Hi folks, I'm having a problem with mencoder latest svn version: ffmpeg can produce an AC3 audio file: ffmpeg -y -i video.mkv -acodec ac3 -ac 6 -ab 384k -ar 48000 -f ac3 audioout but mencoder not: mencoder" -ss 0 video.mkv -channels 6 -ovc copy -of rawaudio -mc 0 -noskip -oac lavc -lavcopts acodec=ac3:abitrate=384 -af lavcresample=48000 -srate 48000 -o audioout -v ... ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders dec_audio: Allocating 192000 + 65536 = 257536 bytes for output buffer. FFmpeg's libavcodec audio codec INFO: libavcodec "dca" init OK! [dca @ 0x1009e5bc0]Stream with high frequencies VQ coding AUDIO: 48000 Hz, 6 ch, s16le, 1536.0 kbit/33.33% (ratio: 192000->576000) Selected audio codec: [ffdca] afm: ffmpeg (FFmpeg DTS) ========================================================================== [file] File size is 0 bytes STREAM: [file] audioout STREAM: Description: File STREAM: Author: Albeu STREAM: Comment: based on the code from ??? (probably Arpi) Ignoring video stream! videocodec: framecopy (1280x544 0bpp fourcc=34363248) Building audio filter chain for 48000Hz/6ch/s16le -> 48000Hz/0ch/??... [libaf] Adding filter lavcresample [libaf] Removing filter lavcresample [libaf] Adding filter dummy [dummy] Was reinitialized: 48000Hz/6ch/s16le [libaf] Adding filter format [format] Changing sample format from little-endian 16-bit signed int to big-endian 8-bit signed int [dummy] Was reinitialized: 48000Hz/6ch/s16le [format] Changing sample format from little-endian 16-bit signed int to big-endian 8-bit signed int [ac3 @ 0x1009e5bc0]Specified sample_fmt is not supported. Couldn't open codec ac3, br=384. Any help whoul dbe appreciated. Thanks Alexandre
On 1 Mar 2011, at 04:53, Alexandre Roberto Zia <alexandre@zia.com.br> wrote:
I'm having a problem with mencoder latest svn version:
ffmpeg can produce an AC3 audio file: ffmpeg -y -i video.mkv -acodec ac3 -ac 6 -ab 384k -ar 48000 -f ac3 audioout
but mencoder not: mencoder" -ss 0 video.mkv -channels 6 -ovc copy -of rawaudio -mc 0 -noskip -oac lavc -lavcopts acodec=ac3:abitrate=384 -af lavcresample=48000 -srate 48000 -o audioout -v
I suspect this has to do with the AC3 encoder now requiring float input, I suspect ae_lavc cannot yet pass through float to the encoder. Shouldn't be that hard to fix (since you're posting to -dev-eng and not to -users I assume you'd like to try fixing it yourself).
Hi, Thanks for the information. I will try to fix by myself but I`m new to mencoder / ffmpeg so there are lots of things I need to learn first. On Mar 1, 2011, at 13:50 , Reimar Döffinger wrote:
On 1 Mar 2011, at 04:53, Alexandre Roberto Zia <alexandre@zia.com.br> wrote:
I'm having a problem with mencoder latest svn version:
ffmpeg can produce an AC3 audio file: ffmpeg -y -i video.mkv -acodec ac3 -ac 6 -ab 384k -ar 48000 -f ac3 audioout
but mencoder not: mencoder" -ss 0 video.mkv -channels 6 -ovc copy -of rawaudio -mc 0 -noskip -oac lavc -lavcopts acodec=ac3:abitrate=384 -af lavcresample=48000 -srate 48000 -o audioout -v
I suspect this has to do with the AC3 encoder now requiring float input, I suspect ae_lavc cannot yet pass through float to the encoder. Shouldn't be that hard to fix (since you're posting to -dev-eng and not to -users I assume you'd like to try fixing it yourself). _______________________________________________ MPlayer-dev-eng mailing list MPlayer-dev-eng@mplayerhq.hu https://lists.mplayerhq.hu/mailman/listinfo/mplayer-dev-eng
I also added a return error code into ffmpeg: libavcodec/utils.c Without this return error code mencoder freezes. This way, at least mencoder dont hang forever. Alexandre diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 7fba5c6..f83ea64 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -557,6 +557,7 @@ int attribute_align_arg avcodec_open(AVCodecContext *avctx, AVCodec *codec) break; if (avctx->codec->sample_fmts[i] == AV_SAMPLE_FMT_NONE) { av_log(avctx, AV_LOG_ERROR, "Specified sample_fmt is not supported.\n"); + ret=-1; goto free_and_end; } } On Mar 1, 2011, at 21:20 , Alexandre Roberto Zia wrote:
Hi,
Thanks for the information. I will try to fix by myself but I`m new to mencoder / ffmpeg so there are lots of things I need to learn first.
On Mar 1, 2011, at 13:50 , Reimar Döffinger wrote:
On 1 Mar 2011, at 04:53, Alexandre Roberto Zia <alexandre@zia.com.br> wrote:
I'm having a problem with mencoder latest svn version:
ffmpeg can produce an AC3 audio file: ffmpeg -y -i video.mkv -acodec ac3 -ac 6 -ab 384k -ar 48000 -f ac3 audioout
but mencoder not: mencoder" -ss 0 video.mkv -channels 6 -ovc copy -of rawaudio -mc 0 -noskip -oac lavc -lavcopts acodec=ac3:abitrate=384 -af lavcresample=48000 -srate 48000 -o audioout -v
I suspect this has to do with the AC3 encoder now requiring float input, I suspect ae_lavc cannot yet pass through float to the encoder. Shouldn't be that hard to fix (since you're posting to -dev-eng and not to -users I assume you'd like to try fixing it yourself). _______________________________________________ MPlayer-dev-eng mailing list MPlayer-dev-eng@mplayerhq.hu https://lists.mplayerhq.hu/mailman/listinfo/mplayer-dev-eng
On Tue, Mar 01, 2011 at 09:20:07PM -0300, Alexandre Roberto Zia wrote:
Thanks for the information. I will try to fix by myself but I`m new to mencoder / ffmpeg so there are lots of things I need to learn first.
If you update to latest SVN you'll see it at least does something. The quality is horrible for me though, I guess some part in mencoder still assumes audio is always 16-bit.
participants (2)
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Alexandre Roberto Zia -
Reimar Döffinger