gmane.comp.video.ffmpeg.libav.user

Re: MPEG-2 NTSC video shows horizontal lines on thered color

nuo yang — 2013-05-19T09:52:47

Hi Carl,
 
Thanks for your help, and now I fixed my issue!
 
You are right, my issue is with upsampling the 4:2:0 mpeg video to 4:2:2 feeding the decklink card. I read the article and found an upsampling filter to convert  chroma format from 4:2:0 to 4:2:2. The upsampling filter can handle two types of frame, it is progressive and interlaced. Previous I used a ffmpeg function sws_scale to convert the chroma format, it has this issue, and it is the same issue when I convert the chroma format using this upsampling filter by assuming the frame is progressive. But things changed when I assume the frame is interlaced, it is difficult to see horizontal lines on the red color. I think ffmpeg treated a frame as a progressive frame when doing the chroma upsampling using sws_scale in my video player.

Thanks,
Nuo 

________________________________

From: nuo yang <yangnuo0408-/E1597aS9LRv1O+Z8WTAqQ< at >public.gmane.org>
To: "This list is about using libavcodec, libavformat, libavutil, libavdevice and libavfilter." <lib

# of audio samples, calculated vs. codec context

Brad O'Hearne — 2013-05-18T20:11:09

I have the following audio use-case: 

audio capture -> resample captured audio to destination format for encoding -> encode audio -> stream audio

I have developed an app which has worked decently well for fairly common sample rates (44100, 48000). However, I came across a sample rate of 16000 which is breaking the app. The problem stems from the calculation of the number of audio samples in the destination data exceeding the codec context's frame size. 

In the resampling_audio.c example, there it shows the following means to calculate the destination number of samples: 

 dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

Here are the values for these variables: 

src_rate = 16000
src_nb_samples = 512
dst_rate = 44100

and the calculated value: 

dst_nb_samples = 1412

However, the codec context's frame size is set (by the encoder, as per the documentation) at 1152, smaller than the calculated value.

sws_scale crash

yy-zed — 2013-05-18T15:08:51

Hi Guys, 

I'm trying to figure out to cause of constant crashes on sws_scale calls.
I'm reading in a video frame, decoding it, then trying to scale, encode and
write to video. Reading the frame and decoding it seem to be working fine,
but sws_scale keeps crashing.

Here's where I set the context:


p_codec_ctx
=p_in_video_ctx->p_fmt_ctx->streams[p_in_video_ctx->i_video_stream_idx]->codec;

p_out_video_ctx->p_sws_ctx = sws_getContext(p_codec_ctx->width,
p_codec_ctx->height, p_codec_ctx->pix_fmt,
p_out_video_ctx->i_vwidth, p_out_video_ctx->i_vheight,
AV_PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);

and here's where I make the call to scale the frame:

debugLogR->debug("here1");
sws_scale(p_out_video_ctx->p_sws_ctx,
(uint8_t const * const *) p_in_video_ctx->p_video_frame->data,
p_in_video_ctx->p_video_frame->linesize, 0, p_codec_ctx->height,
p_out_video_ctx->p_video_frame->data,
p_out_video_ctx->p_video_frame->linesize);

debugLogR->debug("here2");

here1 is printed, but the program exits wit

Re: iZeranoe latest build and glut

Gonzalo Garramuño — 2013-05-17T16:40:18

El 17/05/2013 01:09 p.m., Gonzalo Garramuño escribió:
Oh yes.  This is on the 64-bits builds.

Re: iZeranoe latest build and glut

Carl Eugen Hoyos — 2013-05-17T16:37:31


I believe there is a forum dedicated to such 
questions.

Carl Eugen

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iZeranoe latest build and glut

Gonzalo Garramuño — 2013-05-17T16:09:22

I am getting a strange behavior in the latest zeranoe builds of ffmpeg 
when I link them to my program.  The program fails to start and 
complains about avcodec not knowing glutStrokeCharacter.  I used 
dependency walker and indeed there are failed entries for glut.  I am 
currently linking against freeglut.  ffmpeg and ffplay don't have a 
problem.  Using the previous zeranoe build does not show the problem, 
but it uses avcodec-54 and I would like to link against the latest.  My 
guess is something is wrong with the makefile but cannot figure what.
Can someone suggest what could be the cause of the problem and something 
to try?
Thanks in advance.

Re: Audio encoding using avcodec_fill_audio_frame() and memory leaks

Pradeep Karosiya — 2013-05-17T13:07:48

After adding code to free memory manually, just before calling
av_free_packet(), basically I added the same code as in av_destruct_pkt().
But to my surprise I'm getting a crash in av_interleaved_write_frame for
video till now video was working fine. I'm writing video and audio frame
sequentially and I'm using sperate AVPackets and AVFrames for both audio and
video. There is no shared data between audio and video except
AVFormatContext which is global.
Please someone suggest what could be going wrong.

Thanks and Regards,
Pradeep




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Re: Audio encoding using avcodec_fill_audio_frame() and memory leaks

Pradeep Karosiya — 2013-05-17T12:20:37

Hi Kalileo

Thanks for the reply. I've noted that If I just disable
av_interleaved_write_frame and keep the remaining code as it is then there
is no memory leak. The memory for AVPacket is getting allocated in
avcodec_encode_audio2 and if I use av_free_packet with destruct field set to
av_packet_destruct than there is no crash. However if I enable
av_interleaved_write_frame then I'm seeing a crash if destruct field is set. 
I tried following
if(pkt.data)
{
  av_free(pkt.data);
}
av_free_packet(&pkt)
But again I'm seeing exception.
So what all memory do I need to free before calling av_free_packet. And
strangely this is happening only for audio my video encoding is working
fine.

Thanks
Pradeep



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Re: Audio encoding using avcodec_fill_audio_frame()and memory leaks

Kalileo — 2013-05-17T11:45:17

On May 17, 2013, at 17:47 , Pradeep Karosiya wrote:


I don't think you need to go the route of av_destruct_packet.

If you really see a leak, then try freeing the data in the packet before you run av_free_packet.

Re: Audio encoding using avcodec_fill_audio_frame() and memory leaks

Pradeep Karosiya — 2013-05-17T10:47:13

Ok after successively disabling the code I found that memory leak is coming
from av_interleaved_write_frame(), even though I'm using av_free_packet, it
is still there.
I tried to use av_destruct_packet just before av_free_packet but then my
application is crashing while calling av_destruct_packet. 
Actaully I'm setting mpkt.destruct to av_destruct_packet before calling
av_interleaved_write_frame but 
somehow mpkt.destruct is setting to NULL after the call of
av_inteleaved_write_frame, so I'm setting it again before calling
av_free_packet.
Can someone explain the correct use to AVPacket.destruct and
av_destruct_packet.

Thanks
Pradeep



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Audio encoding using avcodec_fill_audio_frame() andmemory leaks

Pradeep Karosiya — 2013-05-17T08:25:01

Hi,

As a part of encoding decoded audio packets, I'm using
avcodec_fill_audio_frame(). I'm passing allocated AVFrame pointer to along
with buffer containing the decoded samples and other parameters number of
channels, sample format, buffer size. Though the encoding is working fine
I'm not able to completely eliminate the memory leaks. I've taken care of
most of things but still I'm not able detect the leakage.
Below is the function which I'm using for encoding. Please suggest
something.
AudioSample contains decoded data and it is completely managed in different
class(free in class destructor). I'm freeing the AVFrame in FFmpegEncoder
destructor and AVPacket is freed every time using av_free_packet() with
av_packet_destruct enabled. What more do I need to free?


void FfmpegEncoder::WriteAudioSample(AudioSample *audS)
{


    int num_audio_frame = 0;
    AVCodecContext *c = NULL;
   // AVFrame *frame;
    AVPacket pkt;

    av_init_packet(&pkt);
    pkt.destruct = av_destruct_packet;
    pkt.data = NULL;

FFmpeg, raspberry

Denis — 2013-05-16T19:46:44

Do the FFmpeg libraries support the raspberry h264 hardware encoding?
Have you got any example on how I can use it for encoding?

Re: Audio playing to fast using fflplay

Andrey Utkin — 2013-05-16T15:24:53

2013/5/15 iwastemoretimethanu <willbeas88-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org>:

I've used another pts assignment formula in transcoding apps.
For audio, you should set pts of uncompressed-yet AVFrame with raw
samples to _offset from stream beginning, in samples_. It means if you
have audio < at > 48000 Hz, the raw audio timestamps should be stamped in
units of  1/48000 of second.
Then you put this AVFrame into encoder and get AVPacket. I don't
remember to what AVPacket.dts, .pts is set to after this, you should
check. Either libavcodec rescales timestamp to timebase of media file
(i.e. 1/1000 for FLV, 1/90000 for MPEG TS), or you have to do it by
yourself.
That's all.

--
Andrey Utkin

Re: Does anyone know how to generate quicktime compatiable H264 video by ffmpeg?

Paul B Mahol — 2013-05-16T10:52:37

Its important where you added it.

Without full uncut ffmpeg output its guessing game.

Re: Does anyone know how to generate quicktime compatiable H264 video by ffmpeg?

Xian Yan Yang — 2013-05-16T10:46:54

It's already added,still not work. Same problem happened to realplayer,I
wonder if there is any special parameters that I need to take care?

Thanks
BR.Luffy


2013/5/16 Paul B Mahol <onemda-Re5JQEeQqe8AvxtiuMwx3w< at >public.gmane.org>

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Re: Does anyone know how to generate quicktime compatiable H264 video by ffmpeg?

Paul B Mahol — 2013-05-16T10:38:03

Add -pix_fmt yuv420p, because quicktime supports only baseline h264.

Re: Reference app with ffmpeg n1.2 libs that works onIOS ?

Carl Eugen Hoyos — 2013-05-16T08:58:21



Please consider reading this paragraph again.

Please do not top-post here, it is considered rude.

Carl Eugen

Re: How to disable zlib include

Carl Eugen Hoyos — 2013-05-16T07:48:07


./configure --disable-zlib

Carl Eugen

How to disable zlib include

YIRAN LI — 2013-05-16T07:31:32

Hi,

Several days ago I succeeded in building ffmpeg with zlib support, I think
I just got zlib code, made and installed, then ffmpeg build automatically
included it.

But today I tried to disable zlib support, but couldn't do it. I
uninstalled header files and libs. and I can see the configure result is
"zlib enable no, bzlib enabled yes'

But why, after I opened generated ffmpeg with dependency walker and found
that AVCODEC-54.DLL still needs ZLIB1.DLL

Could anyone tell me how to do? Because zlib support is auto detected, is
there anyway to explicitly disable that?

Great thanks!
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Re: Audio playing to fast using fflplay

Alex Cohn — 2013-05-16T05:18:28

I think that most container formats don't support audio at 5 FPS. I would
recommend to split the audio stream into 20 ms packets.

BR
Alex
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Re: Reference app with ffmpeg n1.2 libs that works on IOS ?

Will Price — 2013-05-16T03:40:57

After a lot of work, I have arrived at the same issue last posted to this
list by Lars on April 5. That is the following warnings on link:

ld: warning: ARM function not 4-byte aligned: __b_evaluation from
ffmpeg/lib/libavcodec.a(simple_idct_arm.o)
ld: warning: ARM function not 4-byte aligned: __a_evaluation from
ffmpeg/lib/libavcodec.a(simple_idct_arm.o)
ld: warning: ARM function not 4-byte aligned: __end_b_evaluation2 from
ffmpeg/lib/libavcodec.a(simple_idct_arm.o)
ld: warning: ARM function not 4-byte aligned: __a_evaluation2 from
ffmpeg/lib/libavcodec.a(simple_idct_arm.o)
ld: warning: ARM function not 4-byte aligned: __end_simple_idct_arm from
ffmpeg/lib/libavcodec.a(simple_idct_arm.o)

The thread ensued from there to fix a different R_ABS reloc issue, but did
not come back to these link problems. Any suggestions on getting rid of the
above link warnings?

Thanks.



On Fri, May 10, 2013 at 11:46 PM, Carl Eugen Hoyos <cehoyos-iVVjmtzCFAg< at >public.gmane.org> wrote:

__________________________________________