This makes the function accept the format of creation_time
as output by demuxers (e.g. the mov demuxer), making the
creation timestamp stay intact if transcoding.
Signed-off-by: Martin Storsjö <martin@martin.st>
This function is used in muxers for parsing the 'creation_time'
metadata key, for converting it to a time value.
This makes it match the behaviour of the exported 'creation_time'
metadata from demuxers, where it is in UTC, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
Converting to double before the multiplication rather than after
avoids an integer overflow in some cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The Apple HTTP Live Streaming demuxer's implementation of seeking searches for
the MPEG TS segment which contains the requested timestamp. In its current
implementation it assumes that the first segment will start from 0.
But, MPEG TS streams do not necessarily start with timestamp (near) 0, causing
seeking to fail for those streams.
This also occurs when using live streaming of HTTP Live Streams. In this case
sliding playlists may be used, which means that in that case only the last x
encoded segments are stored, the earlier segments get deleted from disk and
removed from the playlist. Because of this, when starting playback of a stream
in the middle of such a broadcast, the initial segment fetched after parsing
the m3u8 playlist will not start from timestamp (near) 0, causing (the
admittedly limited live) seeking to fail.
This patch changes this demuxers seeking implementation to use the initial DTS
as an offset for searching the segments containing the requested timestamp.
* qatar/master:
binkvideo: simplify and remove invalid shifts
pulse: compute frame_duration once and fix it
lavf: simplify format_child_class_next()
hwaccel: OS X Video Decoder Acceleration (VDA) support.
doc: add support for an optional navigation bar in texi2html pages
Conflicts:
configure
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/vda.c
libavcodec/vda.h
libavcodec/vda_h264.c
libavcodec/vda_internal.h
libavcodec/version.h
libavformat/options.c
libavutil/avutil.h
libavutil/pixfmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: pass options from AVFormatContext to avio.
avformat: Use avio_open2, pass the AVFormatContext interrupt_callback onwards
avio: add avio_open2, taking an interrupt callback and options
avio: add support for passing options to protocols.
avio: add and use ffurl_protocol_next().
avformat: Pass the interrupt callback on to chained muxers/demuxers
avio: Add an AVIOInterruptCB parameter to ffurl_open/ffurl_alloc
avformat: Use ff_check_interrupt
avio: Add an internal utility function for checking the new interrupt callback
avio: Add AVIOInterruptCB
texi2html: remove stray \n
doc: prettyfy the texi2html documentation
swscale: handle unaligned buffers in yuv2plane1
Conflicts:
libavformat/avformat.h
libavformat/avio.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The interrupt callback has to be passed in during opening (setting it
after opening isn't enough), since a blocking open couldn't be
interrupted otherwise.
Options are passed down to procotols and also need to be available
during open() in most cases.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This is a better io interrupt callback function, which has an
opaque parameter, which is given to the interrupt callback.
This allows callers to precisely cancel IO for one single
AVFormatContext, without interrupt other ones in the same
process.
Note, it's not needed in AVIOContext, at the moment.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
vble: remove vble_error_close
VBLE Decoder
tta: use an integer instead of a pointer to iterate output samples
shorten: do not modify samples pointer when interleaving
mpc7: only support stereo input.
dpcm: do not try to decode empty packets
dpcm: remove unneeded buf_size==0 check.
twinvq: add SSE/AVX optimized sum/difference stereo interleaving
vqf/twinvq: pass vqf COMM chunk info in extradata
vqf: do not set bits_per_coded_sample for TwinVQ.
twinvq: check for allocation failure in init_mdct_win()
swscale: add padding to conversion buffer.
rtpdec: Simplify finalize_packet
http: Handle proxy authentication
http: Print an error message for Authorization Required, too
AVOptions: don't return an invalid option when option list is empty
AIFF: add 'twos' FourCC for the mux/demuxer (big endian PCM audio)
Conflicts:
libavcodec/avcodec.h
libavcodec/tta.c
libavcodec/vble.c
libavcodec/version.h
libavutil/opt.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add a decoder for the VBLE Lossless Codec, which
still has a cult following. Used to be popular
several years ago on doom9.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This is needed because the twinvq decoder cannot rely on bit_rate to be set.
The API documentation says that bit_rate is set by libavcodec, not by the
user.
Tested with both Basic and Digest authentication, and tested with
both proxy authentication and authentication for the requested
resource at the same time.
Signed-off-by: Martin Storsjö <martin@martin.st>
The error was hidden before, to avoid showing an error on the
first request where no auth has been provided, when the server
indicates which authentication method to use.
Now the error is printed if an authentication method was used,
but failed.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (29 commits)
doc: update libavfilter documentation
tls: Use the URLContext as logging context
aes: Avoid illegal read and don't generate more key than we use.
mpc7: Fix memset call in mpc7_decode_frame function
atrac1: use correct context for av_log()
apedec: consume the whole packet when copying to the decoder buffer.
apedec: do not needlessly copy s->samples to nblocks.
apedec: check output buffer size after calculating actual output size
apedec: remove unneeded entropy decoder normalization.
truespeech: use memmove() in truespeech_update_filters()
vorbisdec: remove AVCODEC_MAX_AUDIO_FRAME_SIZE check
vorbisdec: remove unneeded buf_size==0 check
vorbisdec: return proper error codes instead of made-up ones
http: Don't add a Range: bytes=0- header for POST
sunrast: Check for invalid/corrupted bitstream
http: Change the chunksize AVOption into chunked_post
http: Add encoding/decoding flags to the AVOptions
avconv: remove some codec-specific hacks
crypto: add decoding flag to options.
tls: use AVIO_FLAG_NONBLOCK instead of deprecated URL_FLAG_NONBLOCK
...
Conflicts:
doc/libavfilter.texi
libavcodec/atrac1.c
libavcodec/sunrast.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The chunksize internal variable has two different uses - for
reading, it's the amount of data left of the current chunk
(or -1 if the server doesn't send data in chunked mode), where
it's only an internal state variable. For writing, it's used
to decide whether to enable chunked encoding (by default), by
using the value 0, or disable chunked encoding (value -1).
This, while consistent, doesn't make much sense to expose
as an AVOption. This splits the usage of the internal variable
into two variables, chunksize which is used for reading (as
before), and chunked_post which is the user-settable option,
with the values 0 and 1, where 1 is default.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avcodec: add support for planar signed 8-bit PCM.
ra144enc: add sample_fmts list to ff_ra_144_encoder
smackaud: use uint8_t* for 8-bit output buffer type
smackaud: clip output samples
smackaud: use sign_extend() for difference value instead of casting
sipr: use a function pointer to select the decode_frame function
sipr: set mode based on block_align instead of bit_rate
sipr: do not needlessly set *data_size to 0 when returning an error
ra288: fix formatting of LOCAL_ALIGNED_16
udp: Allow specifying the local IP address
VC1: Add bottom field offset to block_index[] to avoid rewriting (+10L)
vc1dec: move an if() block.
vc1dec: use correct hybrid prediction threshold.
vc1dec: Partial rewrite of vc1_pred_mv()
vc1dec: take ME precision into account while scaling MV predictors.
lavf: don't leak corrupted packets
Conflicts:
libavcodec/8svx.c
libavcodec/ra288.c
libavcodec/version.h
libavformat/iff.c
libavformat/udp.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is found in some 8svx files (e.g. ones created by SoX).
Currently the decoder reuses the 8svx functions because we already have
handling of a single large planar packet for the compressed 8svx codecs.
not to mention race conditions and that its used for stream copy, used to determine IPB type by
applications and other things.
Fixes various frame drop/timestamp issues with frame multithreading.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tls: Use ERR_get_error() in do_tls_poll
indeo3: Fix a fencepost error.
mxfdec: Fix comparison of unsigned expression < 0.
mpegts: set stream id on just created stream, not an unrelated variable
ra288: return error if input buffer is too small
ra288: utilize DSPContext.vector_fmul()
ra288: use memcpy() to copy decoded samples to output
mace: only calculate output buffer size once
Remove redundant filename self-references inside files.
indeo3data: add missing config.h #include for HAVE_BIGENDIAN
x86: drop pointless ARCH_X86 #ifdef from files in x86 subdirectory
avplay: reset rdft when closing stream.
doc/git-howto: expand format-patch and send-email notes.
lavf: expand doxy for some AVFormatContext fields.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The return value ret isn't an error code that can be passed
to ERR_error_string().
This makes the error messages printed actually contain useful
information.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avformat: Avoid a warning about mixed declarations and code
BMV demuxer and decoder
matroskaenc: Make sure the seekhead struct is freed even on seek failure
mpeg12enc: Remove write-only variables.
mpeg12enc: Don't set up run-level info for level 0.
msmpeg4: Don't set up run-level info for level 0.
avformat: Warn about using network functions without calling avformat_network_init
avformat: Revise wording
rdt: Set AVFMT_NOFILE on ff_rdt_demuxer
rdt: Check the return value of avformat_open
rtsp: Discard the dynamic handler, if it has an alloc function which failed
dsputil: use cpuflags in x86 versions of vector_clip_int32()
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The caller expects the seekhead struct to be freed when calling
matroska_write_seekhead. Currently, the structure is leaked if the
seek fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is to make developers aware of the fact that they will
start using the new init function at some point.
Signed-off-by: Martin Storsjö <martin@martin.st>
It might make sense not to make the function completely mandatory
immediately at the next bump, which might be quite soon after
the function was introduced.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes rdt work again, which has been broken since
603b8bc2a1. This commit made
opening a demuxer without a file (or in this case, with a filename
which can't be opened) fail, unless the demuxer actually declared
AVFMT_NOFILE.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (23 commits)
x86inc: use sse versions of common macros instead of sse2 when applicable
doc/APIchanges: add missing dates and hashes
lavf: don't return from void av_update_cur_dts()
Changelog: add more entries.
Changelog: update ffmpeg/avconv incompatibility list.
avconv: remove some redundant temporary variables.
avconv: fix broken indentation
avconv: move copy_initial_nonkeyframes to the options context.
avconv: use file:stream instead of file.stream in log messages.
doc/avconv: elaborate on basic functionality.
doc/avconv: -sample_fmts, not -help sample_fmts prints the sample formats
openssl: Only use CRYPTO_set_id_callback on OpenSSL < 1.0.0
Call avformat_network_init/deinit in the programs
Remove leftover includes of strings.h
avutil: Don't allow using strcasecmp/strncasecmp
Replace all usage of strcasecmp/strncasecmp
avstring: Add locale independent implementations of strcasecmp/strncasecmp
avstring: Add locale independent implementations of toupper/tolower
cosmetics: insert some spaces in explicit enum value assignments
move 8SVX audio codecs to the audio codec list part on the next bump
...
Conflicts:
avprobe.c
doc/APIchanges
ffplay.c
ffserver.c
libavcodec/avcodec.h
libavdevice/bktr.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavformat/matroskaenc.c
libavformat/wtv.c
libavutil/avstring.c
libavutil/avstring.h
libavutil/avutil.h
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Since 1.0.0, this function is deprecated. A new function,
CRYPTO_THREADID_set_callback is available, but if not set at all,
it uses the address of errno as thread id, which should be
sufficient for most systems.
On windows, it never was necessary to use this function even
before 1.0.0, it used the right win32 API function for this
by default.
Signed-off-by: Martin Storsjö <martin@martin.st>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
http: Remove the custom function for disabling chunked posts
rtsp: Disable chunked http post through AVOptions
movdec: Set frame_size for AMR
h264_weight: remove duplication functions.
swscale: align vertical filtersize by 2 on x86.
libavfilter: reindent.
matroskadec: empty blocks are in fact valid.
avfilter: don't abort() on zero-size allocations.
h264: improve calculation of codec delay.
movenc: Set a correct packet size for AMR-NB mode 15, "no data"
avformat: Add functions for doing global network initialization
avformat: Add the https protocol
avformat: Add the tls protocol, using OpenSSL or gnutls
avformat: Initialize gnutls in ff_tls_init()
w32threads: Wrap the mutex functions in inline functions returning int
configure: Allow linking to the gnutls library
avformat: Add ff_tls_init()/deinit() that initialize OpenSSL
configure: Allow linking to openssl
avcodec: Allow locking and unlocking an avformat specific mutex
avformat: Split out functions from network.h to a new file, network.c
Conflicts:
Changelog
configure
doc/APIchanges
libavcodec/internal.h
libavcodec/version.h
libavfilter/formats.c
libavformat/matroskadec.c
libavformat/mov.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Earlier, sc->samples_per_frame was used for setting the frame size,
but all files don't have that set properly. The frame size is a
known constant for these codecs.
If frame_size isn't set, the mov/3gp muxer refuses to mux it.
This fixes stream copy of audio from
https://roundup.libav.org/file1248/Video_With_AMR-NB_Audio.3gp
to another 3gp file (roundup issue 2468).
Signed-off-by: Martin Storsjö <martin@martin.st>
These packets are valid packets, and consist of 1 byte (which
contains the mode bits).
This had been analyzed and reported by Igor Levin, igor d levin comverse com.
Signed-off-by: Martin Storsjö <martin@martin.st>
Note, this protocol doesn't yet check verify the server
certificate against a local database of trusted CA root
certificates.
Signed-off-by: Martin Storsjö <martin@martin.st>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (44 commits)
replacement Indeo 3 decoder
gsm demuxer: do not allocate packet twice.
flvenc: use first packet delay as global delay.
ac3enc: doxygen update.
imc: return error codes instead of 0 for error conditions.
imc: return meaningful error codes instead of -1
imc: do not set channel layout for stereo
imc: validate channel count
imc: check for ff_fft_init() failure
imc: check output buffer size before decoding
imc: use DSPContext.bswap16_buf() to byte-swap packet data
rtsp: add allowed_media_types option
libgsm: add flush function to reset the decoder state when seeking
libgsm: simplify decoding by using a loop
gsm: log error message when packet is too small
libgsmdec: do not needlessly set *data_size to 0
gsmdec: do not needlessly set *data_size to 0
gsmdec: add flush function to reset the decoder state when seeking
libgsmdec: check output buffer size before decoding
gsmdec: log error message when output buffer is too small.
...
Conflicts:
Changelog
ffplay.c
libavcodec/indeo3.c
libavcodec/mjpeg_parser.c
libavcodec/vp3.c
libavformat/cutils.c
libavformat/id3v2.c
libavutil/parseutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adds support for year (TYER) and day/month (TDAT) tags when writing
id3v2 version 3 metadata by splitting the "date" tag. The date tag
should have a format of "YYYY-MM-DD" or "YYYY".
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.
This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
* qatar/master: (51 commits)
cin audio: use sign_extend() instead of casting to int16_t
cin audio: restructure decoding loop to avoid a separate counter variable
cin audio: use local variable for delta value
cin audio: remove unneeded cast from void*
cin audio: validate the channel count
cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
dsicin: fix several audio-related fields in the CIN demuxer
flacdec: use av_get_bytes_per_sample() to get sample size
dca: handle errors from dca_decode_block()
dca: return error if the frame header is invalid
dca: return proper error codes instead of -1
utvideo: handle empty Huffman trees
binkaudio: change short to int16_t
binkaudio: only decode one block at a time.
binkaudio: store interleaved overlap samples in BinkAudioContext.
binkaudio: pre-calculate quantization factors
binkaudio: add some buffer overread checks.
atrac3: support float or int16 output using request_sample_fmt
atrac3: add CODEC_CAP_SUBFRAMES capability
atrac3: return appropriate error codes instead of -1
...
Conflicts:
libavcodec/atrac1.c
libavcodec/dca.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
bits_per_coded_sample should be 8.
block_align is calculated incorrectly, but it is not needed anyway.
packet pts should be calculated in samples.
packet duration can be set.
This reverts commit 5dd514af93.
Silently ignoring errors allows some broken files to simply be played, instead of failing.
(cherry picked from commit 7804b0693375c1a7ba1046f7a3579e9f63c2b15a)
The intended goal (as confirmed with its author) of fixing a crash has been
fixed differently prior to the application of this patch and this patch does
notsucessfully propagate parse errors either.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes false positives of has_codec_delay_been_guessed() for
streams where not every input picture generates an output picture,
such as interlaced H264.
* qatar/master: (53 commits)
probe: Restore identification of files with very large id3 tags and no extension.
probe: Remove id3 tag presence as a criteria to do file extension checking.
mpegts: MP4 SL support
mpegts: MP4 OD support
mpegts: Add support for Sections in PMT
mpegts: Replace the MP4 descriptor parser with a recursive parser.
mpegts: Add support for multiple mp4 descriptors
mpegts: Parse mpeg2 SL descriptors.
isom: Add MPEG4SYSTEMS dummy object type indication.
aacdec: allow output reconfiguration on channel changes
nellymoserenc: take float input samples instead of int16
nellymoserdec: use dsp functions for overlap and windowing
nellymoserdec: do not fail if there is extra data in the packet
nellymoserdec: fail if output buffer is too small
nellymoserdec: remove pointless buffer size check.
lavf: add init_put_byte() to the list of visible symbols.
seek-test: free options dictionary after use
snow: do not draw_edge if emu_edge is set
tools/pktdumper: update to recent avformat api
seek-test: update to recent avformat api
...
Conflicts:
doc/APIchanges
libavcodec/mpegaudiodec.c
libavcodec/nellymoserdec.c
libavcodec/snow.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/avformat.h
libavformat/mpegts.c
libavformat/mxfdec.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prevent error condition in case sample_rate is unset or set to a negative
value. In particular, fix divide-by-zero error occurring in ffmpeg due to
sample_rate set to 0 in output_packet(), in code:
ist->next_pts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
ist->st->codec->sample_rate;
Fix trac ticket #324.
This atom typically is used for a track title.
(cherry picked from commit a356137816b4ea20a892d1fb203b11dbfedbc543)
Reviewed-by: Baptiste Coudurier
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
g722dec: check output buffer size before decoding
g722dec: cosmetics: reindent/linewrap
g722dec: remove the use of lowres for half-rate decoding.
tta: check for extradata allocation failure in tta demuxer
tta: check for allocation failure of decode_buffer
tta: use correct frame_length calculation.
tta: add support for decoding 24-bit sample format
cosmetics: indentation
tta: remove pointless braces
tta: check output buffer size after adjusting frame length for last frame
tta: fix reading of format in TTA header.
tta: remove useless commented-out lines
tta: check remaining bitstream size while reading unary value
lavf: deprecate AVStream.stream_copy
avconc: split choose_codec() to choose_decoder/choose_encoder.
lavf: simplify by using FFMAX/FFMIN.
mpegenc: add preload private option.
cosmetics: simplify latm_decode_init
latm: avoid unnecessary reinit of the aac decoder
aacdec: initialize sbr context only in new channel elements
...
Conflicts:
avconv.c
libavcodec/resample.c
libavcodec/tta.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The situation was not clear when support was added but it is now:
CELT and Opus are really two different codecs.
The current code supports CELT via libcelt, not Opus.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Move id3v2 tag writing to a separate file.
swscale: add missing colons to x86 assembly yuv2planeX.
g722: split decoder and encoder into separate files
cosmetics: remove extra spaces before end-of-statement semi-colons
vorbisdec: check output buffer size before writing output
wavpack: calculate bpp using av_get_bytes_per_sample()
ac3enc: Set max value for mode options correctly
lavc: move get_b_cbp() from h263.h to mpeg4videoenc.c
mpeg12: move closed_gop from MpegEncContext to Mpeg1Context
mpeg12: move full_pel from MpegEncContext to Mpeg1Context
mpeg12: move Mpeg1Context from mpeg12.c to mpeg12.h
mpegvideo: remove some unused variables from MpegEncContext.
Conflicts:
libavcodec/mpeg12.c
libavformat/mp3enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
id3v2: fix doxy comment - 'machine byte order' makes no sense on char arrays
VC1: restore mistakenly removed code
twinvq: check output buffer size before decoding
twinvq: return an error when the packet size is too small
lavf: export some forgotten symbols with non-av prefixes.
swscale: update altivec yuv2planeX asm to new per-plane API.
swscale: make yuv2yuvX_10_sse2/avx 8/9/16-bits aware.
yuv2planeX10 SIMD
swscale: decide whether to use yuv2plane1/X on a per-plane basis.
swscale: reintroduce full precision in 16-bit output.
Split up yuv2yuvX functions
Split out yuv2yuv1 luma and chroma in order to make them generic DSP functions
lavc: replace references to deprecated AVCodecContext.error_recognition to use AVCodecContext.err_recognition
lavc: translate non-flag-based er options into flag-based ef options at codec open
add -err_filter AVOptions to access flag-based error recognition
h264_weight: initialize "height" function argument properly.
presets: spelling error in libvpx 1080p50_60
avplay: fix fullscreen behaviour with SDL 1.2.14 on Mac OS X
Conflicts:
ffplay.c
libavformat/libavformat.v
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
flvdec: Do not call parse_keyframes_index with a NULL stream
libspeexdec: include system headers before local headers
libspeexdec: return meaningful error codes
libspeexdec: cosmetics: reindent
libspeexdec: decode one frame at a time.
swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
Move timefilter code from lavf to lavd.
mov: add support for hdvd and pgapmetadata atoms
mov: rename function _stik, some indentation cosmetics
mov: rename function _int8 to remove ambiguity, some indentation cosmetics
mov: parse the gnre atom
mp3on4: check for allocation failures in decode_init_mp3on4()
mp3on4: create a separate flush function for MP3onMP4.
mp3on4: ensure that the frame channel count does not exceed the codec channel count.
mp3on4: set channel layout
mp3on4: fix the output channel order
mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
mp3on4: copy MPADSPContext from first context to all contexts.
fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
...
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_ps.c
libavcodec/h264dsp_template.c
libavcodec/h264idct_template.c
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavcodec/x86/h264dsp_mmx.c
libavdevice/Makefile
libavdevice/jack_audio.c
libavformat/Makefile
libavformat/flvdec.c
libavformat/flvenc.c
libavutil/pixfmt.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (47 commits)
lavc: hide private symbols.
lavc: deprecate img_get_alpha_info().
lavc: use avpriv_ prefix for ff_toupper4.
lavc: use avpriv_ prefix for ff_copy_bits and align_put_bits.
lavc: use avpriv_ prefix for ff_ac3_parse_header.
lavc: use avpriv_ prefix for ff_frame_rate_tab.
lavc: rename ff_find_start_code to avpriv_mpv_find_start_code
lavc: use avpriv_ prefix for ff_split_xiph_headers.
lavc: use avpriv_ prefix for ff_dirac_parse_sequence_header.
lavc: use avpriv_ prefix for some dv symbols used in lavf.
lavc: use avpriv_ prefix for some flac symbols used in lavf.
lavc: use avpriv_ prefix for some mpeg4audio symbols used in lavf.
lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
lavc: use avpriv_ prefix for ff_aac_parse_header().
lavf: hide private symbols.
lavf: use avpriv_ prefix for some dv functions.
lavf: use avpriv_ prefix for ff_new_chapter().
avcodec: add CODEC_CAP_DELAY note to avcodec_decode_audio3() documentation
avcodec: clarify the CODEC_CAP_DELAY note in avcodec_decode_video2()
avcodec: clarify documentation of CODEC_CAP_DELAY
...
Conflicts:
configure
doc/general.texi
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dv.c
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/libspeexenc.c
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/avidec.c
libavformat/dv.c
libavformat/dv.h
libavformat/flvenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/oggparsespeex.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
presets: rename presets directory
lavc: make avcodec_get_context_defaults3 "officially" public
lavf: replace av_new_stream->avformat_new_stream part II.
lavf,lavd: replace av_new_stream->avformat_new_stream part I.
lavf: add avformat_new_stream as a replacement for av_new_stream.
Use correct scaling table for bwd-pred MVs in second B-field
Ut Video decoder
Makefile: change presets extension to .avpreset
lavfi: add rgbtestsrc source, ported from MPlayer libmpcodecs
lavfi: add testsrc source
AVOptions: add documentation.
presets: update libx264 ffpresets
Conflicts:
Changelog
doc/APIchanges
doc/ffmpeg.texi
ffpresets/libx264-ipod320.ffpreset
ffpresets/libx264-ipod640.ffpreset
ffserver.c
libavcodec/avcodec.h
libavcodec/options.c
libavcodec/version.h
libavdevice/libdc1394.c
libavfilter/avfilter.h
libavfilter/vsrc_testsrc.c
libavformat/flvdec.c
libavformat/riff.c
libavformat/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
