* qatar/master:
lavf: don't guess r_frame_rate from either stream or codec timebase.
avconv: set discard on input streams automatically.
Fix parser not to clobber has_b_frames when extradata is set.
lavf: don't set codec timebase in avformat_find_stream_info().
avconv: saner output video timebase.
rawdec: set timebase to 1/fps.
avconv: refactor vsync code.
FATE: remove a bunch of useless -vsync 0
cdxl: bit line plane arrangement support
cdxl: remove early check for bpp
cdxl: set pix_fmt PAL8 only if palette is available
Reinhard Tartler [Sun, 26 Feb 2012 09:50:45 +0000 (10:50 +0100)]
Fix parser not to clobber has_b_frames when extradata is set.
Because in contrast to the decoder, the parser does not setup low_delay.
The code in parse_nal_units would always end up setting has_b_frames
to "1", except when stream is explicitly marked as low delay.
Since the parser itself would create 'extradata', simply reopening
the parser would cause this.
This happens for instance in estimate_timings_from_pts(), which causes the
parser to be reopened on the same stream.
Anton Khirnov [Tue, 7 Feb 2012 10:03:33 +0000 (11:03 +0100)]
avconv: saner output video timebase.
r_frame_rate should in theory have something to do with input framerate,
but in practice it is often made up from thin air by lavf. So unless we
are targeting a constant output framerate, it's better to just use input
stream timebase.
* qatar/master: (34 commits)
mlp_parser: fix the channel mask value used for the top surround channel
vorbisenc: check all allocations for failure
roqaudioenc: return AVERROR codes instead of -1
roqaudioenc: set correct bit rate
roqaudioenc: use AVCodecContext.frame_size correctly.
roqaudioenc: remove unneeded sample_fmt check
ra144enc: use int16_t* for input samples rather than void*
ra144enc: set AVCodecContext.coded_frame
ra144enc: remove unneeded sample_fmt check
nellymoserenc: set AVCodecContext.coded_frame
nellymoserenc: improve error checking in encode_init()
nellymoserenc: return AVERROR codes instead of -1
libvorbis: improve error checking in oggvorbis_encode_init()
mpegaudioenc: return AVERROR codes instead of -1
libfaac: improve error checking and handling in Faac_encode_init()
avutil: add AVERROR_UNKNOWN
check for coded_frame allocation failure in several audio encoders
audio encoders: do not set coded_frame->key_frame.
g722enc: check for trellis data allocation error
libspeexenc: export encoder delay through AVCodecContext.delay
...
Justin Ruggles [Wed, 22 Feb 2012 21:20:49 +0000 (16:20 -0500)]
roqaudioenc: use AVCodecContext.frame_size correctly.
It is not allowed to change mid-stream like it does currently. Instead we need
to buffer the first 8 frames before returning them as a single packet, then
only return single frame packets after that.
Ronald S. Bultje [Sat, 25 Feb 2012 00:12:18 +0000 (16:12 -0800)]
matroska: don't overwrite string values until read/alloc was succesful.
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* qatar/master:
docs: use -bsf:[vas] instead of -[vas]bsf.
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
lavf: move the packet keyframe setting code.
oggenc: free comment header for all codecs
lcl: error out if uncompressed input buffer is smaller than framesize.
mjpeg: abort decoding if packet is too large.
golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
get_bits: add HAVE_BITS_REMAINING macro.
lavf/output-example: use new audio encoding API correctly.
lavf/output-example: more proper usage of the new API.
tiff: Prevent overreads in the type_sizes array.
tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
apetag: do not leak memory if avio_read() fails
apetag: propagate errors.
SBR DSP x86: implement SSE sbr_hf_g_filt
SBR DSP x86: implement SSE sbr_sum_square_sse
SBR DSP: use intptr_t for the ixh parameter.
Dale Curtis [Fri, 24 Feb 2012 18:17:39 +0000 (13:17 -0500)]
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
Instead of clipping extrasize based on EXTRABYTES, clip based on the
amount of buffer actually left. Without this fix, there are warbles
and other distortions in the test case below.
Anton Khirnov [Wed, 15 Feb 2012 07:38:51 +0000 (08:38 +0100)]
lavf: move the packet keyframe setting code.
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
Justin Ruggles [Thu, 23 Feb 2012 00:31:40 +0000 (19:31 -0500)]
oggenc: free comment header for all codecs
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.
* qatar/master: (40 commits)
swf: check return values for av_get/new_packet().
wavpack: Don't shift minclip/maxclip
rtpenc: Expose the max packet size via an avoption
rtpenc: Move max_packet_size to a context variable
rtpenc: Add an option for not sending RTCP packets
lavc: drop encode() support for video.
snowenc: switch to encode2().
snowenc: don't abuse input picture for storing information.
a64multienc: switch to encode2().
a64multienc: don't write into output buffer when there's no output.
libxvid: switch to encode2().
tiffenc: switch to encode2().
tiffenc: properly forward error codes in encode_frame().
lavc: drop libdirac encoder.
gifenc: switch to encode2().
libvpxenc: switch to encode2().
flashsvenc: switch to encode2().
Remove libpostproc.
lcl: don't overwrite input memory.
swscale: take first/lastline over/underflows into account for MMX.
...
The 32bits targets have been compiled with -mfpmath=sse for proper reference.
sbr_sum_square C /32bits: 82c (unrolled)/102c
C /64bits: 69c (unrolled)/82c
SSE/32bits: 42c
SSE/64bits: 31c
Use of SSE4.1 dpps to perform the final sum is slower.
Not unrolling to perform 8 operations in a loop yields 10 more cycles.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Martin Storsjö [Thu, 23 Feb 2012 09:56:15 +0000 (11:56 +0200)]
rtpenc: Expose the max packet size via an avoption
This allows opting for a lower MTU than what the AVIOContext
indicated, and allows writing into outputs that don't indicate
an MTU at all (such as plain files, which is useful for testing).
This also allows querying for the MTU via the avoption.
projects / ffmpeg / log