On Wed, Nov 16, 2016 at 8:09 PM, Michael Niedermayer
wrote:
>> Sorry for the delay, it turned out to be more complex than that.
>>
>> There were a few potential violations that I had already identified in
>> a WIP patch but they did not apply to the fuzzed sample. That
LGTM.
However,
On Sat, Oct 8, 2016 at 12:20 PM, Rostislav Pehlivanov
wrote:
> +/**
> + * linear congruential pseudorandom number generator
> + *
> + * @param previous_valpointer to the current state of the generator
> + *
> + * @return Returns a 32-bit pseudorandom
On Mon, Oct 3, 2016 at 3:53 PM, Rostislav Pehlivanov
wrote:
> Hopefully whoever wants to have support for crazy formats can help.
> The table in aacenc.h (temporary position) tells the encoder what
> to put in the bitstream and how to encode. Problem is, the specifications
>
On Sat, Sep 10, 2016 at 3:37 AM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Thu, Aug 25, 2016 at 8:57 AM, Rostislav Pehlivanov
> <atomnu...@gmail.com> wrote:
>>> 64ed96a710787ba5d0666746a8562e7d.dee
>>>
>>> Found-by: Mateusz "j00ru"
On Thu, Aug 25, 2016 at 8:57 AM, Rostislav Pehlivanov
wrote:
>> 64ed96a710787ba5d0666746a8562e7d.dee
>>
>> Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
>> Signed-off-by: Michael Niedermayer
>> ---
>> libavcodec/aaccoder.c | 8 +++-
>> 1
On Mon, May 16, 2016 at 4:10 PM, Michael Niedermayer
wrote:
> yes,
>
> with the patch fate fails:
>
> Test aac-pred-encode failed. Look at tests/data/fate/aac-pred-encode.err for
> details.
> make: *** [fate-aac-pred-encode] Error 134
>
>
>>
>> A threshold of 0 would in
On Mon, May 16, 2016 at 12:26 PM, Kieran Kunhya wrote:
>> Testcase is fate-aac-pred-encode
>>
>> Signed-off-by: Michael Niedermayer
>> ---
>> libavcodec/aacenc_is.c |3 +++
>> 1 file changed, 3 insertions(+)
>>
>> diff --git a/libavcodec/aacenc_is.c
On Sun, Apr 10, 2016 at 4:13 PM, Michael Niedermayer
wrote:
> that said, to be blunt, make your encoder be capable to encode as fast
> and as good quality as libfaac and after that remove libfaac support
> if you want.
I could make a small patch that could do just that,
On Mon, Apr 4, 2016 at 9:00 AM, Claudio Freire <klaussfre...@gmail.com> wrote:
>> http://fate.ffmpeg.org/report.cgi?time=20160403222945=x86_64-archlinux-gcc-valgrindundef
>>
>> No memleaks, but a handful of "Conditional jump or move depends on
>> uninitiali
On Sun, Apr 3, 2016 at 10:41 PM, James Almer <jamr...@gmail.com> wrote:
> On 4/3/2016 3:29 PM, Claudio Freire wrote:
>> On Sun, Apr 3, 2016 at 3:05 PM, Claudio Freire <klaussfre...@gmail.com>
>> wrote:
>>> On Sat, Apr 2, 2016 at 4:00 PM, Michael Niedermaye
On Sun, Apr 3, 2016 at 3:05 PM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Sat, Apr 2, 2016 at 4:00 PM, Michael Niedermayer
> <mich...@niedermayer.cc> wrote:
>> Ideally that should be fixed before its pushed
>>
>> Signed-off-by: Michael
On Sat, Apr 2, 2016 at 4:00 PM, Michael Niedermayer
wrote:
> Ideally that should be fixed before its pushed
>
> Signed-off-by: Michael Niedermayer
> ---
> tests/fate/aac.mak | 11 +++
> 1 file changed, 11 insertions(+)
>
> diff --git
On Wed, Mar 30, 2016 at 10:15 AM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Wed, Mar 30, 2016 at 8:33 AM, Michael Niedermayer
> <mich...@niedermayer.cc> wrote:
>> On Wed, Mar 30, 2016 at 02:04:20AM -0300, Claudio Freire wrote:
>>> On Wed, Mar 30
On Wed, Mar 30, 2016 at 8:33 AM, Michael Niedermayer
<mich...@niedermayer.cc> wrote:
> On Wed, Mar 30, 2016 at 02:04:20AM -0300, Claudio Freire wrote:
>> On Wed, Mar 30, 2016 at 1:18 AM, Claudio Freire <klaussfre...@gmail.com>
>> wrote:
>> > On Tue, Mar 29, 2
On Wed, Mar 30, 2016 at 1:18 AM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Tue, Mar 29, 2016 at 10:51 PM, Michael Niedermayer
> <mich...@niedermayer.cc> wrote:
>> This is a hotfix and not a real fix of the underlaying bug
>> The underlaying bug is ATM
On Tue, Mar 29, 2016 at 10:51 PM, Michael Niedermayer
wrote:
> This is a hotfix and not a real fix of the underlaying bug
> The underlaying bug is ATM not fully understood
>
> iam not sure if we should apply this or not
>
> Signed-off-by: Michael Niedermayer
On Sun, Mar 13, 2016 at 10:30 PM, Ganesh Ajjanagadde wrote:
> /**
> * Calculate rate distortion cost for quantizing with given codebook
> @@ -105,7 +106,7 @@ static av_always_inline float
> quantize_and_encode_band_cost_template(
> curbits += 21;
>
On Thu, Mar 10, 2016 at 8:34 PM, Michael Niedermayer
wrote:
> Hi all
>
> if you fix speed regressions of the AAC encoder please backport them
> also to release/3.0
>
> Thanks
Even if that implies much bigger changes? (like the one I commented
about aacpsy allocation)
On Wed, Mar 9, 2016 at 3:52 PM, Reimar Döffinger
<reimar.doeffin...@gmx.de> wrote:
> On Wed, Mar 09, 2016 at 02:20:35PM -0300, Claudio Freire wrote:
>> On Mon, Feb 29, 2016 at 11:30 PM, Ganesh Ajjanagadde <gajja...@gmail.com>
>> wrote:
>> > On Mon, Fe
On Mon, Feb 29, 2016 at 11:30 PM, Ganesh Ajjanagadde wrote:
> On Mon, Feb 22, 2016 at 11:34 PM, Andrey Utkin
> wrote:
>> Hi!
>> I am aware of news that AAC encoder got stable status recently.
>>
>> But you could find this interesting. We've got an
On Fri, Jan 29, 2016 at 1:08 AM, Timothy Gu wrote:
> On Sun, Jan 24, 2016 at 04:33:31PM +, Kieran Kunhya wrote:
>> The internal encoder is superior to libvo-aacenc.
>> ---
>> configure | 6 --
>> doc/encoders.texi | 25 --
>>
On Wed, Jan 20, 2016 at 11:05 AM, Michael Niedermayer wrote:
> From: Michael Niedermayer
>
> This is needed as near infinite values on the input side result in only some
> output to be non finite.
> Also it may still be insufficient if subsequent
On Thu, Jan 14, 2016 at 7:57 PM, Ganesh Ajjanagadde
wrote:
> Signed-off-by: Ganesh Ajjanagadde
> ---
> libavcodec/aacenc.c | 17 +
> 1 file changed, 9 insertions(+), 8 deletions(-)
>
> diff --git a/libavcodec/aacenc.c
On Mon, Jan 11, 2016 at 7:23 PM, Ganesh Ajjanagadde
wrote:
> This is quite an accurate approximation; testing shows ~ 2ulp error in
> the floating point result. Tested with FATE.
>
> Alternatively, if one wants "full accuracy", one can use powf, or sqrt
> instead of sqrtf.
On Tue, Dec 29, 2015 at 1:12 PM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Tue, Dec 29, 2015 at 6:18 AM, Rostislav Pehlivanov
> <atomnu...@gmail.com> wrote:
>> Wouldn't it be simpler to just check if the maximum codebook was 0 after
>> calculating cost1 and
On Tue, Dec 29, 2015 at 6:18 AM, Rostislav Pehlivanov
<atomnu...@gmail.com> wrote:
> Wouldn't it be simpler to just check if the maximum codebook was 0 after
> calculating cost1 and skipping the rest of the code in the loop?
>
> On 29 December 2015 at 08:23, Claudio Freire <k
creating/removing zeroes, perhaps an
overly conservative approach, but a safe one. More permissive
and sophisticated approaches may be attempted in the future.
Attached
From dbda62b2a59054a76e317850a4a0a7037ca3cc02 Mon Sep 17 00:00:00 2001
From: Claudio Freire <klaussfre...@gmail.com>
Date: T
On Fri, Dec 18, 2015 at 10:59 AM, Rostislav Pehlivanov
wrote:
> The type of last_frame_pb_count was chosen to be an int since overflow
> is impossible (the spec says the maximum bits per frame is 6144 per
> channel and the encoder checks for that).
LGTM
On Wed, Dec 9, 2015 at 11:15 AM, Carl Eugen Hoyos wrote:
> On Sunday 06 December 2015 01:31:17 am Carl Eugen Hoyos wrote:
>> Hi!
>>
>> I prefer attached patch over removing the encoder immediately.
>
> No opinions?
>
> Carl Eugen
LGTM, but I don't use the libvo-aac, so... :-p
On Sun, Dec 6, 2015 at 6:36 PM, Andreas Cadhalpun
wrote:
> The other is a regression since 01ecb71, so I hope you know how to fix that.
> In search_for_pns in libavcodec/aaccoder.c:
> for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
> [...]
On Wed, Dec 9, 2015 at 4:42 PM, Andreas Cadhalpun
wrote:
>>> [...]
>>> for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
>>> [...]
>>> }
>
> Now we are after the w2-loop and thus:
> w2 = sce->ics.group_len[w] = 2
>
Ah, I see, it's outside
On Wed, Dec 9, 2015 at 5:29 PM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Wed, Dec 9, 2015 at 4:42 PM, Andreas Cadhalpun
> <andreas.cadhal...@googlemail.com> wrote:
>>>> [...]
>>>> for (w2 = 0; w2 < sce->ics.group_len[w]; w2
On Fri, Dec 4, 2015 at 2:23 PM, Andreas Cadhalpun
wrote:
> If minq is negative, the range of sf_idx can be larger than
> SCALE_MAX_DIFF allows, causing assertion failures later in
> encode_scale_factors.
>
> Signed-off-by: Andreas Cadhalpun
On Fri, Dec 4, 2015 at 9:21 PM, Andreas Cadhalpun
<andreas.cadhal...@googlemail.com> wrote:
> On 04.12.2015 23:49, Claudio Freire wrote:
>> On Fri, Dec 4, 2015 at 2:23 PM, Andreas Cadhalpun
>> <andreas.cadhal...@googlemail.com> wrote:
>>> If minq is negative, th
On Fri, Dec 4, 2015 at 9:52 PM, Andreas Cadhalpun
wrote:
> Pushed.
>
>> Do you have the problematic input at hand? If so, send it privately.
>
> Sure, I'll send you a sample.
>
>> If I find a better solution I may try to push that instead, or at
>> least add the
On Thu, Dec 3, 2015 at 3:01 PM, Rostislav Pehlivanov
wrote:
> Any opposition?
Not from me
___
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
So, here comes the discussion again.
This time, the AAC encoder is in good shape. It's not perfect. I have
a list of known bugs to address that still has some issues, but I'm
not really certain whether they should block the flag's removal.
The bugs will be addressed in time, but maybe the
On Wed, Dec 2, 2015 at 12:50 PM, Hendrik Leppkes <h.lepp...@gmail.com> wrote:
> On Wed, Dec 2, 2015 at 4:42 PM, Clément Bœsch <u...@pkh.me> wrote:
>> On Wed, Dec 02, 2015 at 12:37:00PM -0300, Claudio Freire wrote:
>>> So, here comes the discussion again.
>
On Wed, Dec 2, 2015 at 4:14 PM, Rostislav Pehlivanov
<atomnu...@gmail.com> wrote:
> This commit removes the experimental flag from the native AAC Encoder
> and thus makes it the default.
>
> After a lot of work, done by myself and Claudio Freire, the quality of
> this encoder
On Wed, Dec 2, 2015 at 4:47 PM, Rostislav Pehlivanov
wrote:
> This commit marks any coders beside twoloop as experimental and gives
> out a warning that some of they might be silently removed in the future.
>
> Users are highly encouraged to use the twoloop coder, which is
On Wed, Dec 2, 2015 at 12:51 PM, Ganesh Ajjanagadde <gajja...@mit.edu> wrote:
> On Wed, Dec 2, 2015 at 10:37 AM, Claudio Freire <klaussfre...@gmail.com>
> wrote:
>> So, here comes the discussion again.
>>
>> This time, the AAC encoder is in good shape. It's no
On Tue, Dec 1, 2015 at 11:12 PM, Claudio Freire <klaussfre...@gmail.com> wrote:
> On Tue, Dec 1, 2015 at 10:47 PM, Michael Niedermayer <michae...@gmx.at> wrote:
>>> libavcodec/aaccoder.c | 60 --
>>> lib
On Tue, Dec 1, 2015 at 10:47 PM, Michael Niedermayer wrote:
>> libavcodec/aaccoder.c | 60 --
>> libavcodec/aaccoder_twoloop.h | 136
>> --
>> libavcodec/aacenc.c |2
>> libavcodec/aacenc_is.c
On Mon, Nov 30, 2015 at 2:20 PM, Rostislav Pehlivanov
<atomnu...@gmail.com> wrote:
> On Mon, 2015-11-30 at 12:50 -0300, Claudio Freire wrote:
>> Also I don't see how a static var would help or even be correct here.
>> Perhaps you meant something else?
> static uint8_t
On Mon, Nov 30, 2015 at 12:27 PM, Rostislav Pehlivanov
<atomnu...@gmail.com> wrote:
> On Sun, 2015-11-29 at 16:54 -0300, Claudio Freire wrote:
>> Before pushing this, I'd like some feedback,
>> especially about
>> the implementation of point 3. I'm not sure the AAC en
On Mon, Nov 30, 2015 at 1:04 PM, Hendrik Leppkes <h.lepp...@gmail.com> wrote:
> On Mon, Nov 30, 2015 at 4:50 PM, Claudio Freire <klaussfre...@gmail.com>
> wrote:
>> On Mon, Nov 30, 2015 at 12:27 PM, Rostislav Pehlivanov
>> <atomnu...@gmail.com> wrote:
&
On Mon, Nov 30, 2015 at 1:04 PM, Hendrik Leppkes <h.lepp...@gmail.com> wrote:
> On Mon, Nov 30, 2015 at 4:50 PM, Claudio Freire <klaussfre...@gmail.com>
> wrote:
>> On Mon, Nov 30, 2015 at 12:27 PM, Rostislav Pehlivanov
>> <atomnu...@gmail.com> wrote:
&
, though it can still
use some further tweaks.
From e236ab0021eb28012ed05f46e7e520b5cbec413e Mon Sep 17 00:00:00 2001
From: Claudio Freire <klaussfre...@gmail.com>
Date: Sun, 29 Nov 2015 16:33:31 -0300
Subject: [PATCH] AAC encoder: improve SF range utilization
This patch does 4 things, all of
On Tue, Nov 10, 2015 at 1:41 AM, Timothy Gu <timothyg...@gmail.com> wrote:
> On Sun, Nov 8, 2015 at 9:28 AM Claudio Freire <klaussfre...@gmail.com>
> wrote:
>>
>> This particular piece of code is going to disappear soon, so not sure
>> it's worth applying the
On Fri, Sep 25, 2015 at 10:39 PM, James Almer <jamr...@gmail.com> wrote:
>> ffmpeg | branch: master | Claudio Freire | Fri
>> Sep 25 03:56:32 2015 -0300| [9458a62decfcaa1313b1ba69276466de536d0768] |
>> committer: Claudio Freire
>>
>> AAC encoder: t
On Tue, Sep 15, 2015 at 8:11 AM, Michael Niedermayer <michae...@gmx.at> wrote:
> On Tue, Sep 15, 2015 at 04:24:02AM -0300, Claudio Freire wrote:
>> This patch refactors the AAC coders to reuse code
>> between the MIPS port and the regular, portable C code.
>> There were
On Wed, Sep 16, 2015 at 12:30 PM, Nedeljko Babic
wrote:
Patch attached.
I thought it was worth a review.
It does include lots of copypaste.
FTR, I tested MIPS 74Kf and x86_64 with make fate-aac
>>>
>>> full fate passes on
on cross gcc 4.5.3, a slight
variation on this fate failure:
http://fate.ffmpeg.org/log.cgi?time=20150914220602=compile=mips-linux-gcc-4.3.2
- let me know if I can provide some more info on this to get it fixed
somehow.
From 3a530153a0c4ef18f5a0f936fd356e9493b5a00e Mon Sep 17 00:00:00 2001
From: Claudio
I'm also looking into it.
It's simply that it tends to get out of sync with the rest really.
I will remove some duplicated code (namely the twoloop and trellis
functions which are verbatim copies of their counterparts in
aaccoder), while somehow keeping the fact that they used the optimized
On Tue, Sep 1, 2015 at 5:54 AM, Robert Krüger wrote:
>> It's been the plan all along to negotiate the removal of the
>> experimental flag after pushing those changes discussed and heavily
>> tested in ticket #2686.
>>
>> If anything, this thread expresses support for that
On Mon, Aug 31, 2015 at 6:59 AM, Robert Krüger wrote:
> On Mon, Aug 31, 2015 at 10:44 AM, Carl Eugen Hoyos wrote:
>
>> Hi!
>>
>> I didn't test myself but iiuc, the aac encoder produces better
>> quality than all other libavcodec audio encoders and better
>>
On Sat, Aug 22, 2015 at 4:51 AM, Nicolas George geo...@nsup.org wrote:
Le quintidi 5 fructidor, an CCXXIII, Claudio Freire a écrit :
They were included in the symbol table but only as local, the
libavcodec.v
file makes sure to make everything not explicitly mentioned for export
local
On Sat, Aug 22, 2015 at 1:01 AM, Timothy Gu timothyg...@gmail.com wrote:
On Fri, Aug 21, 2015 at 10:49:01PM -0400, Ganesh Ajjanagadde wrote:
There are too many entries here for me to verify which ones are exposed,
etc.
I trust you identified them correctly.
To the best of my knowledge,
On Wed, Jul 29, 2015 at 1:44 AM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
As well as tables littered everywhere, functions were spread
out all across the encoder's files. This moves them to a single
place where they can be used by either the encoder's main files
or additional encoder
On Wed, Jul 29, 2015 at 1:44 AM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
With the new method to determine the phase, the functions
got sufficiently large to have their own file.
There are absolutely no changes from the IS patch from
this patchset, this commit simply moves it out of
On Wed, Jul 29, 2015 at 1:44 AM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit moves any tables specific to the encoder from aacenc
and aaccoder to a separate file called 'aacenctab.c/.h'.
This was done as a clean up attempt as the encoder was filled with
tables pasted in between
On Wed, Jul 29, 2015 at 1:44 AM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+if (cpe-ms_mode)
+phase = 1 - 2 * cpe-ms_mask[w*16+g];
Shouldn't it be ?
phase *= 1 - ... ;
phase is an argument, the original code would step on it, with a value
that doesn't depend on phase, so it
On Fri, Jul 31, 2015 at 9:56 PM, Michael Niedermayer
mich...@niedermayer.cc wrote:
Rostislav, Claudio
please both of you send me your public SSH keys, i think you both
should have git write access
Sent mine on a private email. Let me know if you didn't get it.
On Wed, Jul 29, 2015 at 1:44 AM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit removes a redundant argument from the functions in aaccoder.
The argument lambda was redundant as it was just a copy of s-lambda,
to which all functions have access to anyway. This cleans up the
On Wed, Jul 29, 2015 at 4:15 PM, Michael Niedermayer
mich...@niedermayer.cc wrote:
On Mon, Jul 27, 2015 at 07:28:35PM +0200, Michael Niedermayer wrote:
On Mon, Jul 20, 2015 at 11:40:54PM -0300, Claudio Freire wrote:
On Mon, Jul 20, 2015 at 11:39 PM, Claudio Freire klaussfre...@gmail.com
On Wed, Jul 22, 2015 at 6:18 AM, Nedeljko Babic
nedeljko.ba...@imgtec.com wrote:
On Mon, Jul 20, 2015 at 8:50 AM, Nedeljko Babic
nedeljko.ba...@imgtec.com wrote:
This commit moves the tables required for encoding and decoding
LTP and TNS AAC files out of the decoder's standalone tables file
and
On Mon, Jul 20, 2015 at 5:08 AM, Venelin Efremov veffremov...@gmail.com wrote:
The error message I get from the latest git head is:
[aac_latm @ 0x3a226e0] Non-byte-aligned audio-specific config is not
implemented. Update your FFmpeg version to the newest one from Git. If the
problem still
On Tue, Jul 21, 2015 at 2:50 PM, Claudio Freire klaussfre...@gmail.com wrote:
On Mon, Jul 20, 2015 at 5:08 AM, Venelin Efremov veffremov...@gmail.com
wrote:
The error message I get from the latest git head is:
[aac_latm @ 0x3a226e0] Non-byte-aligned audio-specific config is not
implemented
On Mon, Jul 20, 2015 at 4:15 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
Yep, with this patch compute_lpc_coefs() (referenced in aacdec_template.c
when applying TNS) in ltp.c prints a warning when compiling the fixed
decoder, which is fine since it changed a type from INTFLOAT to float.
On Mon, Jul 20, 2015 at 8:50 AM, Nedeljko Babic
nedeljko.ba...@imgtec.com wrote:
This commit moves the tables required for encoding and decoding
LTP and TNS AAC files out of the decoder's standalone tables file
and into the shared aactab.h, where they can be used by both the
encoder and the
On Fri, Jul 17, 2015 at 11:19 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
But even if not used for avoding I/S, it can be used to pick whether
to invert the phases, where it was clearly more stable.
In case the phase is very clearly wrong then there will be an increase in
the distortion
This will need rebasing, the fixed tablegen got in recently
On Fri, Jul 17, 2015 at 6:20 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit moves the generation of ff_aac_pow34sf_tab[] out of the
encoder and into the table generator. The original commit log for
this table in 2011
factor
2. Computes a whole-frame factor (minimum of all)
3. Applies the clip avoidance factor to the whole frame (to make it
smooth and almost linear, and avoid adding harmonic distortion).
From 57522de7c5fcdbef222c2425a4add6fa4528f0e7 Mon Sep 17 00:00:00 2001
From: Claudio Freire klaussfre
On Mon, Jul 20, 2015 at 11:39 PM, Claudio Freire klaussfre...@gmail.com wrote:
On Fri, Jul 17, 2015 at 8:42 PM, Michael Niedermayer
mich...@niedermayer.cc wrote:
If you mean a transition in time, I don't think it makes any
difference. 0.95 is a ~0.5db change in intensity, which ought
On Mon, Jul 20, 2015 at 10:05 PM, Claudio Freire klaussfre...@gmail.com wrote:
This will need rebasing, the fixed tablegen got in recently
On Fri, Jul 17, 2015 at 6:20 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit moves the generation of ff_aac_pow34sf_tab[] out
weighted error would make for
a representative test, so that will be left for a future patch.
From 9da94f02574b34025a56c225c11269802f49949b Mon Sep 17 00:00:00 2001
From: Claudio Freire klaussfre...@gmail.com
Date: Fri, 17 Jul 2015 05:47:25 -0300
Subject: [PATCH] AAC Encoder: clipping avoidance
Avoid
On Fri, Jul 17, 2015 at 7:09 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
attachment.bin:357: trailing whitespace.
attachment.bin:436: trailing whitespace.
attachment.bin:454: trailing whitespace.
Oops
You should probably configure your editor to strip that off automatically.
I don't
On Fri, Jul 17, 2015 at 10:32 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
That previous idea discriminated way too many bands for it to be actually
useful. And it would require special cases for coefficients which 'blow up'
and have an insane value.
...
What happened to the idea of
On Fri, Jul 17, 2015 at 6:20 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit adds a slightly more robust way of determining whether the phases
match or are too different for IS to be used.
---
libavcodec/aaccoder.c | 7 +--
1 file changed, 5 insertions(+), 2 deletions(-)
On Fri, Jul 17, 2015 at 8:42 PM, Michael Niedermayer
mich...@niedermayer.cc wrote:
If you mean a transition in time, I don't think it makes any
difference. 0.95 is a ~0.5db change in intensity, which ought to be
inaudible, and windowing will already take care to make the transition
smooth. And
On Thu, Jul 2, 2015 at 3:13 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit undoes commit c5d4f87e8427c0952278ec247fa8ab1e6e52 and
removes PNS band marking from the twoloop coder, which has been reimplemented
in a better way in this series of patches.
LGTM as long as the
On Thu, Jul 2, 2015 at 3:13 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit enables the function added with commit 7c10b87 and uses that new
function for setting any special scalefactor indices. This commit does not
change the behaviour of the encoder since no bands are being
to code
as noise. The spread[w+g] used before this patch behaved more like a low-pass
filter for PNS band_types, which could mistakingly mark some low frequency
bands as noise.
Reviewed-by: Claudio Freire klaussfre...@gmail.com
---
libavcodec/aacpsy.c | 2 +-
1 file changed, 1 insertion
being that
this version uses the new band-spread metric from aacpsy and normalizes the
energy using the group size. These changes were suggested by Claudio Freire
on the mailing list. Another change is the use of lambda to alter the
frequency threshold. This change makes the actual threshold
, but
marking M/S as present in encode_ms_info() is okay). Much of the changes here
were taken from the decoder and inverted. This commit does not change the
functionality of the decoder as the previous patch in this series zeroes
ms_mask and is_mask.
Reviewed-by: Claudio Freire klaussfre
function which handles setting of scalefactor indices for
PNS and IS bands.
Reviewed-by: Claudio Freire klaussfreire at gmail.com
Signed-off-by: Michael Niedermayer michaelni at gmx.at
http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7c10b87b5744179f16411f5981e96738021ec7ca
On Thu, Jul 2, 2015 at 3:13 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
There were some mistakes in the code for M/S stereo, this commit fixes them.
The start variable was not being reset for every window and every access to
the coefficients was incorrect as well. This fixes that by
On Thu, Jul 2, 2015 at 3:13 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit implements intensity stereo coding support to the native aac
encoder. This is a way to increase the efficiency of the encoder by zeroing
the right channel's spectral coefficients (in a channel pair)
LGTM, but it cannot be committed without the others, or a memset to
zero of is_mask somewhere at the beginning of each frame (or encoder
init).
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit adds support for the coding of intensity stereo spectral
On Mon, Jun 29, 2015 at 10:58 PM, Claudio Freire klaussfre...@gmail.com wrote:
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+if (spread NOISE_SPREAD_THRESHOLD
+((sce-zeroes[w*16+g] energy = threshold
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+if (dist2 = dist1) {
+cpe-is_mask[w*16+g] = 1;
+cpe-ch[0].is_ener[w*16+g] = ener1/ener01;
+cpe-ch[1].is_ener[w*16+g] = ener0/ener1;
+
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+++ b/libavcodec/aacpsy.c
@@ -781,6 +781,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx,
int channel,
psy_band-threshold = band-thr;
psy_band-energy= band-energy;
+
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+if (spread NOISE_SPREAD_THRESHOLD
+((sce-zeroes[w*16+g] energy = threshold) ||
+energy threshold*(NOISE_LAMBDA_NUMERATOR/lambda))) {
+
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit adds support for both PNS and IS (intensity stereo) codebooks to
the encode_window_bands_info() quantizer, used by the faast, faac and anmr
non-default, native coders. This does not mean that both
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+memset(cpe-ms_mask, 0, sizeof(uint8_t)*128);
sizeof(cpe-ms_mask) would be safer
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Seems straightforward enough. LGTM.
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit adds support for the coding of intensity stereo scalefactor
indices. It does not do any marking of such bands and as such does no
functional changes to the
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
This commit essentially undoes commit
c5d4f87e8427c0952278ec247fa8ab1e6e52 and removes PNS band marking from
the twoloop coder.
LGTM, but I wouldn't apply it before #09
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
+/* Energy spread threshold value below which no PNS is used, this
corresponds to
+ * typically around 17Khz, after which PNS usage decays ending at 19Khz */
+#define NOISE_SPREAD_THRESHOLD 152234544.0f
+
+/*
LGTM
On Fri, Jun 26, 2015 at 5:16 PM, Rostislav Pehlivanov
atomnu...@gmail.com wrote:
-float distortion;
-float perceptual_weight;
Those are in fact in disuse. Thought I'd clarify.
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On Mon, Apr 20, 2015 at 8:32 PM, Andreas Cadhalpun
andreas.cadhal...@googlemail.com wrote:
The long version:
ath should approximate the shape of the absolute hearing threshold, so
yes, it's best if it really uses the minimum, since that will prevent
clipping of the ath curve and result in a
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