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.
Related
I have some aac raw data from an mp4 file which does not have moov header. I want to know the number of samples in this data and size of each sample. I have the decoder specific info as well.
After searching I have found out that aac samples in m4 files does not have ADTS header, they are just CPE/SCE. I want to know the size of each CPE/SCE.
For what I know AAC frames does not encode their own size so usually the framing is provided by a container. That said with some luck and some heuristics maybe you can guess the start of each frame, e.g. looks for 0x01 (SCE with instance tag 0 and part of global gain field) or 0x2? (CPE) and usually the samples don't very that much is size.
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 use ffmpeg to convert videos from one format to another.
Is bitrate the only parameter which decides the output size of a video/audio file?
Yes, bitrate is essentially what will control the file size (for a given playback duration). It is the number of bits used to represent each second of material.
However, there are some subtleties, e.g. :
a video file encoded at a certain video bitrate probably contains a separate audio stream, with a separately-specified bitrate
most file formats will contain some metadata that won't be counted towards the basic video stream bitrate
sometimes the algorithm will not actually aim to achieve the specified bitrate - for example, using the CRF factor. http://trac.ffmpeg.org/wiki/x264EncodingGuide explains how two-pass would be preferred if targeting a specific file size.
So you may want to do a little experimenting with a particular set of options for a particular file format.
Bitrate describes the quality of an audio or video file.
For example, an MP3 audio file that is compressed at 192 Kbps will have a greater dynamic range and may sound slightly more clear than the same audio file compressed at 128 Kbps. This is because more bits are used to represent the audio data for each second of playback.
Similarly, a video file that is compressed at 3000 Kbps will look better than the same file compressed at 1000 Kbps. Just like the quality of an image is measured in resolution, the quality of an audio or video file is measured by the bitrate.
I have AAC-LC audio stream coming directly from audio encoder.
Its a raw stream, No ADTS headers, no container data as I want to stream encoded audio directly as it arrives.(before file gets saved).
I want to determine the frame boundaries/frame lengths/packets lengths in incoming encoded raw AAC stream. (AAC has variable packet lengths.)
Can I search for any fixed frame headers/patterns so that I can determine frame boundaries?
Is it possible with AAC?
Thanks in advance for your valuable inputs.
If you are taking AAC encoded data directly from encoder then it's up to encoder to send frame by frame. It should not send "packets", but single frames. Otherwise I don't see a way you can parse for frames.
I'd first check if it really sends more than one frame at a time?
If yes, then one solution would be to tell encoder to send ADTS header, then parse info from ADTS, and finally strip down ADTS from the frame and stream it as raw.
Does that help?
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