I have an iOS project that play online radio streams, it is use FFmpeg to play. Also I added ability to record streams, decode streams via avcodec_decode_audio4 function, and write output to .wav file. But this files are too big, because it is uncompressed format, so I want to decode files to .mp3.
I have found couple ways to convert audio but only when audio it is ready file, but I want decode to some compressed format as soon as I get chunk of data from stream, not ready file.
Is it possible?
Can you give me some advise how to achieve this?
You can use ffmpeg (aka libav) to encode the audio you're reading with avcodec_decode_audio4 into a file as mp3, as long as libav was configured with lame (--enable-libmp3lame).
Basically, you configure an mp3 codec, then call avcodec_encode_audio2 (who names these things?) on the progressive output of avcodec_decode_audio4.
The canonical example can be confusing because it also deals with video, but you should be able to tease the details you want out of it.
This post on transcoding audio by arashafiei is broadly helpful.
Related
Good day everyone!
So, as the title suggests, i am developing an app with similar functionality to that off Periscope and Facebook Live video streaming. Here is what the end goal is:
A Broadcasting device [user]
EC2 Instance [Hosting an ffmpeg transcoder]
Cloudfront Distrubution [CDN]
1 to n viewers of the live feed
I've been doing a lot of googling and what I cant seem to figure out is:
As you send chunks of video to the server from the Broadcaster, how do
you create an
.m3u8 playlist when you don't have all the chunks of video yet (e.g. the
device sends its first 5second chunk of video)?
It seems a .m3u8 file is created from a .mp4 file that is already complete, then broken down into chunks... But i'm sending chunks of the video to the server, how can it generate the .m3u8 file when more chunks are still coming from the Broadcaster, so the watchers / clients can continuously stitch together the video chunks?
I'll be happy to clarify this question further. Thanks!
If you take a look at the docs for the segment muxer you can specify the m3u8 to be outputted and you can also tell it to update the m3u8 as it goes. It might look something like this:
ffmpeg -i infile.mp4 -c:v copy -c:a copy -map 0 -f ssegment -segment_list playlist.m3u8 -segment_list_type hls -segment_list_size 10 -segment_list_flags +live -segment_time 4 outchunk%07d.ts
Note the segment_list_size is the maximum number of chunks referenced in the m3u8 file at one time and the segment_list_flags tells ffmpeg that this a live stream.
I think your confusion is that you are trying to send HLS fragments to their server. Don’t. Send a stream via another protocol like RTPM. Then let the server convert to HLS.
Which audio file format is best to use for large audio files? I have many large audio files to be used in my app but their current mp3 size is of hundred of MB's
If you want to save more storage on audio files, file format may not change too much on the file size, reducing the bit rate(for example 320Kbps to 128Kbps) can reduce the file size significantly.
:how to do it using microsofts audio compression manager?(practically its not well documented in m.s.d.n.
Windows provide codecs that compress specifically audio files. The audio files tipically are PCM format (WAVE_FORMAT_PCM) and get played by using the simplest directsound method (check msdn it`s at hand and it works)
To play a file using directsound, thus PCM format you first create a directsound object, create a directsoundbuffer, and then pump the PCM data directly to the buffer using a keep-fill-buffer algorithm.
If you wish to use codecs, u try and write a procedure that opens a stream file and passes it through a acm driver object, thus (de)compressing it.
The driver for ACM (audio compression manager) finds a codecs that suits the input source and decompresses it yet again to WAVE_FORMAT_PCM for your app be able to play it.
I am trying to write a live video broadcaster over RTSP from an ios device. I am utilizing AVAssetWriter so I can take advantage of hardware encoding. To send over RTSP I have to get the avcC information out of the MOOV block, however the MOOV block is only written from AVAssetWriter when you have finished the session, which of course is not finished as I am streaming this live.
I have gotten around this with the video by encoding, writing, and then finishing a single sample buffer to file, and the parsing the file to get the avcC information out. That works just fine.
After that for the live stream, since AVAssetWriter will only write to a file, I am writing it out to file and then reading from that file with a chasing file offset. When I do this with video only, I can read the Nalu's from the MDAT Atom in the written file without any MOOV information as the size of each Nalu is given in the first 4 bytes of the Nalu. So I can read that amount, process it, and send it on its way over an RTSP stream. So with video only, everything works perfectly fine and I get real good HD stream to a stream server.
The problem I am now having is when I try to incorporate audio into the stream from the mic. I can encode it just fine with AVAssetWriter and I get proper interleaved formated mp4 file to read from, however unlike the H264 Nalu's, the audio samples in the file do not have the size of the sample as their first byte. So far the only way I can see to define that is with the STSZ and STCO Atoms in the MOOV, which of course I dont have because it is a live stream.
With all that in mind, does any one know a way to identify audio sample segments in an MDAT Atom without the information from the MOOV Atom? As soon as I figure that out, Im home free.
Thanks in advance for any insight.
After a lot of research and emails out to people, I at least have an answer, and the answer is, I cant do it this way. Normally AAC samples in streams where dont have an index is wrapped in ADTS headers which holds the length field for the packet. However, since I am using AVAssetWriter for the audio, and AVAssetWriter writes directly to an MP4 file, the ADTS wrap is stripped off because of the index that will be in the MOOV Atom.
Therefore I will have to encode the audio differently, probably through Audio Queue services and meld it into the Video packets when applying to the RTSP stream.
Maybe this will help someone else in the future looking down this same road.
Many thanks to Geraint Davies at http://www.gdcl.co.uk for leading me down the right path.
Is this possible to access the raw audio PCM data that is being played when using XAudio2 to play file?
I've been searching for several ways to access a decoded version of audio files being played in SL4/Windows Phone, without success.
According to this post someone had success writing a custom XAPO that just grabs samples and is enabled on a Submix Voice. http://social.msdn.microsoft.com/Forums/windowsapps/en-US/05593fad-dfd8-4c77-983b-8c84cd4a324b/xaudio2-saving-output-custom-xapos-slow-down-audio-play-backwards
Please note that if you just want to do this for audio processing this approach is not optimal because you are limited to the speed of audio playback.
I'm trying to teach myself a bit about video streaming and transcoding, with some Roku app development on the side. I have a number of video files (mostly in FLV format (H.264/AAC)) that I would like to stream to a client, which in this case is a Roku box (that accepts MP4 (H.264/AAC) and HTTP Live Streaming (HLS)). I'm wondering if it is possible to transcode/remux the FLV files and stream them to the client on the fly, perhaps over HLS?
I have tried using ffmpeg to remux the files and serve them immediately during the transcoding process, but they are unplayable until the write process is complete. I can get the Roku to play my completed MP4 files just fine via Apache/Rails.
But I'm wondering... is it possible to set up a server to transcode/remux a file and immediately have the output file (from ffmpeg/whatever tool I'm using) streamed to the client? If so, what tools are required to accomplish this? Is it possible to use a media file segmenter to chop up a file as it's being transcoded or remuxed?
I'm well aware that the transcoding process is CPU intensive, but I'm not so much worried about the practicality of transcoding and streaming on the fly since this is simply a personal education project (and I have an idle system that is capable if handling this).
Apologies if I'm way off base here, just trying to hack my way through this.
Thanks!
The trick to getting HLS served immediately that a TS segment has been completed is getting the playlist to dynamically update as the data arrives on disk.
What you are trying to do is essentially stream a Live event over HLS, which absolutely can be done, it just takes co-ordination between tools.
The opensource segmenter is able to do this, the trick is to have ffmpeg write out a single MPEG-TS stream (Unsegmented) and write this to a named pipe (Or equivalent for your OS), then have segmenter read from this named pipe and write the files to a directory within your shared webspace.
The segmenter repeatedly updates the M3U8 file on disk while processing so it can be used as a "Live" stream until the task is finished.
When ffmpeg closes its output the segmenter puts the end tag in the M3U8 and the file becomes "VOD".
The segmenter can be downloaded here