[FFmpeg-user] libcamera_android build fix?

2023-10-25 Thread Andrew Randrianasulu
diff --git a/libavdevice/android_camera.c b/libavdevice/android_camera.c

 index 602b59047d..c981985f10 100644   ---
a/libavdevice/android_camera.c
+++ b/libavdevice/android_camera.c
 @@ -648,8 +648,8 @@ static int add_display_matrix(AVFormatContext *avctx,
AVStream *st)
 av_display_matrix_flip(display_matrix, 1, 0);
}

-side_data = av_packet_side_data_new(>codecpar->side_data,

->codecpar->nb_side_data,

 +side_data = av_packet_side_data_new(>codecpar->coded_side_data,

 +
>codecpar->nb_coded_side_data,
   AV_PKT_DATA_DISPLAYMATRIX,
 sizeof(display_matrix), 0);



something like this.

but it still does not work from console

configure:

./configure --enable-jni --enable-mediacodec --enable-neon
--target-os=android --disable-outdevs --disable-indev=xcbgrab

run:

./ffmpeg  -pixel_format nv12 -loglevel trace -f android_camera
-camera_index 1 -input_queue_size 5 -video_size 1280x720 -i 0:0 -c:v
libx264 -g 30 -b:v 980k -r 30 -s 1280x720 -b:a 28k 11test.mp4

it just sits here doing nothing, I guess waiting for application window?

[android_camera @ 0xb471aebc64d0] Android camera capture session is
active.


I found similar question at
https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg21783.html

and this explanation of  camera api:

https://www.sisik.eu/blog/android/ndk/camera
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[FFmpeg-user] Decode ac3 from multichannel USB input

2023-10-25 Thread Fred Rotbart

Hi,

I have a RME Digiface USB as input to my Mac. This has 32 SPDIF input 
channels, of which the first two have an ac3 signal and the others are 
not used.
I have been trying to isolate and decode one of the ac3 channels to its 
six separate PCM channels but without success.


For example, as one of the many attempts, I tried:

ffmpeg -acodec ac3 -f avfoundation -capture_raw_data true -i :1 \
-map_channel 0.0.0, \
-af 'pan=5.1|c0=c0|c1=-|c2=FC|c3=LFE|c4=BL|c5=BR' \
output.wav

Can someone help me with this?

- Fred

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Re: [FFmpeg-user] MP4-Video does not play audio on Android 8; audio stream size tagged wrong

2023-10-25 Thread Ulf Zibis



Am 25.10.23 um 06:50 schrieb Classic SSG Enjoyer:

Use Opus. Opus delievers better audio quality than AAC/Vorbis

On Tue, 24 Oct, 2023, 11:01 pm Ferdi Scholten,  wrote:


Hi,

Vorbis audio is officially not supported by the mp4 container
specifications.
Matroska does not have this problem as it supports almost all known
codecs.

These are the audio codecs that are in the mp4 specifications: AAC,
ALS, SLS, MP3, MP2, MP1, CELP, HVXC, TTSI, SAOL, Apple Lossless,
FLAC, Opus
Others can be muxed in, but most players will not support playing those.

Much thanks for this valuable information.

Because I want to add several audio streams to a MP4 video, I'm
looking for an audio codec, with optimal compression. As AAC in my
context needs 128 kb/s, but Vorbis only 90 kb/s, I thought, it was a
good choice.

Which audio codec you would suggest, which has a wide range of
compatibility, except AAC?

Maybe Opus?

For the video I want to use HEVC.

Is there a possibility to encode AAC_HE with FFmpeg and would it make
sense?

-Ulf

For good compression and good quality you should use Opus, it delivers
(much) better audio quality than any variant of AAC let alone Vorbis for
the same bitrate. when you don't specify a bitrate, libopus uses 95 kb/s
as default for stereo audio.
-Ferdi


Thanks to Ferdi and Classic.
Yes, Opus plays well on Android and iPhone.

-Ulf

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[FFmpeg-user] ffmpeg git and makeinfo 7.1 error?

2023-10-25 Thread Andrew Randrianasulu
I am trying to build new ffmpeg on my aarch64 termux installation.

pulled git long time ago, now tried to integrate libxcam filter (without
libxcam installed for now) but I think I run into unrelated build error:

makeinfo: error parsing ./doc/t2h.pm: Undefined subroutine
::Config::set_from_init_file called at ./doc/t2h.pm line 24.

using

makeinfo --version
texi2any (GNU texinfo) 7.1


as packaged by termux.

configure line was

./configure --enable-jni --enable-mediacodec --enable-neon
--target-os=android --disable-devices

git log set at base a4877f1ec11e0327684eda37b15bc1ccb16f9ace

and just-compiled (with "make ffmpeg" after docs build err) ffmpeg says:

~/ffmpeg $ ./ffmpeg -h
   ffmpeg version N-112509-g265b40e489 Copyright (c) 2000-2023 the FFmpeg
developers
  built with clang version 17.0.3
  configuration: --enable-jni --enable-mediacodec --enable-neon
--target-os=android --disable-devices
  libavutil  58. 27.100 / 58. 27.100
libavcodec 60. 30.102 / 60. 30.102
  libavformat60. 15.100 / 60. 15.100
libavdevice60.  2.101 / 60.  2.101
libavfilter 9. 11.100 /  9. 11.100
  libswscale  7.  4.100 /  7.  4.100
  libswresample   4. 11.100 /  4. 11.100

libxcam filter at patchwork:

https://patchwork.ffmpeg.org/project/ffmpeg/patch/20200731145710.114479-1-wei.z...@intel.com/#56952

I just added const keyword to libavfilter/allfilters.c change, like in
other cases
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[FFmpeg-user] MKV to MP4

2023-10-25 Thread Mark Filipak
I'm trying to remux MKV as MP4. In the process, I want to remove the
rounding errors that were introduced by MKV so that the video is
24/1.001fps CFR once again with proper PTSs & DTSs and 1/9 time
base. When I do that, audio & video are out of sync.

Does anyone know how to do this? I'd be really grateful.

Regards -- Mark.

PS: Since I wrote this I may have stumbled on a partial answer. I'd
like your experiences with that partial answer. The partial answer is
to add only this bit stream filter:
-bsf:v setts=time_base=1/9
and to NOT add either :pts=... or :dts=...
As I wrote, I stumbled on it through a fortuitous mistake. It seems to
work, even for videos that completely lack PTSs & DTSs. Comments?
Please.
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