malloc() is allowed to return NULL when zero is the argument. This
causes us to think malloc has failed and return AVERROR(ENOMEM). In
addition OS X malloc() returns an unfreeable non-NULL pointer for size
zero when alignment is greater than 16.
instead of when the 2nd stream has been found.
This isnt ideal as we will likely still like before miss a data stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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.
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.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
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>
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>
* 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.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
* qatar/master:
adtsenc: Check frame size.
txd: Fix order of operations.
APIchanges: fill in some blanks
timer: fix misspelling of "decicycles"
Eliminate pointless 0/NULL initializers in AVCodec and similar declarations.
indeo3: cosmetics
md5proto: Fix order of operations.
dca: Replace oversized unused get_bits() with skip_bits_long().
Conflicts:
doc/APIchanges
libavformat/mmsh.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vc1: use an enum for Frame Coding Mode
doc: cleanup filter section
indeo3: error out if no motion vector is set.
x86inc: Flag shufps as an floating-point instruction for the AVX emulation code.
mpegaudio: do not use init_static_data() for initializing tables.
musepack: fix signed shift overflow in mpc_read_packet()
mov: Make format string match variable type.
wmavoice: Make format string match variable type.
vc1: select interlaced scan table by FCM element
Generalize RIFF INFO tag support; support reading INFO tag in wav
pthread: track thread existence in a separate variable.
Conflicts:
doc/filters.texi
libavcodec/pthread.c
libavformat/avi.c
libavformat/riff.c
libavformat/riff.h
libavformat/wav.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using an unsigned variable avoids problems with overflows.
There is further no need for a 64-bit intermediate here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (42 commits)
swscale: fix signed overflow in yuv2mono_X_c_template
snow: fix integer overflows
svq1enc: remove stale altivec-related hack
snow: fix signed overflow in byte to 32-bit replication
adx: rename ff_adx_decode_header() to avpriv_adx_decode_header()
avformat: add CRI ADX format demuxer
adx: add an ADX parser.
adx: move header decoding to ADX common code
adx: calculate the number of blocks in a packet
adx: define and use 2 new macro constants BLOCK_SIZE and BLOCK_SAMPLES
adx: check for unsupported ADX formats
adx: simplify encoding by using put_sbits()
adx: calculate correct LPC coeffs
adx: use 12-bit coefficients instead of 14-bit to avoid integer overflow
adx: simplify adx_decode() by using get_sbits() to read residual samples
adx: fix the data offset parsing in adx_decode_header()
adx: remove unneeded post-decode channel interleaving
adx: validate header values
adx: cosmetics: general pretty-printing and comment clean-up
adx: remove useless comments
...
Conflicts:
Changelog
libavcodec/cook.c
libavcodec/fraps.c
libavcodec/nuv.c
libavcodec/pthread.c
libavcodec/version.h
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies the decoder so it doesn't have to process an in-packet header
or handle arbitrary-sized packets. It also fixes decoding of files with large
headers.
Also reduce verbosity for the unsupported stream message, use
an AVFormatContext for av_log and and print the tag of the
unknown stream.
Improves ticket #672.
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framemd5, as it prints dts.
-vsync 0 for the vp8 test is needed because with vsync 2 the timestamp
guessing code gets confused by an altref frame that is never displayed
and drops a frame later.
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Adding the thread count in frame level multithreading to has_b_frames
as an additional delay causes more problems than it solves.
For example inconsistent behaviour during timestamp calculation in
libavformat.
Thread count and frame level multithreading are both set by the user.
If the additional delay caused by frame level multithreading needs
to be considered in the calling code it has all information to take
it into account.
Should it become necessary to calculate a maximum delay inside
libavcodec it should be exported as its own field and not reusing
an existing field.
Based on a patch by Michael Niedermayer.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Some sample IFF ACBM files can be found here:
http://aminet.net/package/dev/basic/ABdemos
Thanks to Peter Ross for his help with this patch.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall: (24 commits)
Clean-up
dump_int_buffer() to dump samples from a buffer
Implement revert_cdlms()
Doxy for reset_codec()
Store transient state and position of transient area
Implement use_high_update_speed() and use_normal_update_speed()
Initialize num_logged_tiles and remove unnecessary codes
Log index for each line of output
Log tile size
Output decoded residues
Replace placeholders with actual calls to clear_codec_buffers() and reset_codec()
Implement lms_update()
Implement lms_predict()
Implement reset_codec()
Add missing syntax elements to WmallDecodeCtx
Add .recent syntax element to cdlms struct
Implement clear_codec_buffers()
Add buffers to context necessary for reverting cdmls and mclms filter
Use avpriv_copy_bits() instead of ff_copy_bits()
Cosmetics
...
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Note: FCPublish/FCUnpublish are adobe server specific and not described
in the rtmp specification. Some servers might not cope with them at
all.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tls protocol handles connections via proxies internally.
With TLS/SSL, the peer verification requires that the client
speaks directly with the server, since the proxy doesn't have
the remote server's private key.
Signed-off-by: Martin Storsjö <martin@martin.st>
This opens a plain TCP connection through the proxy via the
CONNECT HTTP method. Normally, this is allowed for connections
on port 443, but can in general be used to allow connections
to any port (depending on proxy configuration), and could thus
be used to tunnel any TCP connection via a HTTP proxy.
Signed-off-by: Martin Storsjö <martin@martin.st>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
Our ac3 code chain can handle it fine.
More ideal would be to write a demuxer that actually extracts what can be from the additional
headers and uses it for whatever it can be used for.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
TLSv1 is compatible with SSLv3, so this doesn't change much
in terms of compatibility. By explicitly using TLSv1, OpenSSL
sends the server name indication (SNI) header, which we
already set using SSL_set_tlsext_host_name (earlier, this
didn't have any effect).
SNI allows servers to serve SSL content for different host
names with separate certificates on one single port (vhosts).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpegaudiodec: Don't use a nonexistent log context for av_dlog
avformat: Accept the ISO8601 separate format as input, too
avformat: Interpret times in ff_iso8601_to_unix_time as UTC
avutil: Add av_timegm as a public function
cinepak: Add another special case so that it can handle the following file:
lagarith: add some RGBA decoding support
lagarith: Add correct line prediction for RGB
Conflicts:
doc/APIchanges
libavcodec/cinepak.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
Seems to fix trac issue #569.
Sample is unfortunately not available, but it might be caused by
an index existing for non-existing audio stream (?).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
* qatar/master:
avconv: add presets
rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
rtsp: Make the rtsp flags avoptions set via a define
rtpenc: Set a default video codec
avoptions: Fix av_opt_flag_is_set
rtp: Fix ff_rtp_get_payload_type
doc: Update the documentation on setting options for RTSP
rtsp: Remove the separate filter_source variable
rtsp: Accept options via private avoptions instead of URL options
rtsp: Simplify AVOption definitions
rtsp: Merge the AVOption lists
lavfi: port libmpcodecs delogo filter
lavfi: port boxblur filter from libmpcodecs
lavfi: add negate filter
lavfi: add LUT (LookUp Table) generic filters
AVOptions: don't segfault on NULL parameter in av_set_options_string()
avio: Check for invalid buffer length.
mpegenc/mpegtsenc: add muxrate private options.
lavf: deprecate AVFormatContext.file_size
mov: add support for TV metadata atoms tves, tvsn and stik
Conflicts:
Changelog
doc/filters.texi
doc/protocols.texi
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/internal.h
libavfilter/vf_boxblur.c
libavfilter/vf_delogo.c
libavfilter/vf_lut.c
libavformat/mpegtsenc.c
libavformat/utils.c
libavformat/version.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).
Signed-off-by: Martin Storsjö <martin@martin.st>
avconv doesn't map video streams to a muxer without specifying a
manual stream mapping if the default video codec is CODEC_ID_NONE.
Signed-off-by: Martin Storsjö <martin@martin.st>
It was broken in 3b3ea34655
"Remove all uses of deprecated AVOptions API", where any
presence of a payload_type AVOption caused its value to
be returned, even if it wasn't set (and thus had the default
-1 value).
This caused the RTP muxer to be broken.
Signed-off-by: Martin Storsjö <martin@martin.st>
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.
This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use defines for shortening common parts, omit the .dbl named
initializer (since it's the first element in the union).
Signed-off-by: Martin Storsjö <martin@martin.st>
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
prores: get correct size for coded V plane if alpha is present
prores: do not set pixel format on codec init
pthread: prevent updating AVCodecContext from itself in frame_thread_free
pthread: copy coded frame dimensions in update_context_from_thread
vp8: prevent read from uninitialized memory in decode_mvs
vp8: force reallocation in update_thread_context after frame size change
vp8: fix return value if update_dimensions fails
matroskadec: fix out of bounds write
adpcmdec: calculate actual number of output samples for each decoder.
adpcmdec: check remaining buffer size before decoding next block in the ADPCM IMA WAV decoder.
adpcmdec: do not terminate early in ADPCM IMA Duck DK3 decoder.
adpcmdec: remove unneeded buf_size==0 check.
adpcmdec: remove unneeded zeroing of *data_size
dnxhdenc: fixed signed multiplication overflow
Conflicts:
tests/ref/fate/prores-alpha
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master: (22 commits)
prores: add FATE tests
id3v2: reduce the scope of some non-globally-used symbols/structures
id3v2: cosmetics: move some declarations before the places they are used
shorten: remove the flush function.
shn: do not allow seeking in the raw shn demuxer.
avformat: add AVInputFormat flag AVFMT_NO_BYTE_SEEK.
avformat: update AVInputFormat allowed flags
avformat: don't unconditionally call ff_read_frame_flush() when trying to seek.
truespeech: use sizeof() instead of hardcoded sizes
truespeech: remove unneeded variable, 'consumed'
truespeech: simplify truespeech_read_frame() by using get_bits()
truespeech: decode directly to output buffer instead of a temp buffer
truespeech: check to make sure channels == 1
truespeech: check for large enough output buffer rather than truncating output
truespeech: remove unneeded zero-size packet check.
mlpdec: return meaningful error codes instead of -1
mlpdec: remove unnecessary wrapper function
mlpdec: only calculate output size once
mlpdec: validate that the reported channel count matches the actual output channel count
pcm: reduce pointer type casting
...
Conflicts:
libavformat/avformat.h
libavformat/id3v2.c
libavformat/id3v2.h
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The demuxer does not read the seektable, a parser is not possible without a
full decode, and no shorten decoder can handle random seeking because it needs
side info from the seektable.
Previous code could read 4 bytes past the end of the buffer on a RTCP_SR
packet or offset a pointer by an unchecked external value (payload_len),
though neither will reliably cause a crash or other misbehavior beyond
garbage timestamps.
Additionally, unknown RTCP packet types, even in compounded packets, are
now ignored as per RFC 3550 section 6.1, page 22, though currently this
only has any practical effect if a sender puts an unrecognized type
before RTCP_BYE in a compounded packet, or (incorrectly) does not put
RTCP_SR first.
Signed-off-by: John Brooks <john.brooks@bluecherry.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This will allow the caller to enumerate child contexts in a generic way
and since the API is recursive, it also allows for deeper nesting (e.g.
AVFormatContext->AVIOContext->URLContext)
This will also allow the new setting/reading API to transparently apply
to children contexts.
No application rely on this count being correct as far as
I know, but if we write a nonzero count value, it might just
as well be the right one.
Signed-off-by: Martin Storsjö <martin@martin.st>
DSS enables this automatically if streaming VOD over TCP. If
enabled, the server feeds packets faster than realtime, screwing
up RTCP NTP based timestamps.
Also, DSS doesn't indicate that this was indicated, if it was
enabled automatically (although if it was requested to be enabled,
a header saying that it was enabled is added, but this isn't
added if it is enabled automatically), making it even harder
to detect and work around properly without explicitly asking
for it to be disabled(/enabled, if we were able to support it).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.
Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids writing these entries doubly if transcoding from
flv to flv, since the muxer blindly writes any and all metadata
keys set, in addition to the fixed fields that the muxer
always writes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (23 commits)
fix AC3ENC_OPT_MODE_ON/OFF
h264: fix HRD parameters parsing
prores: implement multithreading.
prores: idct sse2/sse4 optimizations.
swscale: use aligned move for storage into temporary buffer.
prores: extract idct into its own dspcontext and merge with put_pixels.
h264: fix invalid shifts in init_cavlc_level_tab()
intfloat_readwrite: fix signed addition overflows
mov: do not misreport empty stts
mov: cosmetics, fix for and if spacing
id3v2: fix NULL pointer dereference
mov: read album_artist atom
mov: fix disc/track numbers and totals
doc: fix references to obsolete presets directories for avconv/ffmpeg
flashsv: return more meaningful error value
flashsv: fix typo in av_log() message
smacker: validate channels and sample format.
smacker: check buffer size before reading output size
smacker: validate number of channels
smacker: Separate audio flags from sample rates in smacker demuxer.
...
Conflicts:
cmdutils.h
doc/ffmpeg.texi
libavcodec/Makefile
libavcodec/motion_est_template.c
libavformat/id3v2.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (34 commits)
h264: reset h->ref_count in case of errors in ff_h264_decode_ref_pic_list_reordering()
error_resilience: fix the check for missing references in ff_er_frame_end() for H264
4xm: prevent NULL dereference with invalid huffman table
4xmdemux: prevent use of uninitialized memory
4xm: clear FF_INPUT_BUFFER_PADDING_SIZE bytes in temporary buffers
ptx: check for out of bound reads
tiffdec: fix out of bound reads/writes
eacmv: check for out of bound reads
eacmv: fix potential pointer arithmetic overflows
adpcm: fix out of bound reads due to integer overflow
anm: prevent infinite loop
avsdemux: check for out of bound writes
avs: check for out of bound reads
avsdemux: check for corrupted data
AVOptions: refactor set_number/write_number
AVOptions: cosmetics, rename static av_set_number2() to write_number().
AVOptions: cosmetics, move and rename static av_set_number().
AVOptions: split av_set_string3 into opt type-specific functions
avidec: fix signed overflow in avi_sync()
mxfdec: Fix some buffer overreads caused by the misuse of AVPacket related functions.
...
Conflicts:
Changelog
configure
libavcodec/ptx.c
libavcodec/ra144.c
libavcodec/vaapi_vc1.c
libavcodec/vc1.c
libavcodec/version.h
libavformat/4xm.c
libavformat/avidec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Keeping byte values read from the file as unsigned is consistent
with how they are subsequently used and avoids an undefined left
shift by 24 when bit 7 is set.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Audio and video are interleaved via stream cur_dts - no idea how
reliable this is.
It also cannot display the video properly - it is stored with
about 15 in a single JPEG frame, I cannot think of a reasonable
way to implement this.
Samples: http://samples.mplayerhq.hu/smv/
Format description: http://wiki.multimedia.cx/index.php?title=SMV
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
lavf: fix signed overflow in avformat_find_stream_info()
vp8: fix signed overflows
motion_est: fix some signed overflows
dca: fix signed overflow in shift
aacdec: fix undefined shifts
bink: Check for various out of bound writes
bink: Check for out of bound writes when building tree
put_bits: fix invalid shift by 32 in flush_put_bits()
Conflicts:
libavcodec/bink.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
On the first iteration through this code, last_dts is always
INT64_MIN (AV_NOPTS_VALUE) and the subtraction overflows in
an invalid manner. Although the result is only used if the
input values are valid, performing the subtraction is still
not allowed in a strict environment.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (24 commits)
mpegps: Use av_get_packet() instead of poorly emulating it.
motionpixels: decode only the 111 complete frames for fate
mpc8: Check out of bound bands limit
xan: Prevent NULL dereference with missing palette
xan: Check for out of bound reads in xan_huffman_decode()
xan: Fixed out of bound accesses in xan_unpack()
motionpixels: Prevent calling init_vlc() with invalid parameters
shorten: Fix out of bound writes in fix_bitshift()
dsicinav: Check for out of bounds writes
tiertexseqv: Check for out of bound reads
quickdraw: Check for out of bound reads
dsicinav: Check for out of bounds reads
motionpixels: Fix the size of workspace buffers
motionpixels: Clear FF_INPUT_BUFFER_PADDING_SIZE bytes at the end of the temporary buffer
wmavoice: Check for corrupted extra data
wmavoice: Check for out of bound writes
xan: Prevent NULL dereferences with missing reference frame
bink: Prevent NULL dereferences with missing reference frame
wavpack: Reset internal state on corrupted blocks
wmapro: Validate the number of audio channels before using it
...
Conflicts:
libavcodec/h264.c
libavcodec/xan.c
tests/ref/fate/motionpixels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix 'heigth' vs. 'height' typos.
lavc/lavf: use unique private classes.
lavc: use designated initializers for av_codec_context_class
Conflicts:
libavdevice/fbdev.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In case of av_new_packet() error, a correct return error code is raised,
the data memcpy is avoided, and pkt dts/pts are not assigned anymore
(since the defaults are good).
* qatar/master:
qcelpdec: cosmetics: do not add line break before opening bracket in 'for', 'while', 'if/else', and 'switch' statements.
qcelp: check output buffer size before decoding
qcelpdec: fix the return value of qcelp_decode_frame().
sipr: fix the output data size check and only calculate it once.
Synchronize various 4CCs and codec tags from FFmpeg.
qdm2: check output buffer size before decoding
Fix out of bound reads in the QDM2 decoder.
Check for out of bound writes in the QDM2 decoder.
ogg/celt: do not set sample_fmt in the demuxer
Conflicts:
libavcodec/avcodec.h
libavcodec/qdm2.c
libavformat/oggparsecelt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
id3v2: remove pointless casts
id3v2: read TXXX frames with two calls to decode_str() instead of one.
id3v2: don't discard the whole tag when encountering empty frames.
libvpx: fix build with older libvpx versions.
ARM: check for inline asm 'y' operand modifier support
Conflicts:
libavcodec/libvpxenc.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (34 commits)
dpcm: return error if packet is too small
dpcm: use smaller data types for static tables
dpcm: use sol_table_16 directly instead of through the DPCMContext.
dpcm: replace short with int16_t
dpcm: check to make sure channels is 1 or 2.
dpcm: misc pretty-printing
dpcm: remove unnecessary variable by using bytestream functions.
dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
dpcm: consistently use the variable name 'n' for the next input byte.
dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
dpcm: calculate and check actual output data size prior to decoding.
dpcm: factor out the stereo flag calculation
dpcm: cosmetics: rename channel_number to ch
avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
lavf: Avoid using av_malloc(0) in av_dump_format
dxva2_h264: pass the correct 8x8 scaling lists
dca: NEON optimised high freq VQ decoding
avcodec: reject audio packets with NULL data and non-zero size
dxva: Add ability to enable workaround for older ATI cards
latmenc: Set latmBufferFullness to largest value to indicate it is not used
...
Conflicts:
libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Extradata should only be parsed from the avss, fiel, jp2h and alac atoms for
AVS, MJPEG, Motion JPEG 2000 and ALAC respectively.
This also fixes the mov demuxer coming up with bogus extradata for some
AVC-Intra samples due to the presence of fiel atoms.
* qatar/master: (23 commits)
avconv: Reformat s16 volume adjustment.
ARM: NEON optimised vector_fmac_scalar()
dca: use vector_fmac_scalar from dsputil
dsputil: add vector_fmac_scalar()
latmenc: Fix private options
vf_unsharp: store hsub/vsub in the filter context
vf_unsharp: adopt a more natural order of params in apply_unsharp()
vf_unsharp: rename method "unsharpen" to "apply_unsharp"
vf_scale: apply the same transform to the aspect during init that is applied per frame
vf_pad: fix "vsub" variable value computation
vf_scale: add a "sar" variable
lavfi: fix realloc size computation in avfilter_add_format()
vsrc_color: use internal timebase
lavfi: fix signature for avfilter_graph_parse() and avfilter_graph_config()
graphparser: prefer void * over AVClass * for log contexts
avfiltergraph: use meaningful error codes
avconv: Initialize return value for codec copy path.
fate: use 'run' helper for seek-test
fate: remove seek-mpeg2reuse test
Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
...
Conflicts:
doc/filters.texi
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/vf_scale.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avconv: use different variables for decoded and filtered frame.
avconv: add support for copying attachments.
matroskaenc: write attachments.
matroskadec: export mimetype of attachments as metadata.
avconv: factorize common code from new_*_stream()
doc/avconv: expand documentation for some options.
doc/avconv: document -timelimit.
Conflicts:
avconv.c
cmdutils.c
tests/codec-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Whitespace of the patch cleaned up by Aurel
Some of the issues have been reported by Steve Manzuik / Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 956c901c68)
Further suggestions from Kostya <kostya.shishkov@gmail.com> have been
implemented by Reinhard Tartler <siretart@tauware.de>
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master: (21 commits)
fate: allow testing with libavfilter disabled
x86: XOP/FMA4 CPU detection support
ws_snd: misc cosmetic clean-ups
ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
ws_snd: use memcpy() and memset() instead of loops
ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
ws_snd: make sure number of channels is 1
ws_snd: add some checks to prevent buffer overread or overwrite.
ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
flacdec: fix buffer size checking in get_metadata_size()
rtp: Simplify ff_rtp_get_payload_type
rtpenc: Add a payload type private option
rtp: Correct ff_rtp_get_payload_type documentation
avconv: replace all fprintf() by av_log().
avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
cmdutils: replace fprintf() by av_log()
avtools: parse loglevel before all the other options.
oggdec: add support for Xiph's CELT codec
sol: return error if av_get_packet() fails.
cosmetics: reindent and pretty-print
...
Conflicts:
avconv.c
cmdutils.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/oggparsecelt.c
libavformat/utils.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifying the payload type is useful when the type number has
already been negotiated before creating the stream, for example
in SIP protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
This patch also introduces CODEC_ID_CELT.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
If the demuxer did not set a codec_tag, there is none and
inventing one makes no sense. This change stops the rawvideo
"decoder" over-writing user-supplied pixfmt with one derived
from the codec_tag. The pixfmt-codec_tag-pixfmt round-trip
is lossy since several pixfmts map to the same codec_tag.
This fixes fate-lavf-pixfmt with avfilter disabled.
Signed-off-by: Mans Rullgard <mans@mansr.com>
On allocation, the array length is multiplied by sizeof(int64_t),
this prevents the multiplication from overflowing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Authors are Vladimir Voroshilov and Dobrica Pavlinušić based on svn blame/log
For full details of authorship see http://code.google.com/p/amv-codec-tools/
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
flvdec: Fix invalid pointer deferences when parsing index
configure: disable hardware capabilities ELF section with suncc on Solaris x86
Use explicit struct initializers for AVCodec declarations.
Use explicit struct initializers for AVOutputFormat/AVInputFormat declarations.
adpcmenc: Set bits_per_coded_sample
adpcmenc: fix QT IMA ADPCM encoder
adpcmdec: Fix QT IMA ADPCM decoder
permit decoding of multichannel ADPCM_EA_XAS
Fix input buffer size check in adpcm_ea decoder.
fft: avoid a signed overflow
mpegps: Handle buffer exhaustion when reading packets.
Conflicts:
libavcodec/adpcm.c
libavcodec/adpcmenc.c
libavdevice/alsa-audio-enc.c
libavformat/flvdec.c
libavformat/mpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtp: factorize dynamic payload type fallback
flvdec: Ignore the index if it's from a creator known to be different
cmdutils: move grow_array out of #if CONFIG_AVFILTER
avconv: actually set InputFile.rate_emu
ratecontrol: update last_qscale_for sooner
Fix unnecessary shift with 9/10bit vertical scaling
prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
prores: return more meaningful error values
prores: improve error message wording
prores: cosmetics: prettyprinting, drop useless parentheses
prores: lowercase AVCodec name entry
Conflicts:
cmdutils.c
libavcodec/proresdec_lgpl.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move the identical code in rtp_write_header() and
ff_sdp_write_media() inside ff_rtp_get_payload_type()
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
Add LATM demuxer
avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
8svx/iff: fix decoding of compressed stereo 8svx files.
8svx: log an error message if output buffer is too small
8svx: check packet size before reading the initial sample value.
8svx: output 8-bit samples instead of 16-bit.
8svx: split delta decoding into a separate function.
mp4: Don't read an empty Decoder Config Descriptor
fate.sh: Ignore errors from rm command during cleanup.
fate.sh: Run git-pull in quiet mode to avoid console spam.
Apple ProRes decoder
rtmp: Make the input FLV parser handle data cut at any point
rv34: Check for invalid slices offsets
eval: test isnan(sqrt(-1)) instead of just sqrt(-1)
Conflicts:
Changelog
libavcodec/8svx.c
libavcodec/proresdec.c
libavcodec/version.h
libavformat/iff.c
libavformat/version.h
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This change fixes a bug where seeking doesn't work properly for
matroska files that have the CUES element before the first cluster.
This bug was accidentally introduced a few months ago by my deferred CUES
loading patch<http://git.videolan.org/?p=ffmpeg.git;a=commit;h=31ad14c21e0735387ba8082c6e3436241f7ccfc8>
.
When the CUES element appears before the first cluster in the file, the data
is parsed and placed in matroska->index but that data is never added to the
seek index. Currently the transfer from matroska->index to the seek index
only happens when matroska_parse_cues() is called.
Matroska_parse_cues() only gets called on a seek if cues_parsing_deferred is
set. Cues_parsing_deferred only gets set if parsing the CUES requires
seeking past the first cluster. There is no code to handle the case where
CUES is before the first cluster.
This fix essentially restores the matroska->index processing that was
happening at the end of matroska_read_header() before I made my CUES
deferral change. In the case where CUES is before the first
cluster, matroska->index will have data and the seek index will be updated.
In the case where CUES is later in the file, matroska->index will be empty
and cues_parsing_deferred will be set so loading will happen later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.
Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da2a
This makes the RTMP writing code able to handle FLV data
fed in arbitrarily small or large chunks, with multiple
consecutive packets in one write call, or having the FLV
packet header split over numerous write calls.
When used in conjunction with the flv muxer, the AVIO buffer
size still needs to be large enough to fit the initial metadata
packet though, since the size of that packet is written with a
seekback.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
swfdec: Add support for sample_rate_code 0 (5512 Hz)
dct-test: factor out some common code and do whas was likely intended
doc: library versions need to be bumped in version.h
Revert "ffmpeg: get rid of useless AVInputStream.nb_streams."
Remove some forgotten AVCodecContext.palctrl usage.
lavc/utils: move avcodec_init() higher in the file.
lavc: replace some deprecated FF_*_TYPE with AV_PICTURE_TYPE_*
ac3dec: actually use drc_scale private option
lavc: undeprecate AVPALETTE_SIZE and AVPALETTE_COUNT macros
alsa: add missing header
msmpeg4: remove leftover unused debug variable declaration
Fix assert() calls that need updates after FF_COMMON_FRAME macro elimination.
Fix av_dlog invocations with wrong or missing logging context.
vf_yadif: add support to yuva420p
vf_yadif: correct documentation on the parity parameter
vf_yadif: copy buffer properties like aspect for second frame as well
oma: support for encrypted files
id3v2: add support for non-text and GEOB type tag frames
des: add possibility to calculate DES-CBC-MAC with small buffer
Conflicts:
ffmpeg.c
libavcodec/dct-test.c
libavformat/mpegts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This extends the ID3v2 parser to allow for reading of non-text (i.e.
other than T***) meta tag frames providing a ff_id3v2_read_all()
function. An additional data structure 'ID3v2ExtraMeta' is introduced
for these tags since AVDictionary is string oriented and unsuitable
for binary data.
A parser for tag frames of type GEOB is implemented, which is needed
to extract keyring information from encrypted OMA files. GEOB data
is parsed into 'ID3v2ExtraMetaGEOB' data structures.
The routine to decode characters from different encodings to UTF-8,
formerly part of the read_ttag() function, is moved to its own
function. Because some tag frames contain subparts of unknown length,
the function is now also able to read until a null character is found.
In addition, the function now takes care of allocating a buffer long
enough to hold the decoded characters.
Signed-off-by: David Goldwich <david.goldwich@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
rtmp: Don't blindly skip the 4 trailer bytes from the FLV packets
rtmp: Handle FLV packets written in more than one write call
rv34: Check for invalid slice offsets
Conflicts:
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
(A minimal RTP packet is 12 bytes, but a minimal RTCP packet can be
much smaller, at least as small as 8 bytes.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If not enough bytes are available, keep track of them and skip
them on next call.
In practice, if these trailer bytes are written in a separate
call, there is no other data written in this call, making it
fall into the "FLV packet too small" case currently - working,
but not as intended.
This patch makes the code more robust, handling all cases
except for having the FLV packet header split over multiple
write calls.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the FLV packet is larger than the AVIO buffer, a partial
FLV packet will be flushed to the RTMP protocol.
This commit handles the most common cases of FLV packets
being written in more than one call.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
movenc: fix NULL reference in mov_write_tkhd_tag
rmdec: Reject invalid deinterleaving parameters
rv34: Fix potential overreads
rv34: Fix buffer size used for MC of B frames after a resolution change
rv34: Avoid NULL dereference on corrupted bitstream
rv10: Reject slices that does not have the same type as the first one
vf_yadif: add an option to enable/disable deinterlacing based on src frame "interlaced" flag
vsrc_color: set output pos values to -1
vsrc_color: add @file doxy
vsrc_buffer: remove duplicated file description
eval: implement not() expression
eval: add sqrt function for computing the square root
rmdec: use the deinterleaving mode and not the codec when creating audio packets.
lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
Conflicts:
doc/eval.texi
doc/filters.texi
libavcodec/rv10.c
libavfilter/vsrc_color.c
libavformat/rmdec.c
libavutil/avutil.h
libavutil/eval.c
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Original code had the assumption of only one FLV packet per RTMP packet. But that assumption is incorrect for higher bit rates. Made changes to the code to accommodate more than one FLV packet per RTMP
+packet.
* qatar/master:
lavfi: add select filter
oggdec: fix out of bound write in the ogg demuxer
movenc: create an alternate group for each media type
lavd: add libcdio-paranoia input device for audio CD grabbing
rawdec: refactor private option for raw video demuxers
pcmdec: use unique classes for all pcm demuxers.
rawdec: g722 is always 1 channel/16kHz
Conflicts:
Changelog
configure
doc/filters.texi
libavdevice/avdevice.h
libavfilter/avfilter.h
libavfilter/vf_select.c
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
pixel_format/video_size only apply to 'rawvideo' (==uncompressed) demuxer
and make no sense for the other raw (== containerless) demuxers. Keep
only the framerate option for those.
Also use unique classes for all raw video demuxers
* qatar/master:
swscale: fix byte overreads in SSE-optimized hscale().
matroskadec: fix typo.
matroskadec: bail on parsing of incorrect seek index segments
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
* qatar/master:
lavf utils: Remove write-only variable
lavf utils: Rename shadowing variable
smacker: fix a few off by 1 errors
Check for invalid VLC value in smacker decoder.
Check and propagate errors when VLC trees cannot be built in smacker decoder.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoid retrying to read ASF index in files for every
attempt to seek. This makes a big difference to protocols
with slow seeking (for example http)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
adpcm: split ADPCM encoders and decoders into separate files.
doc/avconv: fix typo.
rv34: check that subsequent slices have the same type as first one.
smacker demuxer: handle possible av_realloc() failure.
lavfi: add split filter from soc.
lavfi: add showinfo filter
libxavs: add private options corresponding to deprecated global options
Conflicts:
Changelog
libavcodec/adpcm.c
libavfilter/avfilter.h
libavfilter/vf_showinfo.c
libavfilter/vf_split.c
libavformat/smacker.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reduces problems when underlying protocol is not
seekable even if marked as such or if the file has been
cut short.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The code path using for mpegts over rtp doesn't open the demuxer using
mpegts_read_header,
so it never starts listening for PAT/SDT, only uses auto_guess
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This changes so we assume EOF when we can't find the next
streams index entry for non interleaved file.
http://trac.xbmc.org/ticket/5585
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This uses the RIFF header stored size to figure out the expected AVI file size, instead
of the actual file. To work fully it requires handling failed avio_seek() instead
of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which previously
was not output during demuxing. This frame is now output to encoder, thus fate
diff update.
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Employ FF_ARRAY_ELEMS instead of manually calculating array length.
Fixed invalid access in wavpack decoder on corrupted bitstream.
Fixed invalid writes in wavpack decoder on corrupted bitstreams.
Fixed invalid access in wavpack decoder on corrupted extra bits sub-blocks.
rtpdec_asf: Fix integer underflow that could allow remote code execution
Conflicts:
libavformat/rtpdec_asf.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If there is only 1 stream in an flv avformat_find_stream_info will continually
read until probesize is reached. This should stop it reading if the metadata
also claims there to be 1 stream.
Fixes MSVR-11-0088.
Credit: Jeong Wook Oh of Microsoft and Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes MSVR-11-0088
Credit: Jeong Wook Oh of Microsoft and Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: fix type for thread_type option
avconv: move format to options context
avconv: move limit_filesize to options context
avconv: move start_time, recording_time and input_ts_offset to options context
avconv: add a context for options.
cmdutils: allow storing per-stream/chapter/.... options in a generic way
cmdutils: split per-option code out of parse_options().
cmdutils: add support for caller-provided option context.
cmdutils: declare only one pointer type in OptionDef
cmdutils: move grow_array() from avconv to cmdutils.
cmdutils: move exit_program() declaration to cmdutils from avconv
http: Consider the stream as seekable if the reply contains Accept-Ranges: bytes
nutenc: add namespace to the api facing functions
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
ffserver.c
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The initial request contains "Range: 0-", which servers normally
have responded with "HTTP/1.1 206 Partial Content" reply with
a Content-Range header, which was used as indicator for seekability.
Apache, since 2.2.20, responds with "HTTP/1.1 200 OK" for these
requests, which is more friendly to caches and proxies, but the
seekability still is indicated via the Accept-Ranges: bytes header.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
AVOptions: fix av_set_string3() doxy to match reality.
cmdutils: get rid of dummy contexts for examining AVOptions.
lavf,lavc,sws: add {avcodec,avformat,sws}_get_class() functions.
AVOptions: add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find().
cpu detection: avoid a signed overflow
Conflicts:
avconv.c
cmdutils.c
doc/APIchanges
ffmpeg.c
libavcodec/options.c
libavcodec/version.h
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
currently libavformat only allows seeking if a request with "Range:
0-" results in a 206 reply from the HTTP server which includes a
Content-Range header. But according to RFC 2616, the server may also
reply with a normal 200 reply (which is more efficient for a request
for the whole file). In fact Apache HTTPD 2.2.20 has changed the
behaviour in this way and it looks like this change will be kept in
future versions. The fix for libavformat is easy: Also look at the
Accept-Ranges header.
If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.
The Makito encoder sets stream type 0x11 for AAC.
This patch should be reverted if it breaks decoding valid streams (and
the problem can't be fixed in the probe function).
* qatar/master:
Revert "h264: Properly set coded_{width, height} when parsing H.264."
isom: add missing AVC-Intra tags, rearrange list and update comments
avconv: remove stubs of crop* and pad* options
avconv: re-add nb_streams to InputFile.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Use deinterleavers for demangling audio packets in RealMedia.
vf_scale: don't leak SWS context.
doxygen: drop another pointless star from pointer variable name
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes trac issue #343.
Carl Eugen Hoyos actually made a patch first, but I missed it because
trac does not send notification emails when an attachment is added.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also extend the functionality to use the last found program to start the search
after that program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Unlike other containers RealMedia stores its audio packets in scrambled form,
with interleaver ID preceeding audio codec ID. Currently deinterleaving
decision is tied to the codec while it's possible to have non-default
deinterleaver with audio codec (like Int0 deinterleaver instead of specific
one for Sipro).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
fifo: add FIFO API test program, and fate test
fifo: add av_fifo_peek2(), and deprecate av_fifo_peek()
postprocess.c: filter name needs to be double 0 terminated
doxygen: fix wrong comment syntax, //< vs. ///<
doxygen: drop pointless star from pointer variable names
Replace deprecated av_find_stream_info() by avformat_find_stream_info().
xmv: eliminate superfluous zeroing of zero data
configure: fix typo in avconv dependency list
Conflicts:
configure
doc/APIchanges
libavutil/Makefile
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The new function provides a more generic interface than av_fifo_peek()
for peeking at a FIFO buffer data.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
