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Commit Graph

138 Commits

Author SHA1 Message Date
Michael Niedermayer
434f248723 Merge remote branch 'qatar/master'
* qatar/master: (22 commits)
  ac3enc: move extract_exponents inner loop to ac3dsp
  avio: deprecate url_get_filename().
  avio: deprecate url_max_packet_size().
  avio: make url_get_file_handle() internal.
  avio: make url_filesize() internal.
  avio: make url_close() internal.
  avio: make url_seek() internal.
  avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
  avio: make url_write() internal.
  avio: make url_read_complete() internal.
  avio: make url_read() internal.
  avio: make url_open() internal.
  avio: make url_connect internal.
  avio: make url_alloc internal.
  applehttp: Merge two for loops
  applehttp: Restructure the demuxer to use a custom AVIOContext
  applehttp: Move finished and target_duration to the variant struct
  aacenc: reduce the number of loop index variables
  avio: deprecate url_open_protocol
  avio: deprecate url_poll and URLPollEntry
  ...

Conflicts:
	libavformat/applehttp.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-05 02:31:56 +02:00
Justin Ruggles
e05a3ac713 ac3enc: select bandwidth based on bit rate, sample rate, and number of
full-bandwidth channels.

This reduces high-frequency artifacts and improves the quality of the lower
frequency audio at low bit rates.
2011-04-03 20:59:14 -04:00
Michael Niedermayer
2cae9809e2 Merge remote branch 'qatar/master'
* qatar/master:
  fate: fix partial run when no samples path is specified
  ARM: NEON fixed-point forward MDCT
  ARM: NEON fixed-point FFT
  lavf: bump minor version and add an APIChanges entry for avio changes
  avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
  avio: make url_fdopen internal.
  avio: make url_open_dyn_packet_buf internal.
  avio: avio_ prefix for url_close_dyn_buf
  avio: avio_ prefix for url_open_dyn_buf
  avio: introduce an AVIOContext.seekable field
  ac3enc: use generic fixed-point mdct
  lavfi: add fade filter
  Change yadif to not use out of picture lines.
  lavc: deprecate AVCodecContext.antialias_algo
  lavc: mark mb_qmin/mb_qmax for removal on next major bump.

Conflicts:
	doc/filters.texi
	libavcodec/ac3enc_fixed.h
	libavcodec/ac3enc_float.h
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/vf_fade.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-04 02:15:12 +02:00
Mans Rullgard
79997def65 ac3enc: use generic fixed-point mdct
This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation.  The checksum changes are due to
different rounding in the MDCT.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-04-03 19:01:53 +01:00
Michael Niedermayer
116758a358 Fix yuvj420p scaling artefact, issue1108.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-03 16:40:11 +02:00
Anssi Hannula
78431098f9 lavf: inspect more frames for fps when container time base is coarse
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.

Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-03 03:18:06 +02:00
Baptiste Coudurier
93dfda8896 In ipod/mov/mp4 muxer, always write esds descriptor length using 4 bytes,
ipod shuffle doesn't support anything else.
2011-03-30 14:09:09 -07:00
Baptiste Coudurier
efdad9fbc7 In mov muxer, fix yuv range in avid atoms used by dnxhd. 2011-03-30 08:48:08 -07:00
Michael Niedermayer
3c8493074b Merge remote-tracking branch 'newdev/master'
* newdev/master:
  dsputil: allow to skip drawing of top/bottom edges.
  Split fate-psx-str-v3 into a video-only and audio-only test.

Conflicts:
	libavcodec/dsputil.c
	libavcodec/mpegvideo.c
	libavcodec/snow.c
	libavcodec/x86/dsputil_mmx.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-03-27 01:40:18 +01:00
Ronald S. Bultje
c56e618b4b Split fate-psx-str-v3 into a video-only and audio-only test. 2011-03-26 16:39:22 -04:00
Peter Ross
f5607c8361 Make the hflip filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats 2011-03-26 13:24:41 +11:00
Peter Ross
af55573379 Make the crop filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats 2011-03-26 13:24:36 +11:00
Peter Ross
3e2523db20 libswcale: PIX_FMT_BGR48LE and PIX_FMT_BGR48BE scaler implementation 2011-03-26 13:24:32 +11:00
Justin Ruggles
e6e9823488 Add apply_window_int16() to DSPContext with x86-optimized versions and use it
in the ac3_fixed encoder.
2011-03-22 21:08:30 -04:00
Mans Rullgard
2a569799a9 fate: update wmv8-drm reference
This updates the wmv8-drm reference after c47d383.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-22 11:07:46 +00:00
Ronald S. Bultje
c47d383502 vc1: make P-frame deblock filter bit-exact. 2011-03-21 21:28:17 -04:00
Mans Rullgard
487fef2dcc asf: update seek test reference
This updates the seek test reference to match de11ee9.  Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned.  With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-17 19:51:28 +00:00
Justin
323e6fead0 ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage.
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
2011-03-14 08:45:26 -04:00
Peter Ross
e211e255aa bink: prevent overflows within binkidct by using int-sized intermediate array
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-25 15:24:35 -05:00
Justin Ruggles
1108f8998c vmdaudio: output 8-bit audio as AV_SAMPLE_FMT_U8.
There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 21:52:51 -05:00
Justin Ruggles
9b73f78600 vmdaudio: output audio samples for standalone silent blocks.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 21:04:51 -05:00
Justin Ruggles
5b54d4b376 ac3enc: fix bug in stereo rematrixing decision.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-16 23:39:57 +00:00
Justin Ruggles
50d7140441 ac3enc: change default floor code to 7.
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-15 21:40:42 +00:00
Baptiste Coudurier
646739a0a8 Fix qtrle regression test, actually test qtrle.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-11 23:47:09 +00:00
Justin Ruggles
c3beafa0f1 ac3enc: Change EXP_DIFF_THRESHOLD to 500.
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder.  I tested lowering in
increments of 100.  From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-02 20:00:43 +00:00
Mans Rullgard
79dca23dc2 Update mpegts test reference
The output was changed by a7827a17c6.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-28 17:02:54 +00:00
Georgi Chorbadzhiyski
535638b55f mpegtsenc: set reserved bits to 1 in PCR field
The reserved bits between PCR base and extension fields must be
set to 1.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-26 00:02:42 +00:00
Mans Rullgard
e63dd5fb04 fate: add h264 test for extreme cases in planar prediction
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-24 22:26:13 +00:00
Mans Rullgard
76edf2c137 fate: add lossless h264 test
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-22 03:08:21 +00:00
Mans Rullgard
f4b1e21a63 fate: make lavfi tests output only md5
Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly.  This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
2011-01-22 00:30:12 +00:00
Justin Ruggles
a4f5af13fb Add regression test for stereo s16le in voc.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-01-19 12:51:42 +00:00
Baptiste Coudurier
90603f7c93 Update smc fate ref due to r26310
Originally committed as revision 26342 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-14 22:32:26 +00:00
Justin Ruggles
dc7e07ac1f Add stereo rematrixing support to the AC-3 encoders.
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.

Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-08 23:21:17 +00:00
Vitor Sessak
87c1410d11 Add a FATE test for Playstation STR version 3
Originally committed as revision 26231 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-05 19:53:16 +00:00
Justin Ruggles
6fd96d1a85 Change the AC-3 encoder to use floating-point.
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.

Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-04 11:53:44 +00:00
Justin Ruggles
ec44dd5fc2 Change the default dB-per-bit code from 2 to 3.
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.

Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.

Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-29 19:17:22 +00:00
Aurelien Jacobs
2c77c90684 add SubRip decoder
Originally committed as revision 26119 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-28 23:52:53 +00:00
Stefano Sabatini
b567020943 Add copy filter, useful for testing the avfilter_draw_slice() copy
code.

Originally committed as revision 26112 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-28 01:01:09 +00:00
Justin Ruggles
295ab2af6e Change FIX15() back to clipping to -32767..32767.
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.

Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-21 21:18:58 +00:00
Reimar Döffinger
eb066a4ce9 Discard partial packet of last frame for fate-wmv8-drm to avoid test fails
due to VC-1 decoder overreads resulting in different output.

Originally committed as revision 26055 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-18 23:11:31 +00:00
Reimar Döffinger
853395b913 Add test for ASF -cryptokey that tests only demuxing, but both audio and video
to complement the existing video-only decode test.

Originally committed as revision 26054 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-18 16:06:56 +00:00
Reimar Döffinger
bf09a01981 Change ASF demuxer to return incomplete last packets.
Whether the behaviour for streams using scrambling makes sense
is unclear.

Originally committed as revision 26053 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-18 13:18:52 +00:00
Justin Ruggles
8c634b707b Update the test references for lavf-rm and seek-ac3_rm.
The references changed due to r25956.

Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 16:14:52 +00:00
Justin Ruggles
918cd2255c Simplify fix15().
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.

Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 14:51:02 +00:00
Michael Chinen
475ae04a27 Add a FLAC parser.
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.

Patch by Michael Chinen [mchinen at gmail]

Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 14:50:50 +00:00
Carl Eugen Hoyos
ad556addfd Fix h264-conformance-frext-frext_mmco4_sony_b conformance test.
This includes a revert of r25840

Originally committed as revision 25842 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-29 10:35:57 +00:00
Baptiste Coudurier
79561f0ed4 Update fate h264 test due to r25824, this file has 2 frames delay
Originally committed as revision 25840 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-28 23:09:30 +00:00
Baptiste Coudurier
9d9c3e1a70 Make DNxHD encoder produce files that are strictly VC-3 compatible
Originally committed as revision 25756 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-16 01:00:55 +00:00
Vitor Sessak
bbf07bf9b8 Remove now unused file (should have been part of commit r25735)
Originally committed as revision 25736 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-12 20:15:36 +00:00
Vitor Sessak
c51722bf97 Test 4XM decoding (and not only demuxing) in FATE tests
Originally committed as revision 25735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-12 20:04:41 +00:00