (12:13 +0100)]
rtpdec: Add support for G726 audio
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).
projects / ffmpeg / log