Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

[Amber Beriwal]

Hi,

 

Yes, we built FFMPEG libs by ourselves. What is Neon support?

 

Regards

Amber Beriwal

Newgen Software Technologies Ltd.

 

From: Libav-user [mailto:libav-user-boun...@ffmpeg.org] On Behalf Of William 
MANCON
Sent: Wednesday, September 21, 2016 6:53 AM
To: This list is about using libavcodec, libavformat, libavutil, libavdevice 
and libavfilter. 
Subject: Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

 

Did you built ffmpeg libs by yourself ? is there neon support ? 


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Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

[William MANCON]

Did you built ffmpeg libs by yourself ? is there neon support ?
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Re: [Libav-user] Audio encoding: more samples than frame size (avcodec_encode_audio2)

[Yu Ang Tan]

>
> > [libvorbis @ 028b1520] more samples than frame size
> > (avcodec_encode_audio2)
>
> Are you sure that you are encoding to flac?
>
>
You were spot on. I was using "test.ogg" instead of "test.flac".

I made the following changes and now it works fine.
-audio_decode_example("test.ogg", "test.sdp");
+audio_decode_example("test.flac", "test.sdp");

-outCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;   // FLAC uses fixed
point samples

Thanks!
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Re: [Libav-user] Demuxer not giving proper data to decoder ?

[Carl Eugen Hoyos]

2016-09-15 13:54 GMT+02:00 Carl Eugen Hoyos :
> 2016-09-15 13:41 GMT+02:00 ssshukla26 :
>
> [...]
>
> Did you already look at other hardware decoders in current FFmpeg?

You may be interested to see this patch on -devel:

crystalhd: Use up-to-date bsf API
http://ffmpeg.org/pipermail/ffmpeg-devel/2016-September/199734.html

Carl Eugen
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Re: [Libav-user] Audio encoding: more samples than frame size (avcodec_encode_audio2)

[Carl Eugen Hoyos]

2016-09-20 10:02 GMT+02:00 Yu Ang Tan :
> I am trying to read an RTP audio stream, and encode it into a FLAC file.
> However, when I am reading the stream, I get this error:
>
> [libvorbis @ 028b1520] more samples than frame size
> (avcodec_encode_audio2)

Are you sure that you are encoding to flac?

> During debug mode, it seems the frame_size of my input and output codec
> context are mismatched:
>
> inCodecCtx->frame_size = 1152;
> outCodecCtx->frame_size = 64;
>
> I tried to write 1152 to outCodecCtx->frame_size, but it gets overwritten
> with 64 at `avcodec_open2()`. Why can't I set the frame_size to match the
> input frame_size?

My guess is that you are using a different codec for output than for input.

Carl Eugen
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Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

[Amber Beriwal]

Hi,

 

Decode Only. We are not generating PNG. These are the timings of obtaining YUV 
data from frame.

 

Regards

Amber Beriwal

Newgen Software Technologies Ltd.

 

From: Libav-user [mailto:libav-user-boun...@ffmpeg.org] On Behalf Of William 
MANCON
Sent: Tuesday, September 20, 2016 3:31 PM
To: This list is about using libavcodec, libavformat, libavutil, libavdevice 
and libavfilter. 
Subject: Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

 

thoses time are the time to decode and generate the png ? 


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[Libav-user] UDP streaming from one PC to another

[Timur Guseynov]

Hi.

I am wondering if I can stream with ffmpeg from one PC via UDP to another
and the second one has ffmpeg using this stream as an input and save to
file?

Can I do it via CLI?
Will I be able to do the same programatically using its API?

Kind regards,
Timur Guseynov
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Re: [Libav-user] How to achieve 30 FPS using APIs on Android?

[William MANCON]

thoses time are the time to decode and generate the png ?
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[Libav-user] Audio encoding: more samples than frame size (avcodec_encode_audio2)

[Yu Ang Tan]

I am trying to read an RTP audio stream, and encode it into a FLAC file.
However, when I am reading the stream, I get this error:

[libvorbis @ 028b1520] more samples than frame size
(avcodec_encode_audio2)

During debug mode, it seems the frame_size of my input and output codec
context are mismatched:

inCodecCtx->frame_size = 1152;
outCodecCtx->frame_size = 64;

I tried to write 1152 to outCodecCtx->frame_size, but it gets overwritten
with 64 at `avcodec_open2()`. Why can't I set the frame_size to match the
input frame_size? Should I create an additional output frame to copy the
contents over in a loop?

I'd really appreciate it if you could help me some help or suggestions.

Here is my source:

//
// Test with this command: ffmpeg -re -f lavfi -i
aevalsrc="atan(t/2)*sin(400*2*PI*t)" -ar 16000 -c:a pcm_s16be -f rtp rtp://
127.0.0.1:8554
// ffmpeg -f dshow -i audio="Microphone (High Definition Audio Device)"
-ar 16000 -c:a pcm_s16be -ac 1 -f rtp rtp://127.0.0.1:8554
// Working code

#include "stdafx.h"
#include 
extern "C"
{
#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include 
}

#define AUDIO_INBUF_SIZE 20480

#define ERRBUFFLEN 200
char errbuf[ERRBUFFLEN];
#define av_err2str(ret) av_strerror(ret, errbuf, ERRBUFFLEN)
const int samp_rate = 16000;
int count = 0;
// Callback function
int _ffmpeg_interrupt_fcn(void* ptr)
{
int  = *((int*)ptr);
//double  = *((double*)ptr);
r += 1;
printf("Interrupted! %d\n", r);
if (r > 30) return 1;
return 0;
}

static int write_frame(AVFormatContext *fmt_ctx, const AVRational
*time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream
timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;

/* Write the compressed frame to the media file. */
#ifdef DEBUG_PACKET
log_packet(fmt_ctx, pkt);
#endif
return av_interleaved_write_frame(fmt_ctx, pkt);
}

/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char
*filename)
{
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
AVPacket inpkt, outpkt;

AVCodec *inCodec = NULL;
AVCodecContext *inCodecCtx = NULL;
AVFrame *decoded_frame = NULL;
AVFormatContext *inFormatCtx = NULL;

AVCodec *outCodec = NULL;
AVCodecContext *outCodecCtx = NULL;
AVFormatContext *outFormatCtx = NULL;
AVStream * outAudioStream = NULL;


int ret;

av_init_packet();

AVDictionary *d = NULL;   // "create" an empty dictionary
av_dict_set(, "protocol_whitelist", "file,udp,rtp", 0); // add an
entry

// Open video file
ret = avformat_open_input(, filename, NULL, );
if (ret <0)
{
printf_s("Failed: cannot open input.\n");
av_strerror(ret, errbuf, ERRBUFFLEN);
fprintf(stderr, "avformat_open_input() fail: %s\n", errbuf);
exit(1);
}

printf_s("Retrieve stream information.\n");
ret = avformat_find_stream_info(inFormatCtx, NULL);
if (ret <0)
{
printf_s("Failed: cannot find stream.\n");
av_strerror(ret, errbuf, ERRBUFFLEN);
fprintf(stderr, "avformat_find_stream_info() fail: %s\n",
errbuf);
exit(1);
}

av_dump_format(inFormatCtx, 0, filename, 0);

int stream_idx = -1;

for (int i = 0; i < inFormatCtx->nb_streams; i++)
if (inFormatCtx->streams[i]->codec->codec_type ==
AVMEDIA_TYPE_AUDIO) {
stream_idx = i;
break;
}
if (stream_idx == -1)
{
fprintf(stderr, "Video stream not found\n");
exit(1);
}

inCodec =
avcodec_find_decoder(inFormatCtx->streams[stream_idx]->codec->codec_id);
if (!inCodec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}

inCodecCtx = avcodec_alloc_context3(inCodec);
if (!inCodecCtx) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}

inCodecCtx->channels = 1;

ret = avcodec_open2(inCodecCtx, inCodec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
exit(1);
}

// Set output
ret = avformat_alloc_output_context2(, NULL, NULL,
outfilename);
if (!outFormatCtx || ret < 0)
{
fprintf(stderr, "Could not allocate output context");
}
outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER |
AVFMT_FLAG_FLUSH_PACKETS;

[Libav-user] Memory leak while using ff_get_buffer ?

[ssshukla26]

I am using *ff_get_buffer* in my decoder, it seems to me that using this API
causing some memory leak. Please help, need your suggestions on this !



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