Table of Recommended Bitrates for use with psychoacoustic audio compression systems

Table of Recommended Bitrates for use with psycho acoustic audio compression systems

NOTE : The audio source here is understood to be derived from
 Compact Disks
 FM Radio, (analogue or digital) TV audio, Analogue Tape & Long Play Records (LPs) Speech sources
Raw bit rates (32 kbs, 56 kbs, 192 kbs etc...) as a measure of the quality of encoded phycoacoustically compressed audio are a poor source of reliable information.
 The sampling rate of the incoming audio can be at (8000 Hz, 22050 Hz, 32 kHz,  44.1 kHz, 48 kHz, 96 kHz etc ...). One can also re sample from one digitization rate to another (48 kHz to 44.1 kHz is quite common).  The incoming audio itself may be encoded at bit depths of either 8 bits, 16 bits, 24 bits or 32 bit integers -as well as 32 bit floating point. In some cases the incoming sampling system may be signed or unsigned, as with mu-Law or A-Law sources related to telephony.
 Even digital origin sources may have been thru a phycoacoustical encoding stage at some earlier point, leading to generational coding issues. V ariable Bitrate Coding (VBR) can also produce quite varied audio output quality if one does not take care to learn the behaviours of the encoder.
 Magnetic tape and vinyl audio sources have tape hiss and source rumble -- as well as wow and flutter. A sampling rate that is too low coupled with a bitrate that is too low can lead to these noise sources becoming overpowering. Analogue recording artefacts in general can produce substantial distortion if not factored into the digitization process.
 Analogue sources, namely 16 mm and 32 mm film and Dolby encoded magnetic tape can produce dynamic range compression and failed decompression artefacts if not put thru the proper decoder.
Source : CDs and DVDs (True Stereo [DDD], 20hz-20khz) Compression Format Musicam (MP2)

Speech sources : MW Radio, Talking Books (Parametric Stereo, [AAA], 120hz-12khz) Compression Changes in the technology worth noting ...
 The most likely next-generation codec for widespread use is HEVC (high efficiency video coding), which is also known by its ITU designation of H.265. Keep in mind that H.265 is a video codec, and not an audio codec.
 The audio encoding that will accompany HEVC is being worked on by different teams than those working on HEVC/H.265. There is a competitive compression standard being developed by Google called VP9, which will be built into many Web browsers. Available with no royalty payments, Google’s vision for VP9 is that it will have better performance both in terms of encoding efficiency and image quality as compared to HEVC/H.265.
 Still, H.265 looks to be where professional and broadcast video are going in the next couple of years, despite the fact that royalty payments are associated with the standard. There is yet another next-generation video codec on the near horizon, called Daala, being developed by the Xiph.Org Foundation and Mozilla Corp. The founder of Xiph.Org has stated that the performance of Daala should be a generation beyond HEVC and VP9, but an initial release is not expected in 2015.
 Interestingly, the Xiph.Org Foundation is the creator of FLAC (free lossless audio codec), which is well-regarded for its audio performance.
TWICE AS EFFICIENT From a video standpoint, H.265 is roughly twice as efficient as H.264, which was itself about twice as efficient as MPEG-2. In other words, a video stream encoded at 20 Mbps with MPEG-2 would require about 10 Mbps with H.264 and about 5 Mbps with H.265. That’s a little oversimplified, but it’s a useful rule of thumb. MPEG-2 introduced most of us to the term MP3 for audio coding. Unveiled with MPEG-1 compression in the early 1990s, MP3 stands for MPEG Audio Layer III. It has become a popular audio compression standard, although there are many others also in use simultaneously. Like the parent video compression standard, MP3 is a “lossy” form of compression, meaning that it changes the audio in order to achieve its compression and those changes can’t be undone once they are compressed.
MP3 has a wide range of settings that have an effect on the final audio quality, including sampling rate and bit-rate.
 Mainstream MP3 can be sampled at 32, 44.1 and 48 kHz rates, and can be encoded at bit-rates ranging from 56 to 384 kbps.
 At 128 kbps and 44.1 kHz sampling, an MP3 file takes up about 9.1 percent of an uncompressed CD recording.
 Encoding an MP3 file at a flat bit-rate of 320 kbps will create a bitstream that’s about 23 percent the size of an uncompressed CD recording. Advanced audio coding (AAC) was developed after MP3 and takes advantage of what was learned from that initial popular format. AAC generally provides better sound quality than MP3 at similar bitrates. AAC also has an offshoot known as high efficiency advanced audio coding (HE-AAC), which is used for mobile television standards such as DVB-H and ATSC-M/H. Like MP3, AAC is a lossy compression format, and it has a range of settings similar to MP3. Dolby Digital and AC-3 are two names for the same format of audio processing. Developed by Dolby Laboratories, AC-3 is sometimes referred to as “audio codec 3” or “advanced codec 3.” All forms of AC-3 support surround sound, with the initial version carrying 5.1 channels and the later Dolby Digital Plus handling 7.1 channels.
 An enhancement to Dolby Digital Plus called E-AC-3 can carry up to 13.1 channels. The greater coding efficiency of E-AC-3 means that it can provide reasonable 5.1-channel audio in a 256 kbps stream.
NEXT-GEN AUDIO FORMATS The primary audio coding formats associated with HEVC/H.265 are MPEG-H and AC-4, and may also include other codecs over the ensuing months.
 MPEG-H can be thought of as “AAC on steroids,” and last year the ATSC announced that MPEG-H 3D audio was one of the three standards proposed for the audio system of ATSC 3.0.
 In its simplest form, MPEG-H will support eight channels of audio. MPEG-H has many other features, including the ability to provide loudness metadata. Dolby AC-4 is likewise a considerably advanced codec that evolved from AC-3.
 Compared to AC-3, AC-4 improves compression efficiency for broadcast by about 50 percent.
 AC-4 is already standardized with the European Telecommunications Standards Institute and adopted by the Digital Video Broadcasting Project, the UK standards body.
 The standard features native support for Dialogue Enhancement, Intelligent Loudness and Advanced Dynamic Range Control, as well as more efficient support for multiple languages and descriptive services. The interaction of these audio codecs with HEVC is still being worked out and will become part of a final ATSC standard here in the United States. At the recent ATSC Boot Camp in Washington DC, Jim Starzynski of US-NBC gave a presentation on the status of MPEG-H and what we can expect in the future.
Like video codecs, audio codecs are becoming more efficient at compressing audio into smaller streams.
 The enhanced compression features that have evolved will enable future broadcast codecs to provide more channels of audio and provide broadcasters with more choice regarding quantity - vs - quality trade-offs.