I am using ffmpeg to write an .avi file. I have two streams, one for audio and other for video. fps of video varies from 5 - 15 and samples per second of audio varies from 60 - 80.
I am using av_interleaved_write_frame() to write the file. but it is too time consuming. So i want to replace it with av_write_frame().
But for that, i will have to manage the audio video interleaving in my code. I don't understand what combination i should use to interleave audio video properly.
Can anyone please help Or suggest me some sample code.
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I am currently using Kdenlive, but have also used ffmpeg when I have the simple task of adding audio to a video that does not yet have audio. Since it is just a matter of putting the video file together with the audio, it seems like it ought to be simple. Is there something about encoding mp4's that means it must take a lot of processing to complete?
I have good hardware (i7 6700k and gtx 1080), but kdenlive currently estimates 2.5 hours to complete adding audio to a 10 minute video.
Without more info (encoder, settings, video width x height, instructions to duplicate the behavior, etc) we can only guess. It's probably re-encoding the video instead of only muxing it. Encoding is CPU intensive and takes a long time. Although 2.5 hours for 10 minutes seems excessive, but there is not enough info in the question to say why it takes this long.
If you want to add audio with ffmpeg see How to add a new audio into a video using ffmpeg? This will allow you to mux the video (and optionally the audio) without encoding it: like a copy and paste.
As input I get one mp3 file(input.mp3), then I need to split it into two separate parts(done that) and insert between those parts another mp3(second.mp3).
The problem:
I tried using every single command for concatenating files, converting them to .ts etc etc.
But I always get only the sound of the first mp3 file and the sound of the second is lost. I guess I have to transform my mp3 file into the exact same format as the input.mp3(bitrates, sample rates number of channels) file before I can concatenate it to it.
I could concatenate the two parts of the input.mp3, but no matter what I do I cannot concatenate with second.mp3.
I am using php with exec and ffmpeg.exe file. Is it possible to code it so no matter the input, I can transform second.mp3 into suitable for concatenating mp3?
How can I concatenate two very different mp3 files(diff bitrate, channels, samplerate, bitdeph)?
Let's start with each component...
bitrate
This one doesn't matter. MP3 streams can (and do!) change bitrate mid-stream. As long as you join on a frame header, you're fine.
bitdepth
The concept of sample bit depth doesn't exist within MP3. You can capture at 24-bit, encode to MP3, and the decoder will decode to 16-bit. (Or, with some command line switches, vice versa!) It's not a problem because bit depth doesn't apply.
sample rates
This is usually a problem. Most players don't assume they're going to change output sample rate mid-stream. Most players don't attempt to resample to stick to the rate they were already outputting at. I'm not surprised that you'd have some trouble with a changing sample rate.
channels
This is similar to the sample rate problem in that it requires changing the configuration of the output device. Even if the player supports it, it isn't going to be seamless. (Unless you were going from stereo to mono, where the mono could be easily upmixed to stereo.)
As input I get one mp3 file(input.mp3), then I need to split it into two separate parts(done that) and insert between those parts another mp3(second.mp3).
This actually presents another problem you haven't asked about... timing. MP3 works in relatively large frames (typically 576 samples), which becomes the resolution at which you can splice. Not good. Also, the starts of tracks often have a frame or two of initialization.
A third issue is the bit reservoir. This is where content from one frame is stored in a different frame that might have extra space.
At the end of the day, you're going to have to decode everything to regular PCM samples, do your splicing, and re-encode to MP3. You'll also have to re-sample everything to a common clock rate, and mix to a particular channel count. Fortunately, once decoded to PCM, this is all trivial and standard. Once your input streams are compatible, you an arbitrarily splice on a PCM frame which is the most granular possible.
If you want to concatenate or merge different bit rate and Mono and Stereo mp3 files into one mp3 file use ffmpeg libmp3lame library.
Command :
ffmpeg -i "concat:'url1.mp3'|'mono_url2.mp3'|'stereo_url3.mp3'" -c:a libmp3lame output_file.mp3
Use the atrim, asetpts, and concat filters:
ffmpeg -i input.mp3 -i second.mp3 -filter_complex "[0:a]atrim=end=10,asetpts=N/SR/TB[begin];[0:a]atrim=start=10,asetpts=N/SR/TB[end];[begin][1:a][end]concat=n=3:v=0:a=1[a]" -map "[a]" output
Note: All corresponding streams must have the same parameters in all segments; the filtering system will automatically select a common sample format, sample rate, and channel layout for audio streams. These common parameters will vary depending on the input parameters, so add the aformat filter (or equivalent) if you want consistent results:
ffmpeg -i input.mp3 -i second.mp3 -filter_complex "[0:a]atrim=end=10,aformat=sample_rates=44100:channel_layouts=stereo,asetpts=N/SR/TB[begin];[1:a]aformat=sample_rates=44100:channel_layouts=stereo[middle];[0:a]atrim=start=10,aformat=sample_rates=44100:channel_layouts=stereo,asetpts=N/SR/TB[end];[begin][middle][end]concat=n=3:v=0:a=1[a]" -map "[a]" output
I am programatically extracting multiple audio clips from single video files using ffmpeg.
My input data (start and end points) are specified in frames rather than seconds, and the audio clip will be used by a frame-centric user (an animator). So, I'd prefer to work in frames throughout.
In addition, the framerate is 30fps, which means I'd be working in steps of 0.033333 seconds, and I'm not sure it's reasonable to expect ffmpeg to trim correctly given such values.
Is it possible to specify a frame number instead of an ffmpeg time duration for start point (-ss) and duration (-t)? Or are there frame-centric ffmpeg commands that I've missed?
Audio frame or sample numbers don't correspond to video frame numbers, and I don't see a way to specify audio trim points by referencing video frame indices. Nevertheless, see this answer for more details.
I have series of encoded packets, H.264 video and AAC audio. As they're coming on, I'm writing them to a video file, using av_write_frame.
Given the following situation in a row
10 seconds of video, then
10 seconds of video and audio, then
10 seconds of video.
Everything muxes fine and when played back via VLC or QuickTime, everything looks good. If I play it in Windows Media Player, the audio is played immediately.
It seems I'm doing something wrong, but checking the PTS of the audio stream packets, they are set to 10 seconds based on the time base of the audio stream.
It seems that it's best to inject empty audio packets at the beginning of the stream. This is the only way that video playback in WMP would work. Every player handles the streams differently and this is the best way to ensure compatibility across players.
I'm looking for a way to continuously grab frames, as jpg, from a RTSP stream. I've stumbled upon ffmpeg but it seems that the time between starting it and grabbing the first frame is quite high. Is there any good tool in order to do this?
Regards
I've used gstreamer libraries in the past to extract frames from mobile video