I am developing a player which plays different audio formats,I have successfully played mp3,acc,wav,ogg,alac,flac formats. But now stuck on APE or Monkey's Audio format, I am using a open source library FFmpeg for decoding.Can someone tell me how to decode APE format,I am successful in creating the packets,storing in a buffer and passing it to FFmpeg in the same fashion as I did for other formats,but FFMpeg is not able to decode those packets.Can someone tell me what wrong I am doing? What should be the buffer size in case of ape that needs to be passed to ffmpeg.Also header parsing, The decode function of ffmpeg is avcodec_decode_audio3.
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I need a software that can stream audio with mp3 format.The audio will come from the microphone at the same time.
I have a software that can stream sound with alaw and ulaw codecs.
And I have an another program that can stream recorded mp3 file. Not capture from the microphone.
I can make stream with VLC.Dotnet wrapper but I didn't succeed with directshow.(namely microphone)
Here my Vlc.Dotnet code;
myVlcControl.Play("dshow://");
myVlcControl.Play(new Uri("dshow://"));
It did not work with this codes. I don't know what causes the problem.
My second software can stream sound that captures form microphone in real time. But its codec format is alaw not mp3. I did not find any converter that convert linear to mp3 file. I find a converter that convert linear to alaw. this is the link Linear to Alaw Codec
I know the LAME and NAudio but it converts wav file to mp3. I need linear to mp3(like in the link)
I am very confused. I really do not know which way to go.
1. Find a codec linear to mp3 (It's very complicated) ?
2. Learn VLC direct Show usage on .NET ?
Thank you so much in advance.
*VLC.DotNet, axVLCPlugin21, LAME, ffmpeg....
As soon as I've successfully solved this problem with VLC.Dotnet wrapper. The problem is compiling with x64 architectural. When I was compiled with x86 architectural, The problem solved.
I'm developing app which sends mpeg2ts stream using FFMPEG API.(avio_open, avformat_new_stream etc..)
The problem is that the app already has AAC-LC audio so audio frame does not need to be encoded because my app just bypass data received from socket buffer.
To open and send mpegts using FFMPEG, I must have AVFormattContext data which is created from FFMPEG API for encoder as far as I know.
Can I create AVFormatContext manually with encoded AAC-LC data? or I should decode and encode the data? The information I know is samplerate, codec, bitrate..
Any help will be greatly appreciated. Thanks in advance.
Yes, you can use the encoded data as-is if your container supports it. There are two steps involved here - encoding and muxing. Encoding compress the data, muxing mixes it together in the output file, so the packets are properly interleaved. Muxing example in FFMpeg distribution helped me with this.
You might also take a look at the following class: https://sourceforge.net/p/karlyriceditor/code/HEAD/tree/src/ffmpegvideoencoder.cpp - this file is from one of my projects, and contains video encoder. Starting from the line 402 you'll see the setup for non-converted audio - it is kind of a hackish way, but it worked. Unfortunately I still end up reencoding audio because for my formats it was not possible to achieve frame-perfect synchronization which I needed
I currently working on a project simulate webcam video transmission in C++, at sender side, I capture the raw webcame video with v4l2, encoded with FFmpeg, video file are put into an array and transmitted. And at decoder side, video data received to an array, decoded and play. The program works fine with codec_id AV_CODEC_ID_MPEG1VIDEO, but when I try replace it with AV_CODEC_ID_H264, some problem happen in decoding, please refer to FFmpeg c++ H264 decoding error. Some people suggest me to use parser but I have no idea how is a parse in ffmpeg looks like. Any simple example of how to build a parser for H.264 in FFmepg? I cannot find such tutorial in google.....
I'm trying to use FFMpeg to create a video. So far i've been playing with a multiplexing example:
http://ffmpeg.org/doxygen/trunk/muxing_8c-source.html, and i'm able to create a compressed video from an already existing video.
Because my program is going to run on an embedded platform I would like to use some custom code (generated by a colleague) to compress the video data and place it into the video file.
So I'm looking for a way to create a video file in c/c++ using ffmpeg in which i have full control over the compression part (to basically circumvent ffmpeg from doing the compression for me and inserting my own code).
To clarify i'm planning to use this to save film from an intelligent camera into a compressed h264 mpeg-4 file.
You could pipe the output with -vcodec rawvideo to your custom program, or write it as a codec and have ffmpeg handle it.
By the way, ffmpeg was superceded by avconv. ffmpeg only exists for backwards compatibility now.
Edit: apparently avconv is a newer fork of ffmpeg, and seems to have more support. Either way, the options are almost the same.
I'm looking for a way to extract the audio part of a FLV file.
I'm recording from the user's microphone and the audio is encoded using the Nellymoser Asao Codec. This is the default codec and there's no way to change this.
ffMpeg is the way to go !
It worked for me with SVN Rev 14277.
The command I used is : ffmpeg -i source.flv -nv -f mp3 destination.mp3
GOTCHA :
If you get this error message : Unsupported audio codec (n),
check the FLV Spec in the Audio Tags section.
ffMpeg can decode n=6 (Nellymoser).
But for n=4 (Nellymoser 8-kHz mono) and n=5 (Nellymoser 16-kHz mono) it doesn't work.
To fix this use the default microphone rate when recording your streams, overwise ffMpeg is unable to decode them.
Hope this helps !
This isn't an exact answer, but some relevant notes I've made from investigating FLV files for a business requirement.
Most FLV audio is encoded in the MP3 format, meaning you can extract it directly from the FLV container. If the FLV was created from someone recording from their microphone, the audio is encoded with the Nellymoser Asao codec, which is proprietary (IIRC).
I'd check out libavcodec, which handles FLV/MP3/Nellymoser natively, and should let you get to the audio.
I'm currently using FFmpeg version SVN-r12665 for this, with no problems (the console version, without any wrapper library). There are some caveats to using console applications from non-console .NET environments, but it's all fairly straightforward. Using the libavcodec DLL directly is much more cumbersome.
I was going to recommend this: http://code.google.com/hosting/takenDown?project=nelly2pcm¬ice=7281.
But its been taken down. Glad I got a copy first :-)