Currently, I am developing a ucma 3.0 vxml application. In the voicexml document , I use a record element to take the caller's recording, and then send the recording to an exteranl web server and save it to a wav file.
whats the format of the saved wav file, 8KHz/8-bit or 8KHz/16-bit or 16KHz/16-bit?
Can I set or change the audio format of the ucma vxml Browser to generate audio with different format?
They supported audio formats for UCMA VoiceXML are:
Raw (headerless) 8kHz 8-bit mono mu-law [PCM] single channel. (G.711) -- audio/basic (from [RFC1521])
Raw (headerless) 8kHz 8 bit mono A-law [PCM] single channel. (G.711) -- audio/x-alaw-basic
WAV (RIFF header) 8kHz 8-bit mono mu-law [PCM] single channel. -- audio/x-wav
WAV (RIFF header) 8kHz 8-bit mono A-law [PCM] single channel. -- audio/x-wav
The part after the "--" is the mime type. You specify the mime type in the "type" attribute of the "record" element.
Related
Looking at the article :http://www.streamingmedia.com/Articles/Editorial/What-Is-.../What-is-MPEG-DASH-79041.aspx
And it makes statements like:DASH is codec-independent, and will work with H.264, WebM and other codecs
DASH supports both the ISO Base Media File Format (essentially the MP4 format) and MPEG-2 transport streams
DASH does not specify a DRM method but supports all DRM techniques specified in ISO/IEC 23001-7: Common Encryption
But how is audio/video compression, or DRM method is specified in Media Presentation? Where cab i find more details?
DASH is a streaming protocol - the video stream is inside a 'container' and the container is broken into chunks and streamed. A very high level view of the video component is:
elementary video stream encoded with some codec
fragmented mp4 container (broken into chunks to facilitate ABR)
MPEG DASH streaming protocol
The mp4 container header information contains information about all the streams it contains - this will include the codec that it used to encode the stream (e.g. h.264 for a video stream).
ABR essentially allows the client device or player download the video in chunks, e.g 10 second chunks, and select the next chunk from the bit rate most appropriate to the current network conditions.
The DASH manifest (essentially an index file that contains pointers to the different bit rate streams etc) contains header information about the protections systems in use, for example Widevine or PlayReady DRMs.
The mp4 container also contains information about the protection system in a special PSSH (Protection System Specific Headers) header for the protection systems in use, for example again, Widevine or PlayReady.
Generally DASH streams will have the protection information in both places to ensure that all players can play the stream, but last time I looked, I think the spec strictly speaking says it can be in either or both.
The specs themselves are available here:
http://standards.iso.org/ittf/PubliclyAvailableStandards/index.html (search for DASH)
https://www.iso.org/standard/68042.html - unfortunately, this one requires payment AFAIK. You can see a W3C spec which uses it here, however: https://w3c.github.io/encrypted-media/format-registry/stream/mp4.html
And there is a nice overview of DASH here:
https://www.w3.org/2011/09/webtv/slides/W3C-Workshop.pdf
And, of course, the classic reference to some of the drivers for DASH and similar standards:
https://xkcd.com/927/
I am writing a code to extract AAC audio data from mpeg ts stream. I want to get stream properties like sampling frequency, number of channels, Audio type, Audio profile type etc. from Transport stream, without decoding the actual data. How much of the information will be available from stream?
Also I want to know is there any way to find the total duration of the stream without actually finding the last PTS value in the file
Thanks
AAC frames packed in TS use ADTS headers. Its 7 (or 9) bytes, and very easy to parse. ADTS header format is documented well online.
I have a text file contaning payload of RTP packets (in hex,coded with GSM\ opus\speex ) belong to a VoIP conversation, does anyone know how to convert this file into a .wav audio file?
I'm using windows.
Thanks
.wav is just an file container which you can have an any codec format and makes the player to recognize the codec inside. Refer Wiki: WAV and for more technical details refer WaveFormat. And It just wrappers the raw codec content. If you have experience in C program, there are open source available to convert one codec to PCM. Since PCM is the raw audio data in 16 bit format.
But I suggest an solution but don't know it will met your requirement.
install latest wireshark
using wireshark capture the RTP packets.
Select UDP packet and Right click and choose Decode As option
Select Transport tab and choose RTP protocol
Now you can see RTP packet with right codec
Go to Telephony -> RTP -> Stream analysis -> Save the RTP Payload as .raw
At this stage the codec data is available in .raw file format.
There are open software are available are available such as SoX, ffmpeg etc...
From there are you can covert .raw to .wav format.
After that you can play in VLC (PCM, GSM, ADPM, Alaw , uLaw) or any other supported player (Amr)
You won't find Speex, g729 since these are paid
I am trying to capture the video and audio using AVCaptureSession and I done with videocapturing and converted into pixel buffer and I played the output captured video at server side using ffmpeg n rtmp server. But the thing is how can I make the audio to be converted info data and play it at sever side where the data received. And want to know what the audio format is the audio that is captured.
Thank's All,
MONISH
I need to find the codec of an audio file. How can I do this?
Do I need to write code to do this or is there a simpler way?
Please help me. If possible share helpful links.
The good old file utility will reveal lots of information about audio files, sometimes including the codec:
$ file X.wav
X.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
#bhebsquines
"It is important to distinguish between a file format and an audio codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Although most audio file formats support only one type of audio data (created with an audio coder), a multimedia container format (as Matroska or AVI) may support multiple types of audio and video data." - http://en.wikipedia.org/wiki/Audio_file_format
The application gspot does a good job of pulling codecs from audio and video files.
http://www.headbands.com/gspot/
run it and drag a file into the window. It will pull all of the data from there. Note that some audio files will not display a codec as they are made from "Raw" audio.
You can identify your codec by extenion name itself example file1audio.mp3 or fileaudio.avi, .mp3, .avi will be your file type or codec, you can use k-lite codec pack for your different type audio format. use this link http://www.free-codecs.com/download/k_lite_codec_pack.htm