[FFmpeg-user] Not able to open URL using avformat_open_input
Hello all, I tried to open URL using avformat_open_input API function. Initially, I got a warning like, *Using network protocols without global network initialization. Please use avformat_network_init(), this will become mandatory later.* After initializing avformat_network_init(), I am getting a negative return value for avformat_open_input() function call. *INPUT URL:* https://mnmedias.api.telequebec.tv/m3u8/29880.m3u8 Can anyone please help me to find the issue? Regards, Kamalasubha M ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Replacing an audio stream in an MP4
On 5/7/2018 6:26 AM, Dan Bridges wrote: 1. Is this the most efficient way to perform this task? Could I mux the AVC stream in test.mp4 with the AAC stream in replacement.mp4 without first demuxing them. That is, could this all be performed in one FFMPEG command? No need to demux to separate files first. ffmpeg -i test.mp4 -i replacement.mp4 -map 0:v -map 1:a -c copy replaced.mp4 This will also be devoid of the PTS msgs, which were due to your use of .h264, which is a raw stream without timestamps. Regards, Gyan ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Replacing an audio stream in an MP4
I'm a heavy user of the Vegas 12 video editor. I record junior soccer matches with 2 cameras and two extra ext. sound recorders. Sometimes I notice a mixing balance problem in a rendered MP4 and need just to replace a remixed version of the audio stream in the MP4. Vegas won't output just a raw AAC file. But it will output a MP4 containing just the audio. So what I want to do is create a Windows desktop icon of a batchfile where I can drag-and-drop an affected MP4. The batchfile will then look for the presence of "replacement.mp4", the AAC-only MP4, in the same directory as the affected MP4. (I'll always use the same name for the fixed audio file.) If found, it will then perform the following sequence of FFMPEG commands (using "test.mp4" as an example of an AVC+AAC MP4 which needs to have its audio stream replaced): ffmpeg -i test.mp4 -an -c:v copy test.h264 ffmpeg -i replacement.mp4 -vn -c:a copy replacement.aac ffmpeg -i test.h264 -i replacement.aac -c copy test_replacement.mp4 The compression formats and stream indices will always be the same so I don't need to do any checking for that. 1. Is this the most efficient way to perform this task? Could I mux the AVC stream in test.mp4 with the AAC stream in replacement.mp4 without first demuxing them. That is, could this all be performed in one FFMPEG command? 2. I get these warnings during the muxing: "Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly" and a lot of "pts has no value". The audio appears to be in sync in the 18 minute clip I produced, so I don't think these warning are significant here. What do you think? 1 minutes versions of test.mp4 (720p50) and replacement.mp4 are available here: https://dl.dropboxusercontent.com/s/z7a7dh1dgqohx17/test.mp4 https://dl.dropboxusercontent.com/s/s6cax255fugc3ny/replacement.mp4 Dan. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?
On Sun, May 06, 2018 at 00:46:43 +0200, Nomis101 ? wrote: > Am 05.05.18 um 23:45 schrieb Lou Logan: > > Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000 > > OK, thanks. I did not know, that it is possible to look up the supported > sample rates like this. Are there any known plans to support 96000 in > the future? foo86 is the maintainer of ffmpeg's DCA code, but hasn't been active for the last few months. The ffmpeg docs and source aren't absolutely clear about it: I assume ffmpeg's DCA encoder only support DCA Core? It seems to me that the DCA standard says that DCA Core only supports up to 48 kHz. This could be the issue. You need an encoder which supports extensions. (To be clear: ffmpeg's *de*coder does support DTS X96k, which supports up to 96 kHz.) Cheers, Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".