Duration computation can be simplified because number of PCM blocks is
only allowed to be a multiple of 8.
Signed-off-by: James Almer <jamrial@gmail.com>
Padding before the first sync word can be very large for DTS-in-WAV
streams. There is no reason to include this padding in parsed packet.
Signed-off-by: James Almer <jamrial@gmail.com>
Parse core frame size directly when searching for frame end instead of
using value extracted from previous frame.
Account for unused bits when calculating sync word distance for 14-bit
streams to avoid alias sync detection.
Parse EXSS frame size and skip over EXSS frame to avoid alias sync
detection.
Signed-off-by: James Almer <jamrial@gmail.com>
The parser only parses the core, and thus has a duration relative
to the core sample rate only, not the actual stream sample rate.
FATE references changed due to now correct timestamps.
The parser only reads the dca core sample rate, which is limited to a
maximum of 48000 Hz, while X96 and HD extensions can increase the sample
rate up to 192000 Hz.
This change prevents the parser and decoder fighting over the sample rate,
potentially confusing user applications. This also fixes sample rate
display of >48000Hz files with ffmpeg/ffprobe when using libdcadec.
Fixes ticket #4397
The previous version checked for 14-bit streams and did not properly
work across buffer boundaries.
Use the 64-bit parser state to make extended sync word detection work
across buffer boundary and check the extended sync word for 16-bit LE
and BE core streams to reduce probability of alias sync detection.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Check extended sync word for 16-bit LE and BE core streams to reduce
probability of alias sync detection. Previously sync word extension was
checked only for 14-bit streams (and this check did not properly work
across buffer boundary).
Use 64-bit parser state to make extended sync word detection work across
buffer boundary.
This is sufficient to make the sample in ticket #4492 parse
successfully.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '25f613f8be3b51e4396b93cda131e4631ba54302':
dca: Move syncword definitions to a separate header
Conflicts:
libavcodec/dca_parser.c
libavformat/dtsdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A change in framesize caused a perpetual loss of synchronization.
So read (and use) the frame size from the frame header instead of
setting it only once.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* commit '1db6a080bddd14fed6b29140ecd2e21e42b1c022':
dca: Move ff_dca_convert_bitstream() to the DCA common code
vdpau: wrap codec specific functions in appropiate #ifs
Conflicts:
libavcodec/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8ac0f6767bf63d3e6b308ee6648ff02598b81e03':
dcadec: allow the decoder to change the channel layout mid-stream
cook: use av_dlog() for debug logging instead of av_log() with AV_LOG_ERROR
cook: move samples_per_frame from COOKSubpacket to where it is used
cook: use av_get_channel_layout_nb_channels() instead of cook_count_channels()
cook: reverse a condition so that the code makes more sense
cook: remove unneeded COOKContext variable, sample_rate
cook: remove unneeded COOKContext variable, bit_rate
cook: use AVCodecContext.channels instead of keeping a private copy
bmvaudio: set channel layout at init() rather than validating it
atrac1: do not keep a copy of channel count in the private context
dsicinaudio: set channels and channel layout
g722dec: set channel layout at initialization instead of validating it
amrwbdec: set channels, channel_layout, and sample_rate
amrnbdec: set channels, channel_layout, and sample_rate
dca_parser: allow the parser to change the sample rate
lavc: check channel count after decoder init
lavc: move SANE_NB_CHANNELS to internal.h and use it in the PCM decoders
Conflicts:
libavcodec/dcadec.c
libavcodec/pcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dca: Switch dca_sample_rates to avpriv_ prefix; it is used across libs
ARM: use =const syntax instead of explicit literal pools
ARM: use standard syntax for all LDRD/STRD instructions
fft: port FFT/IMDCT 3dnow functions to yasm, and disable on x86-64.
dct-test: allow to compile without HAVE_INLINE_ASM.
x86/dsputilenc: bury inline asm under HAVE_INLINE_ASM.
dca: Move tables used outside of dcadec.c to a separate file.
dca: Rename dca.c ---> dcadec.c
x86: h264dsp: Remove unused variable ff_pb_3_1
apetag: change a forgotten return to return 0
Conflicts:
libavcodec/Makefile
libavcodec/dca.c
libavcodec/x86/fft_3dn.c
libavcodec/x86/fft_3dn2.c
libavcodec/x86/fft_mmx.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/h264_ps.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
The dca parser needs to check that the framesize is actually of a complete
frame, and not of a partial DTS-HD frame, which is not constant size, and
thus the check would fail.
(cherry picked from commit ebc0ccb9af59b78732e82c09f8c90b1d46b478e0)
Review-by: Benjamin Larsson
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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)
Passing an explicit filename to this command is only necessary if the
documentation in the @file block refers to a file different from the
one the block resides in.
Originally committed as revision 22921 to svn://svn.ffmpeg.org/ffmpeg/trunk
Otherwise doxygen complains about ambiguous filenames when files exist
under the same name in different subdirectories.
Originally committed as revision 16912 to svn://svn.ffmpeg.org/ffmpeg/trunk
