If so then I have no problem landing the patch (except its nits) with the documentation changes.

I posted more graphs in the discussion thread of the personal listening test at 96kbps. Also, I think it's beneficial for the end users to set the -q:a value and typically gets a file with the bitrate around the set value. The same AAC output decoded by older FFmpeg is also OK. 72585, 76735 was OK, but the 76851, 76877, 76976, 77208 suffer the same problem. Whitenoise.flac without the sound of right channel. ;). The slowness can be fixed later, encoding quality I believe being more important than speed, if needed. ffmpeg_r62950_v8f -i %i -c:a aac -strict experimental -b:a 128k -aac_coder fast %o The only serious artifact I've heard so far is whitenoise.flac at 8, 16, 24, 32kbps and 192kbps.

your coworkers to find and share information.

Another speech sample reached 216kbps.

Anyway, Kamedo2: I pushed some PNS patches yesterday which should have fixed the drop in quality.

Huh, that's odd. The container has its own overhead. However, whitenoise 384 gives me an error, seems 384kbps is too much for mono. That could have interfered. But signals like Mahler, castanets and harpishcords tended to expose the bugs at lower bit rates. -q:a sets the global_quality parameter, which is specified to have a somewhat standardized interpretation (1.0 = 100%, what 100% means is what some other codec means by it, can't remember which OTOMH).

29.92 tbr, 1k tbn, 2k tbc Recommended cutoff frequency for FFmpeg AAC. Exactly. See /usr/share/doc/ffmpeg/README.Debian.gz. [closed].

Yes, I was in the middle of tweaking rdlambda scale for VBR (which is what gives the tonality boost). No, the cutoff moves up and down, but the LP remains on.

Or am I missing something? If you upload the patch that prevents the speech bitrate bloat, I'll start testing. The bitrate there is like a lower bound (aim for 256k, spend more if needed). ​http://www.rarewares.org/test_samples/, A sound that degrades on VBR. I used tx->frame_bits = FFMIN(2560, and psymodel.h line 32: This is better on mono, surround, and on very low bitrates(such as 32kbps stereo).

Well, it is based on it, but it sounded very different to me, as it has important bugfixes. I resumed the ABC/HR test, and I've done 13 samples out of 20.

I have tested -b:a 128k -ar 44100, -b:a 192k -ar 32000, -b:a 320k -ar 48000, -q:a 1 -ar 44100, -q:a 2 -ar 48000, without additional -aac_pns enable nor -aac_is enable settings.

comment:185 was tested, along with qaac(Apple AAC). Bitrate-wise, they're similar. I notice that in -q:a 0.2 and -q:a 0.4, the lower freq is in trouble. something like that) which is very similar but better (avoids adding holes or picking a codebook that cannot encode the coefficients, for instance). You may also want to look at ticket #2706. Normally, I'd apply compression on the IMDCT stage, but since that's on the decoder side, I'll probably have to find a clever way to predict clipping on the encoder and compensate. Can you attach the short_block_test_2.flac? This one also works on VBR. If it doesn't re produce with mpeg2_aac_low, its likely related to PNS, as thats the only feature that gets turned off over aac_low. ​http://www.ffmpeg.org/releases/ffmpeg-1.2.1.tar.bz2. 3x of the "standard" bitrate of the q or something. Give it a test. Does IE7 have a “developer mode” or plugin like Firefox/Chrome/Safari? I'm not sure.

It was sometimes claimed that the wma encoders produce abysmal quality, so your comment on them (possibly with higher bitrates) would also be welcome.

Cool paper. -q:a 0.3 vs -q:a 0.7 vs -q:a 0.7 vs -vbr 3 vs --tvbr 63 vs -V5

If we blocked patches because something could be done even better, then the only acceptable patch series would be “[PATCH 0/85042] Make FFmpeg the ultimate multimedia software”. This paper, fig. 8.1.1 Options b. Considering it's already used in professional broadcasting (with aac_pns 0 since that's what causes the instability) I say it's stable.

Which repo are they referencing? I'm going to upload a combined and rebased patch (v8 indeed doesn't apply cleanly on git head). Because the sine isn't represented on the mdct by a single bin, but rather a ripple pattern that has to be accurately encoded, or the resulting sine fluctuates in amplitude (hence the clipping). Crashes on the aac_low profile are the main priority right now.

It's a matter with tonal band priorization that in VBR doesn't really work as intended. Hopefully soon there'll be an Opus encoder to keep Kamedo2 busy :). The diff of the ItCouldBeSweet?, between the original and FDK-AAC encode, 128kbps. White noise, encoded by native aac encoder at 256kbps.

The track plays just fine. So I think we'll catch FDK with the dynamic coder (which can also do the M/S part, so it'll fix both with one shot). By Edgewall Software. Convert an audio file to AAC in an M4A (MP4) container: From a video file, convert only the audio stream: Convert the video with libx264, and mix down audio to two channels: This is a pair of AAC profiles tailored for low bit rates (version 1 and version 2).

Because I've noticed ffmpeg -i file tends to give bogus rates when used on VBR-encoded files (not even average). ), I don’t remember if the medibuntu package of mencoder includes x264 vid encoding, since I build my own from git x264 and svn mplayer sources.

I've got it. So... latest patch attached. faac will probably be scrapped, fast will have to be rewritten, and anmr is a big question mark at present.

Keeping all things equal, lowering the cutoff actually increases bitrate, if a fixed RD is forced. Stream #0.0: Video: mpeg4, yuv420p, 320×240 [PAR 1:1 DAR 4:3], Since you never know what SPL will the sample be playing at, I matched the masking curve's lowest point to 16-bit quantization noise. 3.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'ffmpeg_aacvbr_pulse1.wav': Metadata: encoder : Coderium SoundEngine 4.59 Duration: 00:00:12.12, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16 , 1411 kb/s [aac @ 030cbf00] Too many bits per frame requested Output #0, mp4, to 'ffmpeg_aacvbr_pulse1.mp4': …
But I'm getting convinced this is the way for VBR... and maybe for ABR too. In any case, I made CBR also use the same scalefactor-band-based LP filtering to remove the need for the butterworth that didn't save many bits anyway, and now it responds to the -cutoff argument, so if you don't like the default cutoff you can override yourself. The v9 anmr acts fine on most samples, but vulnerable to sine waves. Billie Holiday : I'm A Fool To Want You (trimmed to 20sec, first and last), Did the output (aac) files sound better with the (original!) I've got a request of testing libfaac, mp2, and eac3, but I'm running out of the "slot". q=2-31, 200 kb/s, 90k tbn, 29.92 tbc

I think I know how to fix it (and it might indeed fix some other stuff). I can't wait to test! Craptastic. Do you have any idea? I just need to clean it up a bit. The v8g(the triangle plots) is about 2x faster than the v9. Sorry, urgent personal issues prevented me from reaching my self-imposed deadline. I have encoded over 200 GB of diverse sounds on diverse settings without apparent problems. Detailed information about the FDK AAC library (not FFmpeg specific) can be found at ​HydrogenAudio Knowledgebase: Fraunhofer FDK AAC. Can ffmpeg convert audio to raw PCM? ABR with -aac_coder fast So go install x264, mplayer/mencoder, and Nero’s AAC encoder. I've been sick lately so no progress, but soon I'll re-engage and I'll priorize sending simple bugfixes to the ML first.
I'm using current git head 54813 + aac-improvements-wip-v2-rclookahead.2.patch + aacpsy.c Line 308. 1.2 just happens to be the point at which the increased quantization floor starts zeroing out all components. The patch is getting huge, and the more you wait, the less relevant the review on ffmpeg-devel will be.

But instead of using 13, I replaced it by sizeof which is easier to maintain.

Well, v6 is almost ready.

filesize[Byte]*8/Sample_length[Sec], But be careful of very short files, it can be bogus too. The speech sample 26.French was encoded in 257kbps, more than twice bitrate than the average bitrate of large set of diverse CD sounds.


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