audiobitrate
In a MP3 audio file, I know theres a bitrate of audio which tells the number of bits in seconds are being used by the player, in other words a measure of quality of audio.
But I'm confused with the terms sample rate and sample size. Are they independent of the bitrate or quality of sound? or Can it be explained in some understandable terms?
This is a WMV file with WMA audio. You should be able to play the file just fine using only Windows Media Player and Microsoft pre-installed windows codecs. It's possible that you've downloaded and installed some other codec which has hijacked WMA from MS's codec, and that might not be fun to debug. What I usually do is using Media Player Classic, go into the options. Go to Internal Filters. Uncheck WMV1/2/3 in the Transform Filters side. Then, Play your video. While the video is playing, right-click on the video, and click on the "Filters" submenu. You should see a few items listed. There will usually be 1 source filter (seen as the file name), 1 Audio Output filter (usually Default Direct Sound Device), 1 Video Output filter (usually something like Video Renderer or VMR9 or Overlay Mixer). And possibly an Audio Switcher. This leaves 2 other filters responsible for actually processing the video and Audio codecs. For a WMV/WMA file, you should see WMVideo Decoder DMO and WMAudio Decoder DMO. These are the MS filters. If you see anything else, that's probably your problem.
If they are there correctly. Then it's time to look into driver issues. But there's one more thing to test while we are in Media Player Classic. Go into the options and into the Output settings page. At the bottom is a dropdown labeled DirectShow Audio. Switch it to Default WaveOut Device and try playing the video. (Close the file that is playing before testing. Changing these settings while playing a video will have no effect. Or simply close MPC and reopen it.) This changes the output device. If you hear any difference at all, then there's probably something wierd going on with one of your drivers. Some sound cards have strange audio effects that you can set, and that's probably what is going on.
AAC has different ratios against MP3, so thats good question to ask. AAC has slightly better quality than MP3 in higher bitrates, and is superior to in lower bitrates.
Sample rate: You should keep sample ratio, as resampling looses some quality and up-sample wouldnt do any good.
Bitrate: If you like to have quality comparable to MP3, choose lower when using AAC. For example, for 320kbps MP3 like, I would choose 192 to about 226. This doesn apply to speech in which AAC has better compression ratio, hence same bitrate would yield higher quality.
For higher quality (not highest) movies generally rippers use 192kbps AAC. That would suit good even movies with lots of music and background sounds and effects. For general use and mediocre DVD films 128kbps would do the thing. Regular person should choose 128kbps, audiophile or movie lover may choose slightly more.
Best Answer
Here is a great article on all three terms you're asking about. To summarize, here are the three definitions:
So at this point, we have to do some math and see that bit rate is bits per second (usually measured in Mbits/seconds), so bit rate = sample rate x bit depth. As far as I know, your sample size would simply be one of these one-second chunks of data.
Doing the pure math will reveal that these files would be very large, however there are a number of compression algorithms employed to keep the files lower without losing significant quality.