The AVSampleFormat list of sample_fmts_s16p is missing the trailing "P" for planar formats. AV_SAMPLE_FMT_S16 vs AV_SAMPLE_FMT_S16P
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
AV_CODEC_ID_ADPCM_EA_R1/R2/R3 all use an internal offset. For some
samples there is padding between the offset table and ADPCM data.
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These ADPCM codecs include a per-frame flag that enables a raw 16-bit mode. Therefore
the the number of samples returned by get_nb_samples() is only ever approximate.
Fixes ticket #3460.
Signed-off-by: Peter Ross <pross@xvid.org>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes use of uninitialized memory
Fixes: msan_uninit-mem_7f9b9902ed90_7462_new_alaw.voc
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There are several containers that support adpcm_thp (Also known as Gamecube DSP)
streams, but only thp files contain the coeff table and previous sample inside
each frame.
Some don't even contain previous sample information at all.
This change will make it easier to implement demuxers for said containers
without having to create a new decoder.
Signed-off-by: James Almer <jamrial@gmail.com>
* commit 'e57daa876bf0cf50782550e366e589441cd8c2bd':
adpcm: decode directly to the user-provided AVFrame
ac3: decode directly to the user-provided AVFrame
aac: decode directly to the user-provided AVFrame
8svx: decode directly to the user-provided AVFrame
Conflicts:
libavcodec/8svx.c
libavcodec/ac3dec.c
libavcodec/adpcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f':
adxdec: use planar sample format
adpcmdec: use planar sample format for adpcm_thp
adpcmdec: use planar sample format for adpcm_ea_xas
adpcmdec: use planar sample format for adpcm_ea_r1/r2/r3
adpcmdec: use planar sample format for adpcm_xa
adpcmdec: use planar sample format for adpcm_ima_ws for vqa version 3
adpcmdec: use planar sample format for adpcm_4xm
adpcmdec: use planar sample format for adpcm_ima_wav
adpcmdec: use planar sample format for adpcm_ima_qt
pcmdec: use planar sample format for pcm_lxf
mace: use planar sample format
atrac1: use planar sample format
build: non-x86: Only compile mpegvideo optimizations when necessary
rtpdec_mpeg4: au_headers is a single array, simple av_free is enough
avcodec: free extended_data instead address of it
fate: Add tests of the ff_make_absolute_url function
url: Handle relative urls starting with two slashes
url: Handle relative urls being just a new query string
url: Don't treat slashes in query parameters as directory separators
Conflicts:
libavcodec/adxdec.c
libavcodec/mips/Makefile
libavcodec/pcm.c
libavcodec/utils.c
libavformat/Makefile
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bfcd4b6a1691d20aebc6d2308424c2a88334a9f0':
adpcmdec: set AVCodec.sample_fmts
twinvq: use planar sample format
ralf: use planar sample format
mpc7/8: use planar sample format
iac/imc: use planar sample format
dcadec: use float planar sample format
cook: use planar sample format
atrac3: use float planar sample format
apedec: output in planar sample format
8svx: use planar sample format
Conflicts:
libavcodec/8svx.c
libavcodec/dcadec.c
libavcodec/mpc7.c
libavcodec/mpc8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avplay: use libavresample for sample format conversion and channel mixing
Fix compilation with YASM/NASM without AVX support.
WMAL: do not output last frame again if nothing was decoded in current packet
WMAL: do not start decoding if frame does not end in current packet
adpcm-thp: fix invalid array indexing
ppc: add const where needed in scalarproduct_int16_altivec()
ppc: remove shift parameter from scalarproduct_int16_altivec()
ppc: dsputil: do unaligned block accesses correctly
dvenc: do not call dsputil functions with stride not a multiple of 16
APIchanges: fill in some dates and commit hashes
Conflicts:
doc/APIchanges
ffplay.c
libavcodec/adpcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
adpcm: Clip step_index values read from the bitstream at the beginning of each frame.
oma: don't read beyond end of leaf_table.
doxygen: Remove documentation for non-existing parameters; misc small fixes.
Indeo3: fix crashes on corrupt bitstreams.
msmpeg4: Replace forward declaration by proper #include.
segment: implement wrap around
avf: reorder AVStream and AVFormatContext
aacdec: Remove erroneous reference to global gain from the out of bounds scalefactor error message.
Conflicts:
libavcodec/indeo3.c
libavformat/avformat.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
ipmovie: do not read audio packets before the codec is known
truemotion2: check size before GetBitContext initialisation
avio: Only do implicit network initialization for network protocols
avio: Add an URLProtocol flag for indicating that a protocol uses network
adpcm: ADPCM Electronic Arts has always two channels
matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
fate: Add missing reference file from 9b4767e4.
mov: Support MOV_CH_LAYOUT_USE_DESCRIPTIONS for labeled descriptions.
4xm: Prevent buffer overreads.
mjpegdec: parse RSTn to prevent skipping other data in mjpeg_decode_scan
vp3: add fate test for non-zero last coefficient
vp3: fix streams with non-zero last coefficient
swscale: remove unused U/V arguments from yuv2rgb_write().
timer: K&R formatting cosmetics
lavf: cosmetics, reformat av_read_frame().
lavf: refactor av_read_frame() to make it easier to understand.
Report an error if pitch_lag is zero in AMR-NB decoder.
Revert "4xm: Prevent buffer overreads."
4xm: Prevent buffer overreads.
4xm: pass the correct remaining buffer size to decode_i2_frame().
...
Conflicts:
libavcodec/4xm.c
libavcodec/mjpegdec.c
libavcodec/truemotion2.c
libavformat/ipmovie.c
libavformat/mov_chan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
build: fix standalone compilation of OMA muxer
build: fix standalone compilation of Microsoft XMV demuxer
build: fix standalone compilation of Core Audio Format demuxer
kvmc: fix invalid reads
4xm: Add a check in decode_i_frame to prevent buffer overreads
adpcm: fix IMA SMJPEG decoding
options: set minimum for "threads" to zero
bsd: use number of logical CPUs as automatic thread count
windows: use number of CPUs as automatic thread count
linux: use number of CPUs as automatic thread count
pthreads: reset active_thread_type when slice thread_init returrns early
v410dec: include correct headers
Drop ALT_ prefix from BITSTREAM_READER_LE name.
lavfi: always build vsrc_buffer.
ra144enc: zero the reflection coeffs if the filter is unstable
sws: readd PAL8 to isPacked()
mov: Don't stick the QuickTime field ordering atom in extradata.
truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
Conflicts:
configure
libavcodec/4xm.c
libavcodec/avcodec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavformat/Makefile
libswscale/swscale_internal.h
Merged-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>
* 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: (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>
Add Multimedia Wiki link.
Mark dead links with [dead]. Some can still be accessed through archive.org.
Update URLs for pages which have moved.
Replace duplicated links in adpcmenc.c with a note to see the ADPCM decoder
reference documents.
* 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>
Unfortunately the output buffer size check assumes that the
input buffer is never over-consumed, thus this actually
also allowed to write outside the output buffer if "lucky".
Based on:
git.videolan.org/ffmpeg.git
commit 701d0eb185
* 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>
Unfortunately the output buffer size check assumes that the
input buffer is never over-consumed, thus this actually
also allowed to write outside the output buffer if "lucky".
None of these symbols should be accessed directly, so declare them as
hidden.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d36beb3f69)
Wraparound in ssd is mainly avoided by subtracting the ssd of the
best node from all the others once it has grown large enough.
If using very large trellis sizes (e.g. -trellis 15), the frontier
is so large that the difference between the best and the worst is
large enough to cause wraparound, even if the ssd of the best one
is subtracted regularly.
When using -trellis 10 on a 30 second sample, this causes only a slight
slowdown, from 61 to 64 seconds.
Originally committed as revision 25858 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the wording consistent with how people usually talk about heaps.
Originally committed as revision 25775 to svn://svn.ffmpeg.org/ffmpeg/trunk
This increases the PSNR slightly (about 0.1 dB) for trellis sizes
below 8, and gives equal PSNR for sizes above that.
Originally committed as revision 25769 to svn://svn.ffmpeg.org/ffmpeg/trunk
This lowers the run time from 158 to 21 seconds, for -trellis 8
with a 30 second sample on my machine.
This requires 64 KB additional memory.
Originally committed as revision 25768 to svn://svn.ffmpeg.org/ffmpeg/trunk
By not looking for the exactly largest node, we avoid an O(n) seek through
the leaf nodes. Just pick one (not the same one every time) and try replacing
that node with the new one.
For -trellis 8, this lowers the run time from 190 to 158 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25733 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having to memmove the large parts of the array when inserting into
it.
For -trellis 8, this lowers the run time from 245 seconds to 190 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25731 to svn://svn.ffmpeg.org/ffmpeg/trunk
