Re: [Libav-user] Repeating frame while encoding using ffmpeg api

2014-09-30 Thread Pradeep Karosiya
Hi All, Thanks for the reply. So I think there is no straight forward to achieve variable fps, till now I've no success but I'll keep looking into it. However if someone has more information on variable fps and how to use repeat_pict field. Please share. Thanks and Regards, Pradeep -- View

Re: [Libav-user] Frame decoding and sw_scale timings for HD video

2013-08-26 Thread Pradeep Karosiya
Hi All, Thanks for all your suggestion. I'm considering using audio to synchronizing video instead of other way round. ìThough this may lead to frame dropping it is fine for me as long as audio sound good during play. For final display frames are converted into Iplimage image pointer and then

Re: [Libav-user] Frame decoding and sw_scale timings for HD video

2013-08-26 Thread Pradeep Karosiya
Hi, One of the things which I couldn't mention in my previous mail was that in my application. I've two video decoder one is based on ffms2 (a wrapper over ffmpeg/libav) and the other is based on ffmpeg. Initially I had just ffmpeg based video loader but since I was facing issues with seeking I

[Libav-user] Frame decoding and sw_scale timings for HD video

2013-08-23 Thread Pradeep Karosiya
Hi, I'm working on application which uses both ffmpeg and ffms (wrapper on ffmpeg) for decoding video file. The decoding audio and video is then played on user interface. For low resolution video till 640x360p both audio and video works fine and frame rate is maintained as per source file.

[Libav-user] No accelerated colorspace conversion found... message

2013-08-20 Thread Pradeep Karosiya
Hi, While encoding video file using ffmpeg I'm getting this warning No accelerated colorspace conversion found from yuv420p to bgr24. i think it is coming from sws_scale as I'm using this for conversion. Can someone tell me what could be the reason for this warning? Though things are working

[Libav-user] Warning No accelerated colorspace conversion found from yuv420p to bgr24

2013-08-20 Thread Pradeep Karosiya
Hi, While encoding video file using ffmpeg I'm getting this warning No accelerated colorspace conversion found from yuv420p to bgr24. i think it is coming from sws_scale as I'm using this for conversion. Can someone tell me what could be the reason for this warning? Though things are working

Re: [Libav-user] No accelerated colorspace conversion found... message

2013-08-20 Thread Pradeep Karosiya
I set av_log_set_level(AV_LOG_QUIET), though most of the message are disabled but still I'm getting just one message at the start. It normal or this can be disabled too. -- View this message in context:

Re: [Libav-user] Raw video encoding using AV_CODEC_ID_RAWVIDEO

2013-07-09 Thread Pradeep Karosiya
I've used av_interleaved_write_frame() for writing to .avi file. I'm facing this problem only for raw video. For AV_CODEC_ID_MPEG4 it works fine. Here are the two functions which I'm using for writing video and audio frames. //Member function to video frame void

[Libav-user] Saving video using ffmpeg from rtsp link

2013-06-06 Thread Pradeep Karosiya
Hi I'm trying to save video giev rtsp link using ffmpeg. The video copy continues for some time. But after 5mins it fails with unknown. I'm able to save only part of video. Is there any default timeout? My command ffmpeg -i rtsp:://input_video_link outputfile.3gp. Console log: warning,

Re: [Libav-user] # of audio samples, calculated vs. codec context

2013-05-22 Thread Pradeep Karosiya
Hi Brad, Actually at that time I didn't explore much and since I already had my own buffer class I quickly reused it. But if there are some other better facility available in ffmpeg like libavcodec/audio_frame_queue.c I suggest you cantry that. Thanks Pradeep -- View this message in context:

Re: [Libav-user] # of audio samples, calculated vs. codec context

2013-05-21 Thread Pradeep Karosiya
I had similar issue sometime back. I matched the number of sample to that of codec context. I have used buffering scheme and which still working for me, so I didn't explore any other option. I used intermediate buffer to keep the remaining samples and pass them to encoder when number of sample

[Libav-user] Audio encoding using avcodec_fill_audio_frame() and memory leaks

2013-05-17 Thread Pradeep Karosiya
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

Re: [Libav-user] Audio encoding using avcodec_fill_audio_frame() and memory leaks

2013-05-17 Thread Pradeep Karosiya
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.

Re: [Libav-user] Audio encoding using avcodec_fill_audio_frame() and memory leaks

2013-05-17 Thread Pradeep Karosiya
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

[Libav-user] Memory leak while opening encoders.

2013-05-14 Thread Pradeep Karosiya
Hi, I'm getting memory leaks in avcodec_find_encoder. Although I'm cleaning the resources properly still I'm not able to get rid of the leak. By successive commenting the code I found that memory leaks happen only after the call of avcodec_find_encoder(). I've tried my code with different video

Re: [Libav-user] ctx-request_sample_fmt

2013-05-10 Thread Pradeep Karosiya
I think it also depend upon the codec which you are using. If codec support that particular then probably you may be able to set. Though I'm bit new to ffmpeg most of the codec support just on sample format. For example AAC supports AV_SAMPLE_FMT_FLTP. AC3 has two variant ac3 and ac3_fixed.

Re: [Libav-user] Output encode file audio/video plays too fast in ffplay

2013-05-10 Thread Pradeep Karosiya
I've faced a similar problem before. I was just doing stream copy i.e changing the container format to avi. I think you care correct your pts and dts may not be correct. I too ended up playing with various combination. Try following. it worked for me. This is just for stream copy which involves

[Libav-user] Stream copy given start and end time

2013-05-10 Thread Pradeep Karosiya
Hi, I'm trying a small program which cuts clip from video file given start and end time. For this I'm doing a simple stream copy of both audio and video stream. I just copied the pts and dts from input packet to output with proper re-scaling. However the output which I'm getting is blank for some

Re: [Libav-user] Understanding planar data arrangement in AVFrame

2013-05-08 Thread Pradeep Karosiya
Thanks Nicolas for the brief explanation. It has helped me to understand better. -- View this message in context: http://libav-users.943685.n4.nabble.com/Understanding-planar-data-arrangement-in-AVFrame-tp4657508p4657537.html Sent from the libav-users mailing list archive at Nabble.com.

[Libav-user] Audio quality loss while encoding

2013-04-24 Thread Pradeep Karosiya
Hi I'm trying to encode decoded audio sample to an avi file. The audio samples are decoded from different file. So I've both input and output file. The decoded audio samples are in AV_SAMPLE_FMT_FLTP (float planar) in one large buffer. The first half contains channel 0 while second half of buffer

Re: [Libav-user] AAC with FLV

2013-04-24 Thread Pradeep Karosiya
Hi Brad, Have you found the solution to your issueof audio distortion. I'm also facing a similar issue while encoding with AAC and this is happening for same audio parameters sample rate: 44100, sample format: AV_SAMPLE_FMT_FLTP and number of channels = 2. For mono it is working fine. So I guess