I have a Google Hangouts app and I am trying to let the user play a sound that I provide.
Google has this covered, with its Audio Resource, but it only accepts specifically encoded sound files, PCM 16 wav files.
I have been trying to encode my files using ffmpeg, but it does not seem to be working.
Any idea as to what I am doing wrong?
Here is my ffmpeg command line :
ffmpeg -i sound.mp3 -map_metadata -1 -flags bitexact sound.wav
Thanks for your help
I just wrote hangout app that used audio and I noticed I had to use 44.1KHz sample rate on my 16 bit PCM WAV files or it wouldn't work. See if you can add an option to change the sample rate to that.
Related
In our application, we are processing audio files using ffmpeg. Specifically, we use the NodeJS library fluent-ffmpeg, (npm link).
Our audio files are generated from various text to speech providers. We recently noticed that when we converted audio using ssml to add pauses to the generated audio, the duration on the file is no longer correct. Upon further investigation, we noticed that the standard audios were also incorrect, just more accurate overall due to the more consistent data. When we put a pause at the beginning of the audio, the estimate was the worst, overshooting it by a very large margin (e.g., a 25s audio clip would read as 3 minutes long, but skip to the end when playing past the 25s mark.
I did some searching and research into the structure of MP3 files, and to me it seems like the issue is because the duration gets estimated by various audio players. Windows media player is an example, but Firefox's web player seems to also do this. I tried changing the ffmpeg command from using .audioQuality(0), which sets ffmpeg to use VBR, to .audioBitrate(320), which tells ffmpeg to use a constant bitrate.
For reference, the we are using libmp3lame, and the full command that gets run is the following, for the VBR and CBR cases respectively:
For VBR (broken durations): ffmpeg -i <URL> -acodec libmp3lame -aq 0 -f mp3 pipe:1
For CBR (correct duration): ffmpeg -i <URL> -acodec libmp3lame -b:a 320k -f mp3 pipe:1
Note: we then pipe the output to the requesting client application after sending the appropriate file headers, hence the pipe:1 output. The input is a cloud storage url where the source file is located
This fixes our problem of having a correct duration, and it makes sense to me why this would fix it if the problem was because the duration is being estimated by some of these players / audio consumers. But, this came at the cost that the file size was significantly larger, which also makes sense to me. While testing we found that compared to the same file in WAV, the VBR mp3 was about 10% of the WAV file size, while the CBR mp3 was still 50% of the WAV file size. This practically defeats the purpose of supporting the mp3 format for our use-case, which is a smaller but slightly lossy alternative to the large WAV file.
While researching, I found that there can be ID3 tags in a chunk at the beginning of the mp3 file, specifying information for the consumer of the audio to know the duration before potentially having processed the whole file. But, I also found that there doesn't seem to be a standard, at least for duration. More things like song title, album, artist, etc.
My question is, is there a way to get the proper duration onto an mp3 file, preferably via some ffmpeg mechanism, while still using VBR? Thanks!
FFmpeg does write a Xing header by default with duration info. However, that value is only known after the entire stream data has been received, so ffmpeg has to seek to the head to write it. Since you're piping the output, that can't be done.
Write the file locally or to some seekable destination, and then upload.
I want to add a 5.1 .flac audio track to a .ts file that already has three audio tracks. I tried with tsMuxer and ffmpeg with unsuccessful results. In tsMuxeR the .flac track is not recognized and in ffmpeg everything seems to work fine until the very last moment when I check the file and the .flac audio track is not included in the "output.ts". The .flac track is about 3GB and its lenght is around two and a half hours.
Thank you so much.
I don't think you'll find any existing software that maps FLAC into a MPEG-2 Transport Stream.
This gives you an idea what sort of issues you run into: https://xiph.org/flac/ogg_mapping.html
Let's say you came up with a reasonable way of mapping FLAC into a MPEG-2 Transport Stream - there won't be anything reading it.
Unless there is a specified way of mapping FLAC into a MPEG-2 Tranport Stream - you are on your own.
But PCM is supported in a MPEG-2 Transport Stream (for example Blu-Ray).
I'd use ffmpeg to transcode your audio from FLAC to PCM and then mux it into your transport stream.
Your audio transcode (FLAC to PCM) is lossless.
I would like to use MPEG-DASH technology in situations where I am constantly receiving a live video stream from a client. The Web server gets a live video stream, keeps generating the m4s file, and declares it in mpd. So the new segment can be played back constantly.
(I'm using FFMPEG's ffserver. So the video stream continues to accumulate in /tmp/feed1.ffm file.)
Using MP4Box seems to be able to generate mpd, init.mp4, m4s for already existing files. But it does not seem to support live streaming.
I want fragmented mp4 in segment format rather than mpeg-ts.
A lot of advice is needed!
GPAC maintainer here. The dashcast project (and likely its dashcastx replacement from our Signals platform should help you). Please open issues on github if you have any issues.
Please note that there are some projects like this one using FFmpeg to generate some HLS and then GPAC to ingest the TS segments to produce MPEG-DASH. This introduces some latency but proved to be very robust.
Below information may be useful.
latest ffmpeg supports the live streaming and also mp4 fragmenting.
Example command
ffmpeg -re -y -i <input> -c copy -f dash -window_size 10 -use_template 1 -use_timeline 1 <ClearLive>.mpd
I know that there are a million ways to download a video from youtube and then convert it to audio or do further processing on it. But recently I was surprised to see an app called YoutubeToMp3 on mac actually showing "Skipping X mb of video" and supposedly only downloading the audio from the video, without the need to use bandwith to download the entire video and then convert it. I was wondering if this is actually correct and possible at all because I cant find any way to do that. Do you have any ideas ?
EDIT:
After some tests here is some additional information on the topic. The video which I tried to get the audio from is just a sample mp4 file from the internet:
http://download.wavetlan.com/SVV/Media/HTTP/MP4/ConvertedFiles/MediaCoder/MediaCoder_test6_1m9s_XVID_VBR_306kbps_320x240_25fps_MPEG1Layer3_CBR_320kbps_Stereo_44100Hz.mp4
I tried
ffmpeg -i "input" out.mp3
ffmpeg -i "input" -vn out.mp3
ffmpeg -i “input” -vn -ac 2 -ar 44100 -ab 320k -f mp3 output.mp3
ffmpeg -i “input” -vn -acodec copy output.mp3
Unfortunately non of these commands seems to be using less bandwith. They all download the entire video. Now that you have the video can you confirm if there is actually a command that downloads only the audio stream from it and lowers the bandwith usage? Thanks!
After a lot of research I found out that this is not possible and developed an alternative approach:
Download the mp4 header
Parse the header and get the locations of the audio bytes
Download the audio bytes with http range requests and offsets
Assemble the audio bytes and wrap them in a simple ADTS container to produce a playing m4a file
That way only bandwidth for the audio bytes is used. If you find a better approach of doing it please let me know.
For a sample Android APP and implementation check out:
https://github.com/feribg/audiogetter/blob/master/audiogetter/src/main/java/com/github/feribg/audiogetter/tasks/download/VideoTask.java
FFmpeg is capable of accepting an URL as input. If the URL is seekable, then FFmpeg could theoretically skip all the video frames, and thus it would need to download only the data for the audio stream.
Try using
ffmpeg -i http://myvideo.avi out.mp3
and see if it takes less bandwidth.
I've searched on the web for something like this, but everything is about ´saving the file on ipod´ and some offtopic solutions and examples that i can't really use.
I want my linux laptop to stream the webcam media to my iPod (with my code) I am really lost on this.. I could use VLC to stream it in UDP from the laptop, but how would i get it and show on the iPod side? Should i use mpmovieplayer?
Note: I could send the frames in iplimage (from my linux code:blocks project) to the iPod, and for what i've searched on the web, there are methods to transform IplImage to UIImage on the iPod, but i don't know how to make it work :s
thks for any help.
I think you best bet would be to use ffmpeg. When you just want to stream videos that you have taken with your webcam you can just encode it correctly with ffmpeg and the iPhone will automagically do a progressive download, in most cases this will do.
Something like this :
ffmpeg -i $1 -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -cmp 2 -subcmp 2 -s 320x180 $1.mp4
when you however need real live streaming i would look at the following projects.
I played around with this project: http://www.ioncannon.net/projects/http-live-video-stream-segmenter-and-distributor/ but I remember it to be quite complex. It should however provide what you want.
Considering you are using a linux laptop, you might be interested in http://www.mythtv.org/wiki/Streaming_to_iPod_touch_or_iPhone
I think however it is missing the realtime component.
On the ipod side i think MPMoviePlayerViewController (ios 4.0) is the way to go, just implement the delegate methods and if the iPod can play it, this controller will handle everything.
I hope this helps, if you find an (easy) solution let us know ;-).