Is the endianness of format params guaranteed in RIFF WAV files? I have heard conflicting answers to this including references to a RIFX file format.
Yes.
If the file starts with RIFF, then it's little endian.
If it starts with FFIR or RIFX, then it's probably not.
Generally, supporting the WAV format means supporting RIFF files, although adding RIFX support should not prove difficult.
The AES31 specification for BWF (Broadcast Wave Format) references this specification for RIFF: http://www.tactilemedia.com/info/MCI_Control_Info.html
From this:
RIFF has a counterpart, RIFX, that is
used to define RIFF file formats that
use the Motorola integer byte-ordering
format rather than the Intel format. A
RIFX file is the same as a RIFF file,
except that the first four bytes are
'RIFX' instead of 'RIFF', and integer
byte ordering is represented in
Motorola format.
The KVR article you reference refers to the author's incorrect usage of int* on big-endian systems (Motorola PPC), which will retrieve byte-swapped values. Care is always necessary to handle byte-swapping correctly when writing cross-endian code.
If the zeroth byte of the file on disk is 'R', and the 3rd is 'F', you can be certain that the rest of the contents will be stored little endian.
RIFX is not widely used. Most big-endian RIFF implementations swap bytes to disk, and generate correct RIFF files.
Related
I have several pcap files which captured by wireshark.
I collected that data from pcap file, payload data.
and saved it as file.
I tried to convert it to pcm data with this link
https://github.com/ToughmanL/EVS_CODEC
but it seems the data I collected is not EVS format.
please let me know how to get EVS codec from pcap file.
My understanding is that Stack Overflow is for programming questions, so here is an answer based on existing / example source code. First, in your screen cap Wireshark has found a commonly used form of EVS; the 61 byte packet payload length matches the EVS spec for 24400 bps primary mode, using compact format, which means (i) no ToC header byte in the payload, and (ii) only the payload size determines the bitrate. Second, to feed that to the 3GPP decoder you have to create a file with a MIME header and format, which is needed by the 3GPP reference program.
x86_mediatest.cpp (under the mediaTest folder in the Docker containers here) does this - it reads a pcap record-by-record, pulls out RTP payloads and writes them to a .cod file (coded format), including the MIME header. Running the 3GPP program on that file, with appropriate command line options then gives a PCM raw audio file you can convert to wav using sox, Audacity, Hypersignal, etc.
In x86_mediaTest.cpp, look for "if (pcap_extract) {". Also note the comment "If pcap RTP payloads are in CH (Compact Header) format, they are converted to FH (Header-Full) format by adding a ToC payload header byte". Which may explain why in your case, if you directly copied the compact header format payloads and fed them to the 3GPP decoder, it didn't work.
There is also a mediaMin program that will take the pcap on the command line and do the whole thing, generating a .wav file. You can use that for comparison and audio quality testing vs your code.
Disclaimer: I work for the company that created the mediaTest and mediaMin tools
In audio terms there is no difference between AIF and WAV because they're both uncompressed audio. The only difference is the byte order (endianness).
My question is, can any software tell the difference between an AIF that is recorded as such and an AIF that was recorded WAV and converted? I've looked at a hex editor and there appears to be a difference in the chunks - the recorded AIF has more empty space in the COMM and SSND chunks, it would seem.
Is there a reason for this?
Many Thanks
"...the recorded AIF has more empty space in the COMM and SSND chunks, it would seem."
That might be a problem with the specific recorder you use.
In general there is no size difference in the uncompressed PCM data. I've tested a 10-second AAC file converted into WAVE and also into AIFF, result is both formats have the PCM data at 1572864 bytes long.
Also explain "more empty space in the COMM and SSND chunks" since...
COMM only holds 10 bytes worth of metadata, but in WAV file there'll be up to 84 bytes for metadata.
SSND is 16 bytes followed by PCM data, in .wav the DATA chunk is 8 bytes followed by PCM.
I want relation between time and bytes in ogg file. If I have 5 second ogg and it's length 68*1024 bytes. If I chunk from that ogg file and save it can I knew that size from before chunk? Like I knew it I want to chunk from 2.4 to 3.2.
And give some mathematical calculation and get accurate answer of bytes I can get. Can anyone tell me please if this is possible?
Bit rate 128kbps, 16 bit , sample rate - 44.1Khz, stereo
I used below logic but can't get accurate answer.
Click here
Any such direct mapping between file size and play time will work, but not if the codec uses variable bit rate (vbr) encoding ... meaning the compression algorithm is vbr if its success in compressing is dependent on the informational density of the source media ... repetitive audio is more efficiently compressed than say random noise ... vbr algorithms are typically more efficient since to maintain a constant bit rate the algo pads the buffer with filler data just so its throughput is in constant bytes per second
Can some one tell me the WAV file header information for A-Law and Mu-Law?
In short, the WAVE format is a RIFF specification for mutlimedia files, which uses "chunks" to store data. Similar to a type-length-value representation, each chunk contains information about its own type and length (the fields SubChunkID and SubChunkSize) before the rest of the data.The information about the audio compression is stored in the "fmt" chunk.
Read this and this for more information.
How can we distinguish between PCM and BWF format?
Is it necessary for BWF to have "bext" header?
I have some streams that don't have "bext" header but contains "JUNK" header... Are these files BWF files?
Thanks you.
The JUNK chunk is reserved space to allow a BWF file to be converted into an RF64 file on the fly if the size goes over 4GB. The JUNK chunk is the same size as a ds64 chunk, and will be replaced with a ds64 chunk if the conversion to RF64 is needed. Read more about it here.
My reading of the BWF spec is that you have to have a bext chunk for it to be a BWF.
As far as I know, a broadcast wave file will have the 'bext' header extension.
If a file does not have the 'bext' header extension, it will be a normal WAV/AIFF or whatever file.
Broadcast wave headers are used especially if you want to give a file more information about itself in the header which isn't to be seen immediately from its name.
For playing back, this info isn't necessary to know. Just if you want to show or search the meta information somehow.
PCM isn't a file format. All files that handle uncompressed data are PCM files.
Such as WAV/BWF, AIFF or SD2 for example.
With encoded files like MP3 or AAC you get the raw PCM values after decoding.
Yes. The 'bext' chunk is what distinguishes a BWF file from a wav file.
Some manufacturers actually use '.bwf' as a file extension but mostly the '.wav' extension will be used. It is only the presence of this chunk that makes the difference.
Other chunks can also be present and a well designed player will ignore chunks that it doesn't recognize.
Generally the 'data' chunk containing the audio data will be the last one in the file. However I have seen a few examples of other chunks, usually xml metadata, appearing after the 'data' chunk. This confuses some players.
For more information search for tech3285.pdf from the European Broadcasting Union website (tech.EBU.ch).