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12th April 2012, 14:32 | #11601 | Link |
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Thank you for your help! So I made WAV files with eac3to and want to convert them to WavPack, but the Encoder says: "can't handle .WAV files larger than 4 GB (non-standard)!" - sadly booth streams are over 4 GB. Also Monkey's audio can't encode them: "Error: 1002".
Export to AC3 with ea3to works fine but I absolutely want a lossless codec (it feels better ). In my desperation I even tried to merge the movie MKV with the WAVE files but MKVmerge GUI gives the error "87". Both WAVE streams seams to be a little bit strange: Non of my audio players can play more than a few minutes (even they display only a few minutes of playtime) and also Audacity can only show me a few seconds! The link from @Sparktank is very confusing for me. Maybe I try it later. Downmixing the stream would be the last option but I would be very pleased if it's possible to preserve the original channels... |
12th April 2012, 15:18 | #11603 | Link | |
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12th April 2012, 16:04 | #11604 | Link |
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@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
You are right, the fact that a software is OSS doesn't mean that there are other developers capable with the task. But look at ffmpeg/libav, x264, VLC and mplayer in the media field. It's not soo bad. And consider the chance @madshi, eac3to's author can't work on freeware projects anymore or dies. Then the community would be forced to rewrite the tool from scratch. OSS is freedom and opportunity, not constraint. |
12th April 2012, 22:35 | #11606 | Link |
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eac3to bug when decoding DTS-ES 6.1 using libav
Hi, I am decoding a DTS-ES 6.1 soundtrack to wav format. If I use the Sonic decoder then everything works correctly. However if I use libav, then I end up with a wav file with a corrupt channel mask. It looks like eac3to is not taking into account libav's lack of back channel decoding:
Code:
eac3to j:\backup 1) 4: c:\temp\orig.wav -libav M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 1:38:52, 24p /1.001 1: Chapters, 20 chapters 2: MPEG2, 1080p24 /1.001 (16:9) 3: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB 4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB 5: Subtitle (PGS), English 6: Subtitle (PGS), French 7: Subtitle (PGS), Spanish 8: Subtitle (PGS), English a04 The libav DTS decoder doesn't decode the back channels. a04 Extracting audio track number 4... a04 Removing DTS dialog normalization... a04 Removing XCh extension... a04 Decoding with libav/ffmpeg... a04 Reducing depth from 64 to 24 bits... a04 Writing WAV... a04 Creating file "c:\temp\orig.wav"... I thought using -down6 might force eac3to to work properly, but this ended up with the same problem. So I think the solution is for madshi to update eac3to to write the correct channel mask when libav decodes 6.1 (and probably 7.1) DTS files. |
20th April 2012, 00:31 | #11610 | Link |
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Is it possible to convert TrueHD to DTS-HDMA (TrueHD > WAVS > DTS HD Encoder) and keep the same sound for the TrueHD tracks that apply DRC to improve the sound? (at least I think it's the DRC that's doing this ?).
This problem arises with the Transformers 3 TrueHD 7.1 track (and people tell me Iron Man is probably the only other movie doing the same). The converted DTS-HDMA track does not sound the same as the original TrueHD. Some sounds at different moments in the movie have more presence with the original TrueHD track. It makes the sound better, more dynamic. I think this is caused by the DRC (?) that's applied differently on the different channels at different moments. The TrueHD track sounds better with PowerDVD 12 compared to the converted DTS-HDMA, but with MPC I don't seem to hear a difference (this leads me to think that MPC doesn't apply the effects and so it doesn't handle the TrueHD metadata correctly (?). PCM and DTS-HDMA can't modify the sound at playback time in the way TrueHD does. How can I save the wavs with the full TrueHD sound experience? To summarize, I would like to convert TrueHD to DTS-HDMA and have it sound as good on hardware receivers and PowerDVD (now with eac3to the audio source quality is the same, but many sounds are lacking presence in different parts of the movie). (This is the first movie I am having this problem with. It seems the Transformers 3 sound engineers wanted to TrueHD track to be reproduced with improvements applied to the Lossless track it contains - and the whole experience is indeed more enjoyable to my ears with these improvements). PS: sorry mods, I have opened this thread before, but I think here is the right place as it concerns eac3to Last edited by Bigmango; 20th April 2012 at 00:52. |
20th April 2012, 10:11 | #11611 | Link |
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Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month. eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.
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21st April 2012, 14:40 | #11612 | Link | |
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Quote:
Regarding eac3to: Can eac3to be used with another decoder that can save the "complete" TrueHD to wavs? It seems the open source decoders don't fully support TrueHD as they only extract the lossless track without the effects that enhance it by giving more presence to specific sounds at specific times (most movies only play the lossless track as is, so only a handful of movies seem to be concerned by this issue). Thanks. |
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26th April 2012, 11:29 | #11614 | Link |
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I'm converting 5.1 AC3 from DVD (25.000 -> 23.976), Is it normal that eac3to remaps channels?
Code:
eac3to f:\al.ac3 g:\al.ac3 -slowdown AC3, 5.1 channels, 1:26:43, 384kbps, 48kHz, dialnorm: -27dB The Nero decoder doesn't seem to work, will use libav instead. Removing AC3 dialog normalization... Decoding with libav/ffmpeg... Remapping channels... Changing FPS from 25.000 to 23.976... Encoding AC3 <640kbps> with libAften... Creating file "g:\al.ac3"... eac3to processing took 3 minutes, 42 seconds. Done.
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5th May 2012, 22:22 | #11616 | Link |
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Hello, I am having the following problem:
Code:
eac3to v3.24 command line: "c:\Program Files\Utils\eac3to\eac3to.exe" "track01.dts" "track01.wav" -libav -simple ------------------------------------------------------------------------------ VOB, 1 audio track, 0:08:57 1: DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz Track 1 is used for destination file "track01.wav". [a01] Extracting audio track number 1... [a01] Decoding with libav/ffmpeg... [a01] Reducing depth from 64 to 24 bits... [a01] Writing WAV... [a01] Creating file "track01.wav"... [a01] Clipping detected, a 2nd pass will be necessary. <WARNING> [a01] Starting 2nd pass... [a01] Extracting audio track number 1... [a01] Decoding with libav/ffmpeg... [a01] Reducing depth from 64 to 24 bits... [a01] Writing WAV... [a01] Applying -0.73dB gain... [a01] Creating file "track01.wav"... eac3to processing took 1 minute, 27 seconds. Done. But look at the file sizes: Code:
>>dir track01*.* Volume in drive K is Music Volume Serial Number is 884C-4DCD Directory of K:\DTS 29/04/2012 14:09 105,261,056 track01.dts 05/05/2012 23:54 987 track01 - Log.txt 05/05/2012 23:54 463,804,460 track01.wav 3 File(s) 569,066,503 bytes 0 Dir(s) 88,401,051,648 bytes free I don't know, other WAV DTS files I have are much smaller (given the average MB/min of audio, this one is 8:56 min) Is there any command line switch to keep the WAV size smaller? |
6th May 2012, 02:49 | #11617 | Link | ||
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Quote:
6 channels x 24 bits x 48KHz = 6912 kb/s 6912/1510 = 4,57 Then, yes, is correct. Quote:
6 channels x 16 bits x 48KHz = 4608 kb/s 4608/1510 = 3,05
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7th May 2012, 09:53 | #11618 | Link | |
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Quote:
I am dumb regarding the technical details. I think the DTS file has the same information as the final WAV, that's why the increase in size seemed too much. But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it? EDIT: I just looked in another song I have in DTS 5.1 WAV format (in the AC3 filter properties while it was playing) and it showed this: Code:
Decoder: Stream format: DTS 3/2.1 (5.1) 44100Hz Bitstream type: 14bit low endian Frame size: free format Samples: 1024 Bitrate: unknown SPDIF stream type: 0xc Frame interval: 4096 Actual bitrate: 1411kbps DTS speakers: 3/2.1 (5.1) sample rate: 44100Hz bitrate: 1411kbps stream: 14bit LE frame size: 3584 bytes nsamples: 1024 amode: 9 No CRC It is 13:40 min song and the file only takes 141MB Something is not fitting, according to my understanding On the other hand when I play the file created by eac3to, AC3 filter shows this: Code:
Input format: PCM24 3/2.1 (5.1) 48000 User format: PCM16 - 0 Output format: PCM16 3/2.1 (5.1) 48000 Last edited by ilomambo; 7th May 2012 at 10:13. |
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7th May 2012, 11:33 | #11619 | Link | |
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DTS is a compressed audio format, while WAV is not. Hence the reason that the WAV file is larger after you uncompressed the DTS file. |
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7th May 2012, 13:45 | #11620 | Link |
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But yes, .WAV is a RIFF-based container, and may contain compressed audio. Regarding DTS-in-WAV especifically, there are two types, 1) normal, without SPDIF-padding, and with a .dca TwoCC, and 2) hacky, with SPDIF-padding, disguised as stereo PCM @ 32 / 44.1 / 48 kHz.
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