All decode_channel_map calls together can easily read
more data than the amount of padding available.
Thus below patch adds an input length check before reading them.
Fixes some invalid reads with sample from
http://bugzilla.mplayerhq.hu/show_bug.cgi?id=1138
* qatar/master:
vorbisdec: Rename silly "class_" variable to plain "class".
simple_idct_alpha: Drop some useless casts.
Simplify av_log_missing_feature().
ac3enc: remove check for mismatching channels and channel_layout
If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout.
If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother.
cosmetics: indentation
Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open().
aacdec: remove sf_scale and sf_offset.
aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec.
Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere.
Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway.
qpeg: use reget_buffer() in decode_frame()
ultimotion: use reget_buffer() in ulti_decode_frame()
smacker: remove unnecessary call to avctx->release_buffer in decode_frame()
avparser: don't av_malloc(0).
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Instead, scalefactors are adjusted by the offset amount, removing the need
for sf_scale, and the MDCT scales are adjusted to compensate for the higher
scalefactors. Floating-point output will be handled by modifying the MDCT
scales.
erroring out. A magnitude of 100 corresponds to 2^25 so the will most
likely result in clipped output anyway.
None of the conformance streams fall in the range that need to be clipped.
* qatar/master:
Handle unicode file names on windows
rtp: Rename the open/close functions to alloc/free
Lowercase all ff* program names.
Refer to ff* tools by their lowercase names.
NOT Pulled Replace more FFmpeg instances by Libav or ffmpeg.
Replace `` by $() syntax in shell scripts.
patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled Remove stray libavcore and _g binary references.
vorbis: Rename decoder/encoder files to follow general file naming scheme.
aacenc: Fix whitespace after last commit.
cook: Fix small typo in av_log_ask_for_sample message.
aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
Remove RDFT dependency from AAC decoder.
Add some debug log messages to AAC extradata
Fix mov debug (u)int64_t format strings.
bswap: use native types for av_bwap16().
doc: FLV muxing is supported.
applehttp: Handle AES-128 encrypted streams
Add a protocol handler for AES CBC decryption with PKCS7 padding
doc: Mention that DragonFly BSD requires __BSD_VISIBLE set
Conflicts:
ffplay.c
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Use av_log_ask_for_sample() to request samples from users.
Make av_log_ask_for_sample() accept a variable number of arguments.
vqavideo: We no longer need to ask for version 1 samples.
aacdec: indentation cosmetics
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Partially merged:flvdec: Allow parsing keyframes metadata without seeking in most cases
Error out if vaapi is not found
avio: undeprecate av_url_read_fseek/fpause under nicer names
libvo-*: Don't use deprecated sample format names and enum names
DUPLICATE flvdec: Fix support for flvtool2 "keyframes based" generated index
DUPLICATE libavcodec: Use "const enum AVSampleFormat[]" in AVCodec initialization
Fix the conversion of AV_SAMPLE_FMT_FLT and _DBL to AV_SAMPLE_FMT_S32.
Convert some undefined 1<<31 shifts into 1U<<31.
Conflicts:
configure
libavcodec/libvo-aacenc.c
libavcodec/libvo-amrwbenc.c
libavformat/flvdec.c
Marged-by: Michael Niedermayer <michaelni@gmx.at>
According to ISO 9899:1999 S 6.5.7/4:
The result of E1 << E2 is E1 left-shifted E2 bit positions; vacated bits
are filled with zeros. If E1 has an unsigned type, the value of the
result is E1× 2^E2, reduced modulo one more than the maximum value
representable in the result type. If E1 has a signed type and
nonnegative value, and E1× 2^E2 is representable in the result type, then
that is the resulting value; otherwise, the behavior is undefined.
* qatar/master:
psymodel: extend API to include PE and bit allocation.
avio: always compile dyn_buf functions
Remove unnecessary parameter from ff_thread_init() and fix behavior
Revert "aac_latm_dec: use aac context and aac m4ac"
configure: tell user if libva is enabled like the rest of external libs.
Add silence support for AV_SAMPLE_FMT_U8.
avio: make URL_PROTOCOL_FLAG_NESTED_SCHEME internal
avio: deprecate av_url_read_seek
avio: deprecate av_url_read_pause
ac3enc: NEON optimised extract_exponents
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These windows do not really belong in fft/mdct files and were
easily confused with the similarly named tables used by rdft.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This function is not tightly coupled to mdct, and it's in the way
of making a fixed-point mdct implementation.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This removes the rather pointless wrappers (one not even inline)
for calling the fft_calc and related function pointers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 36864ac354)
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672)
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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)
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f162e988aa)
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Muxlength does not include the 3 bytes of AudioSyncStream() before the
AudioMuxElement(). If these three bytes are not accounted for the last three
bytes of the LATM packet are sent back to the decoder again.
Fixes issue244/mux2.share.ts
Originally committed as revision 25685 to svn://svn.ffmpeg.org/ffmpeg/trunk
Spotted by Alex after Carl Eugen found errors some samples. There no errors or
noticeable artifacts in the samples I used during development.
Originally committed as revision 25676 to svn://svn.ffmpeg.org/ffmpeg/trunk
The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }
Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use avctx in all called functions. This allows passing a NULL AACContext
for LATM since the AACContext is only used in output_configure() which
is skipped for LATM parsing.
Originally committed as revision 25641 to svn://svn.ffmpeg.org/ffmpeg/trunk
aac_decode_frame() remains as AVPacket handling a wrapper. The actual
decoding function takes a GetBitContext as input and will be used be the
AAC LATM decoder to avoid copying the unaligned AAC bitstream.
Originally committed as revision 25640 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used by the latm decoder to avoid AACContext changes during
audio specific config parsing.
Originally committed as revision 25638 to svn://svn.ffmpeg.org/ffmpeg/trunk
For a PCE based configuration map the channels solely based on tags.
For an indexed configuration map the channels solely based on position.
This works with all known exotic samples including al17, elem_id0, bad_concat,
and lfe_is_sce.
Originally committed as revision 25098 to svn://svn.ffmpeg.org/ffmpeg/trunk
The AAC decoder and ADTS-to-ASC BSF both require the header decoder
but not full parsing capabilities.
Originally committed as revision 24217 to svn://svn.ffmpeg.org/ffmpeg/trunk
It created false positives on seeks and where the first frame is STOP or SHORT.
It failed to warn in illegal SHORT->LONG transitions. In general it created
much confusion and many junk bug reports from the users.
Originally committed as revision 24214 to svn://svn.ffmpeg.org/ffmpeg/trunk
