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22nd October 2007, 02:40 | #161 | Link |
Huh?
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I want to open a DXA video in Avisynth, do I need anything else than FFMPEGSource?
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26th October 2007, 01:56 | #162 | Link |
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This Is A Version Only Intended For Testing, Idiots Should Skip This Post
This test version uses a new way to decode audio without using a huge uncompressed audio cache. Matroska files need no additional space and all other containers have the original demuxed audio track stored in c:\ffmpegsource.cache (this test version only). The decoding results seems to be identical (dithering/other parts of codecs that prevent it but makes no difference) when run multiple times when compared to itself. It does however differ a lot from ffmpegsource(atrack=-1) and I have no idea why that is. There are at least no audible artifacts (to me) so report if it works properly. Usage: FFAudioSource(string source) Link
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27th October 2007, 13:39 | #163 | Link |
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@Myrsloik,
Recently MLP decoding was added: ffmpeg-mailinglist. Could you add this to FFAudioSource? I would be very grateful! |
27th October 2007, 18:10 | #164 | Link |
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I tried to compile a newer ffmpeg the usual way but I get this error:
Code:
gcc -shared -Wl,--output-def,avformat-51.def -Wl,--enable-runtime-pseudo-reloc -Wl,--enable-auto-image-base -Wl,--warn-common -Wl,-rpath-link,"/c/libraries/ffmpeg"/libavcodec -Wl,-rpath-link,"/c/libraries/ffmpeg"/libavformat -Wl,-rpath-link,"/c/libraries/ffmpeg"/libavutil -o avformat-51.dll allformats.o cutils.o os_support.o sdp.o utils.o avio.o aviobuf.o raw.o aiff.o riff.o amr.o apc.o ape.o asf.o asfcrypt.o au.o avidec.o avs.o vocdec.o voc.o bethsoftvid.o c93.o daud.o dsicin.o dv.o dxa.o electronicarts.o ffm.o flic.o flvdec.o 4xm.o gifdec.o gxf.o idcin.o img2.o ipmovie.o matroskadec.o matroska.o mm.o mmf.o mov.o isom.o mp3.o mpc.o mpeg.o mpegts.o mtv.o mxf.o nsvdec.o nutdec.o nut.o nuv.o ogg2.o oggparsevorbis.o oggparsetheora.o oggparseflac.o oggparseogm.o rtsp.o rmdec.o idroq.o segafilm.o siff.o smacker.o sol.o psxstr.o swf.o thp.o tiertexseq.o tta.o txd.o sierravmd.o wav.o wc3movie.o westwood.o wv.o yuv4mpeg.o file.o http.o rtpproto.o tcp.o udp.o -L"/c/libraries/ffmpeg"/libavutil -lavutil -lavcodec -L"/c/libraries/ffmpeg"/libavcodec -lz -lm -la52 -lfaad -lws2_32 c:/libraries/ffmpeg/libavcodec/libavcodec.a(cscd.o): In function `decode_frame': c:/libraries/ffmpeg/libavcodec/cscd.c:161: undefined reference to `lzo1x_decode' c:/libraries/ffmpeg/libavcodec/libavcodec.a(nuv.o): In function `decode_frame': c:/libraries/ffmpeg/libavcodec/nuv.c:168: undefined reference to `lzo1x_decode' c:/libraries/ffmpeg/libavcodec/libavcodec.a(cook.o): In function `scalar_dequant_float': c:/libraries/ffmpeg/libavutil/random.h:45: undefined reference to `av_random_generate_untempered_numbers' c:/libraries/ffmpeg/libavcodec/libavcodec.a(cook.o): In function `cook_decode_init': c:/libraries/ffmpeg/libavcodec/cook.c:1069: undefined reference to `av_init_random' c:/libraries/ffmpeg/libavcodec/libavcodec.a(mpc.o): In function `mpc7_decode_frame': c:/libraries/ffmpeg/libavutil/random.h:45: undefined reference to `av_random_generate_untempered_numbers' c:/libraries/ffmpeg/libavcodec/libavcodec.a(mpc.o): In function `mpc7_decode_init': c:/libraries/ffmpeg/libavcodec/mpc.c:83: undefined reference to `av_init_random' c:/libraries/ffmpeg/libavcodec/libavcodec.a(nellymoserdec.o): In function `decode_init': c:/libraries/ffmpeg/libavcodec/nellymoserdec.c:401: undefined reference to `av_init_random' c:/libraries/ffmpeg/libavcodec/libavcodec.a(nellymoserdec.o): In function `nelly_decode_block': c:/libraries/ffmpeg/libavutil/random.h:45: undefined reference to `av_random_generate_untempered_numbers' collect2: ld returned 1 exit status make[1]: *** [avformat-51.dll] Error 1 make[1]: Leaving directory `/c/libraries/ffmpeg/libavformat' make: *** [lib] Error 2
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28th October 2007, 04:25 | #165 | Link |
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New compile of the test version. Includes a recent ffmpeg and it's now statically linked (flac cache not enabled since it didn't want to link).
Link
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30th October 2007, 22:26 | #168 | Link |
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@Myrsloik,
Kurtnoise created a patch for E-AC3/MLP (ffmpeg)! Could you incorporate that in FFAudioSource? patch: http://kurtnoise.free.fr/index.php?dir=misc/ Thanks again! |
30th October 2007, 22:35 | #169 | Link |
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If I did you would have no motivation to get it integrated into the main branch. The source is still available if someone really wants to make their own build with an "improved" ffmpeg and I don't mind answering compilation questions... AS LONG AS YOU CLEARLY LABEL IT AS NOT COMPILED BY ME. You could even get your own link in the first post! Apply now!
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1st November 2007, 11:13 | #171 | Link | |
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Quote:
np: Prefuse-73 - Every Party Has A Winner And A Loser (Unrest)
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1st November 2007, 22:16 | #173 | Link |
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@Kurtnoise13,
http://rapidshare.com/files/66801815...mples.zip.html contains some samples: AAC, MP4/AAC and MLP. The AAC, MP4/AAC played, but the sound was a bit garbled. The MLP (44.1 kHz, 16 bit) didn't play at all, it gave the error: 'Audio decoding error'. Script: Code:
v = BlankClip() a = FFmpegSource("D:\Install\dvd-audio\TheCorrs\DVDAExplorer\Acapela1.mlp", vtrack=-2, atrack=-1) AudioDub(v,a) |
2nd November 2007, 00:33 | #174 | Link |
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The reason is very simple, I'm lazy.
Actually libflac has a nice api that allows sample accurate seeking all the time and is extremely simple to use. With libavformat seeking is much less fun and where you end up is usually a bit uncertain (if it wasn't, someone else probably would've made ffmpegsource before me).
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2nd November 2007, 14:45 | #175 | Link | |
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Quote:
About aac multichannel files, dunno why it doesn't work properly actually...I'll try something else later. |
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2nd November 2007, 22:37 | #176 | Link | |
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Quote:
One additional request though 24 bit stuff is reported as 16 by AviSynth. I'm not sure whether it is converted to 16 bit down the road, or if the patch doesn't support 24 bit. I got an example here: http://rapidshare.com/files/67023394...ples2.zip.html Perhaps you can look at it. If the patch doesn't support it, i will contact Ian Caulfield. |
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2nd November 2007, 22:49 | #177 | Link |
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From avcodec.h:
Code:
/* Currently unused, may be used if 24/32 bits samples are ever supported. */ /* all in native-endian format */ enum SampleFormat { SAMPLE_FMT_NONE = -1, SAMPLE_FMT_U8, ///< unsigned 8 bits SAMPLE_FMT_S16, ///< signed 16 bits SAMPLE_FMT_S24, ///< signed 24 bits SAMPLE_FMT_S32, ///< signed 32 bits SAMPLE_FMT_FLT, ///< float }; Code:
/** * audio sample format * - encoding: Set by user. * - decoding: Set by libavcodec. */ enum SampleFormat sample_fmt; ///< sample format, currently unused
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3rd November 2007, 01:36 | #179 | Link | |
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Quote:
I think we need preserve the original precision (bitdepth) and use a standard channel mapping for decoder-encoder compatibility. Wilbert, don't know this thread? Justin answer about ffmeg bitdepth and multichannel behaviour, there are also a solution to work with NeroAacEnc stdin requested by you in other thread. Last edited by tebasuna51; 3rd November 2007 at 01:44. |
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3rd November 2007, 09:11 | #180 | Link |
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From mlp_parser.c :
Code:
#ifdef CONFIG_AUDIO_NONSHORT avctx->bits_per_sample = mh.group1_bits; if (avctx->bits_per_sample > 16) avctx->sample_fmt = SAMPLE_FMT_S32; #endif Last edited by Kurtnoise; 3rd November 2007 at 09:14. |
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