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23rd November 2008, 15:21 | #7101 | Link | |
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23rd November 2008, 18:12 | #7102 | Link | |
Spielberger
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Can you correct delay calculation in the next build?
Closed GOPs at the beginning of a stream are misinterpreted in eac3to. eac3to calculates delay from first I-frame, but this I-frame is third in presentation order, so -66 ms for NTSC and -80 ms for PAL is calculated. Quote:
eac3to should calculate either from GOP header or PTS from b-picture with temporal_reference = 0 Same in h264 files. My Sony AVCHD Cam e.g. creates h264 streams with 2 b-frames at the beginnung of the stream. |
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23rd November 2008, 18:22 | #7103 | Link | |
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So ArcSoft will lower all channels with 3dB during decoding? I will try to find where I can send the ArcSoft guys a sample. with logdts switch: F:\>eac3to sample.m2ts 3: e:\sample.flac -logdts + DTS-Core - frameSize 2012 - DTS-ES - - channelNo 5 - lfe 1 - channelDescr 5.1 - samplingRate 48000 - bitDepth 24 - bitrate 1536000 - samplesPerFrame 512 - copyHistory 1 + DTS-HD - fullSize 84 - headerSize 32 - refClockCode 1/48000 - frameDurationCode 1 - activeMasks [1], [[1]] + Asset [0] - fullSize 52 - headerSize 14 - corePackets Core - extSubStrPackets XLL - bitResolution 24 - maxSampleRate 48000 - totalNumChannels 8 - activeSpeakers C L R Ls Rs LFE Lsr Rsr ($4f) M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:26 1: h264/AVC, 1080p24 /1.001 (16:9) 2: AC3, 5.1 channels, 640kbps, 48khz 3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz (core: DTS, 5.1 channels, 24 bits, 1536kbps, 48khz) 4: Subtitle (PGS) 5: Subtitle (PGS) 6: Subtitle (PGS) 7: Subtitle (PGS) CAUTION: Decoding this track with ArcSoft results in low volume. |
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23rd November 2008, 18:47 | #7104 | Link |
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Look two comments above yours. I already replied to you there. Delay should be correct with the next build. At least the current work in progress sources don't report any audio delay for both of your samples...
With this specific track: Yes. With most other 7.1 tracks: No. |
23rd November 2008, 19:45 | #7105 | Link | |
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23rd November 2008, 23:32 | #7106 | Link |
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eac3to v2.78 released
http://madshi.net/eac3to.zip Code:
* fixed: h264 interlaced muxing to MKV could result in too long runtime * fixed: transcoding DTS-HD/E-AC3 core sometimes failed to work correctly * improved TS/m2ts audio delay detection * added filter to remove spikes when fixing gaps/overlaps in RAW/PCM audio * each eac3to instance has its own log file now * playlist output now also works with "-log" option * default bitrate for mono & stereo AC3 encodes lowered to 448kbps * default bitrate for mono & stereo DTS encodes lowered to 768kbps * it should be possible to handle TsSplitter splitted TS files via "+" now |
24th November 2008, 01:31 | #7108 | Link | |
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Chumbo |
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24th November 2008, 02:28 | #7109 | Link |
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thanks! will test those interlaced h264 movies during the week!
regarding that improved m2ts audio delay correction, have there been any problems (maybe also such which werent indicated by the log?) and it was a fix, or just more like getting the already fine working detection (for most movies) more towards perfection? |
24th November 2008, 08:18 | #7110 | Link |
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Check out bigotti5's last two posts in this thread. Delay correction itself worked just fine, but delay detection was off by 2 video frames in two samples he provided. I think this problem only occurred with some movies, though, not with all. I think most Blu-Ray movies shouldn't have this problem.
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24th November 2008, 08:35 | #7111 | Link | |
Spielberger
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Thx - works in ts-streams.
------------ Delay correction in VOB files regarding closed GOP: - Closed GOP Video - 2 leading B-frames - Audio AC3 - no delay eac3to reports: Quote:
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24th November 2008, 10:53 | #7113 | Link | |
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- vobu start presentation time: X - PTS first audio: X + 10ms - PTS first video: X + 5ms Now if you demux audio and video, audio should be delayed by 5ms and not by 10ms, or am I wrong? I can't "delay" video, so I have to delay audio by the difference between first audio and video PTS, no? Using CBR for AAC encoding is not really a good thing for quality. VBR encoding gives better quality per average bitrate. So why would you want to use CBR? |
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24th November 2008, 11:24 | #7114 | Link | |
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This "guy" is not me, am I? I'm still working on my resampler's test, and it could take a while to finish it, I have to do a few things first. But my first test gave some interesting results, I only don't know if the results are valid enough... |
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24th November 2008, 11:55 | #7115 | Link | |
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http://src.infinitewave.ca/ He gave me some feedback on my early SSRC implementation, based on which I tweaked the SSRC parameters a bit. But I'm still interesting in your comparison, too. Would be nice if you could use the latest eac3to version, because of the tweaked SSRC parameters... As far as I understand the technical comparison website above, SSRC is a rather steep resampling filter with good results, but with "normal" ringing. r8brain filters out quite a lot of the high frequencies, but on the positive side r8brain has very reduced ringing (see pulse graph). So both filters have their advantages and disadvantages, technically. |
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24th November 2008, 12:11 | #7116 | Link | |
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24th November 2008, 12:42 | #7118 | Link |
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Because sometimes I need to hit a certain filesize. I have no problem with quality based encoding as the default, but adding the option isn't much work and it would save me some typing compared to manually piping the stdout from eac3to to the nero encoder.
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24th November 2008, 12:58 | #7119 | Link | ||
Spielberger
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But look in "Philips DVD Verifier" documentation page 127 Quote:
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24th November 2008, 13:20 | #7120 | Link | ||
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