[FFmpeg-devel] Audiotoolbox fix missing decoder out properties

2022-04-13 Thread MCC CS
This is a repost of the patch by Chris Snowhill who originally sent this patch
on 31 December - which didn't receive any replies. As you can see
on https://trac.ffmpeg.org/ticket/9564 the patch is needed and works perfectly.

I have rebased it and recompiled it.

Original commit description:

Monterey needs mBytesPerFrame and mBytesPerPacket to be set, and I'm
surprised this didn't break any previous system versions.

Fixes bug #9564: Cannot decode xHE-AAC with audiotoolbox (aac_at) on
Mac OS Monterey. Fixes likely bug that none of the AudioToolbox
decoders work on Monterey.
---
libavcodec/audiotoolboxdec.c | 5 +
1 file changed, 5 insertions(+)

diff --git a/libavcodec/audiotoolboxdec.c b/libavcodec/audiotoolboxdec.c
index b8d0e868f7..daccf4abf8 100644
--- a/libavcodec/audiotoolboxdec.c
+++ b/libavcodec/audiotoolboxdec.c
@@ -375,6 +375,11 @@ static av_cold int ffat_create_decoder(AVCodecContext 
*avctx,
 avctx->ch_layout.order   = AV_CHANNEL_ORDER_UNSPEC;
 avctx->ch_layout.nb_channels = out_format.mChannelsPerFrame = 
in_format.mChannelsPerFrame;

+out_format.mBytesPerFrame =
+out_format.mChannelsPerFrame * (out_format.mBitsPerChannel / 8);
+out_format.mBytesPerPacket =
+out_format.mBytesPerFrame * out_format.mFramesPerPacket;
+
 if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT)
 in_format.mFramesPerPacket = 64;

--
2.35.1
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[FFmpeg-devel] avcodec/audiotoolboxdec: fix missing decoder out properties

2022-04-13 Thread MCC CS
Resubmit: fix title.

This is a repost of the patch by Chris Snowhill who originally sent this patch
on 31 December - which didn't receive any replies. As you can see
on https://trac.ffmpeg.org/ticket/9564 the patch is needed and works perfectly.

I have rebased it and recompiled it.

Original commit description:

Monterey needs mBytesPerFrame and mBytesPerPacket to be set, and I'm
surprised this didn't break any previous system versions.

Fixes bug #9564: Cannot decode xHE-AAC with audiotoolbox (aac_at) on
Mac OS Monterey. Fixes likely bug that none of the AudioToolbox
decoders work on Monterey.
---
libavcodec/audiotoolboxdec.c | 5 +
1 file changed, 5 insertions(+)

diff --git a/libavcodec/audiotoolboxdec.c b/libavcodec/audiotoolboxdec.c
index b8d0e868f7..daccf4abf8 100644
--- a/libavcodec/audiotoolboxdec.c
+++ b/libavcodec/audiotoolboxdec.c
@@ -375,6 +375,11 @@ static av_cold int ffat_create_decoder(AVCodecContext 
*avctx,
 avctx->ch_layout.order   = AV_CHANNEL_ORDER_UNSPEC;
 avctx->ch_layout.nb_channels = out_format.mChannelsPerFrame = 
in_format.mChannelsPerFrame;

+out_format.mBytesPerFrame =
+out_format.mChannelsPerFrame * (out_format.mBitsPerChannel / 8);
+out_format.mBytesPerPacket =
+out_format.mBytesPerFrame * out_format.mFramesPerPacket;
+
 if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT)
 in_format.mFramesPerPacket = 64;

--
2.35.1
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[FFmpeg-devel] ping, ping, just 5 lines

2022-04-18 Thread MCC CS
The patch "avcodec/audiotoolboxdec: fix missing decoder out properties" hasn't 
been reviewed. Could anyone with some time, please, review this patch? 
Audiotoolbox decoder on macOS desperately needs this to be useful.
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[FFmpeg-devel] 2nd ping

2022-04-21 Thread MCC CS
It would be great if anyone who reads this could review this small patch 
(original: https://ffmpeg.org/pipermail/ffmpeg-devel/2022-April/295351.html):

This is a repost of the patch by Chris Snowhill who originally sent this patch
on 31 December - which didn't receive any replies. As you can see
on https://trac.ffmpeg.org/ticket/9564 the patch is needed and works perfectly.

I have rebased it and recompiled it.

Original commit description:

Monterey needs mBytesPerFrame and mBytesPerPacket to be set, and I'm
surprised this didn't break any previous system versions.

Fixes bug #9564: Cannot decode xHE-AAC with audiotoolbox (aac_at) on
Mac OS Monterey. Fixes likely bug that none of the AudioToolbox
decoders work on Monterey.
---
libavcodec/audiotoolboxdec.c | 5 +
1 file changed, 5 insertions(+)

diff --git a/libavcodec/audiotoolboxdec.c b/libavcodec/audiotoolboxdec.c
index b8d0e868f7..daccf4abf8 100644
--- a/libavcodec/audiotoolboxdec.c
+++ b/libavcodec/audiotoolboxdec.c
@@ -375,6 +375,11 @@ static av_cold int ffat_create_decoder(AVCodecContext 
*avctx,
 avctx->ch_layout.order   = AV_CHANNEL_ORDER_UNSPEC;
 avctx->ch_layout.nb_channels = out_format.mChannelsPerFrame = 
in_format.mChannelsPerFrame;

+out_format.mBytesPerFrame =
+out_format.mChannelsPerFrame * (out_format.mBitsPerChannel / 8);
+out_format.mBytesPerPacket =
+out_format.mBytesPerFrame * out_format.mFramesPerPacket;
+
 if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT)
 in_format.mFramesPerPacket = 64;

--
2.35.1
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Re: [FFmpeg-devel] 2nd ping

2022-04-25 Thread MCC CS
Nicely applied, many thanks!
 

Sent: Sunday, April 24, 2022 at 11:04 PM
From: "James Almer"
To: ffmpeg-devel@ffmpeg.org
Subject: Re: [FFmpeg-devel] [PATCH] avdevice/dshow: reuse unused variables.
On 4/19/2022 5:13 AM, Diederick Niehorster wrote:
> Fix for f125c504d8fece6420bb97767f9e72414c26312a, requested_sample_rate
> and such should be used.
>
> Signed-off-by: Diederick Niehorster 
> ---
> libavdevice/dshow.c | 6 +++---
> 1 file changed, 3 insertions(+), 3 deletions(-)
>
> diff --git a/libavdevice/dshow.c b/libavdevice/dshow.c
> index 1e69620880..5946a72cc2 100644
> --- a/libavdevice/dshow.c
> +++ b/libavdevice/dshow.c
> @@ -1003,9 +1003,9 @@ dshow_cycle_formats(AVFormatContext *avctx, enum 
> dshowDeviceType devtype,
> continue;
> }
> if (
> - (ctx->sample_rate && ctx->sample_rate != fx->nSamplesPerSec) ||
> - (ctx->sample_size && ctx->sample_size != fx->wBitsPerSample) ||
> - (ctx->channels && ctx->channels != fx->nChannels )
> + (requested_sample_rate && requested_sample_rate != fx->nSamplesPerSec) ||
> + (requested_sample_size && requested_sample_size != fx->wBitsPerSample) ||
> + (requested_channels && requested_channels != fx->nChannels )
> ) {
> goto next;
> }

Will apply.
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[FFmpeg-devel] Guidance needed for a semi-breaking change

2022-04-27 Thread MCC CS
Hi,

It's my first time writing an (original) patch for ffmpeg, so I would like your 
help.

I’m planning to propose a diff to an encoder param for aac_at, which is, as 
currently listed by "ffmpeg -h encoder=aac_at”:

  -aac_at_qualityE...A.. quality vs speed control (from 0 
to 2) (default 0)

Currently, 0 uses HIGH, and 1, 2 use MID, LOW quality encodings.

I’d like to change this param so that all 5 levels listed by Apple (MAX, HIGH, 
MID, LOW, MIN) can be used.

Do I change it to:

  -aac_at_qualityE...A.. quality vs speed control (from 0 
to 4) (default 0)

and migrate everyone to a higher-quality encoding preset

or
  -aac_at_qualityE...A.. quality vs speed control (from -1 
to 3) (default -1)

so that previous levels are kept as they are, adding only MAX and MIN, which 
were previously out of range.

I’d like to know: Which of these aligns better with FFMPEG’s policies?

Many thanks
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Re: [FFmpeg-devel] Guidance needed for a semi-breaking change

2022-04-27 Thread MCC CS
I wasn't considering a breaking change, those using
-aac_at_quality 0 or -aac_at_quality 2 would continue to use those.
However, would moving -aac_at_quality 0 from HIGH (currently highest
used by FFMPEG) to MAX be welcome? I'd assume that those using
-aac_at_quality 0 do so to use the highest quality option available.

If not, extending the range (0,2) to (-1,3) without moving any
previous options would work. From what I understand negative
numbers are OK in ffmpeg since `-q:a -1` produces higher quality
than `-q:a 0` for me.

How about this?

Thanks again

Sent: Wednesday, April 27, 2022 at 1:49 PM
From: "Timo Rothenpieler" 
To: ffmpeg-devel@ffmpeg.org
Subject: Re: [FFmpeg-devel] Guidance needed for a semi-breaking change
You need to keep API and ABI compatibility, cause otherwise it's a
breaking change which can only happen at a major bump.
Any and all existing applications and commandline need to keep working
as they do right now.

What you could do is convert the option into a OPT_TYPE_CONST one, and
offer the new values as actual textual option, and set up the numerical
values so that they align with what's currently offered.
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Re: [FFmpeg-devel] Guidance needed for a semi-breaking change

2022-04-27 Thread MCC CS
The default for AAC_AT is to produce highest quality audio, which I'll keep 
undoubtedly
in the same direction. However, the question is this:

Should the one who explicitly set `-aac_at_quality 0` (highest quality 
available) be moved from HIGH to MAX,
HOWEVER causing the side effect of moving everyone up in the quality-speed 
tradeoff by one,

(or we can eliminate HIGH and just replace it with MAX which has no tradeoffs 
AFAIK)

OR add `-1`, so that only those who entered no `aac_at_quality` be moved to -1,
and keeping explicit people having set `-aac_at_quality 0` at HIGH for them,
although they might have meant MAX when they started using it?
(i.e. no regression nor improvement for them)

---
Sent: Wednesday, April 27, 2022 at 2:18 PM
From: "Nicolas George" 
To: "FFmpeg development discussions and patches" 
Subject: Re: [FFmpeg-devel] Guidance needed for a semi-breaking change
Timo Rothenpieler (12022-04-27):
> Any existing commandline should produce the exact same output after the
> change.

I do not agree.

Small changes in default behavior are acceptable, since the user did not
specify what they want.

(I would argue that changes that fix a bug and changes that are
unambiguously and exclusively an improvement are acceptable.)

Regards,

--
Nicolas George
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Re: [FFmpeg-devel] Guidance needed for a semi-breaking change

2022-04-27 Thread MCC CS
Would you then approve changing current 0 from HIGH to MAX? Alternatively
-1 can be added until a major bump, while keeping 0 as default.
Either would be nondestructive.

And do you have an example/doc text about adding version guards to an arg?
I hope there's a general preprocessor variable for FFMPEGv6

Sent: Wednesday, April 27, 2022 at 4:18 PM
From: "Timo Rothenpieler" 
To: ffmpeg-devel@ffmpeg.org
Subject: Re: [FFmpeg-devel] Guidance needed for a semi-breaking change
On 27.04.2022 16:07, MCC CS wrote:
> The default for AAC_AT is to produce highest quality audio, which I'll keep 
> undoubtedly
> in the same direction. However, the question is this:
>
> Should the one who explicitly set `-aac_at_quality 0` (highest quality 
> available) be moved from HIGH to MAX,
> HOWEVER causing the side effect of moving everyone up in the quality-speed 
> tradeoff by one,
>
> (or we can eliminate HIGH and just replace it with MAX which has no tradeoffs 
> AFAIK)
>
> OR add `-1`, so that only those who entered no `aac_at_quality` be moved to 
> -1,
> and keeping explicit people having set `-aac_at_quality 0` at HIGH for them,
> although they might have meant MAX when they started using it?
> (i.e. no regression nor improvement for them)

The strictly correct approach here would be to add version guards, and
switch to the new behaviour next major bump.
I do agree though that that seems a bit over the top for this, and
specially if there is a higher quality than what ffmpeg currently
offers, I could definitely see keeping 0 as "highest possible quality".
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