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telecom digital link has a propagation delay of 3 ms in each direction. This is comparable to having a conversation with someone standing one meter away. It is obvious that even over a very short distance, the use of a codec with a long processing delay characteristic will have a dramatic effect on operation.
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Bit Rate and Compression Ratio
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The ITU has recommended the following bit rates when incorporating data compression in an audio chain [15]:
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These markers place a 4:1 compression ratio at the safe end in the scale. However, more aggressive compression ratios, currently up to a nominal 20:1, are available. Keep in mind, though, that low bit rate, high-level compression can lead to problems many stages of compression are required or anticipated. With successive stages of compression, either or both the noise floor and the audio bandwidth will be set by the stage operating at the lowest bit rate. It is, therefore, worth emphasizing that after these platform have been set by a low bit rate stage, they cannot be subsequently improved by using a following stage operating at a higher bit rate. Bit Rate Mismatch. A stage of compression may well be followed in the audio chain by another digital stage, either of compression or linear, but more importantly, operating at a different sampling frequency [15]. If a D/A conversion is to be avoided, a sample rate converter must be used. This can be a stand-alone unit or it may already be installed as a hardware or software module in existing equipment. Where a following stage of compression is operating at the same sampling frequency but a different compression ratio, the bit resolution will change by default. If the stages have the same sampling frequencies, a direct PCM or AES/EBU digital link can be made, thus avoiding the conversion to the analog domain.
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128 kilobits per second per mono channel (256 kilobits per second for stereo) as the minimum bit rate for any stage if further compression is anticipated or required. 192 kilobits per second per mono channel (384 kilobits per second for stereo) as the minimum bit rate for the first stage of compression in a complex audio chain.
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Appendix E
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The linear PCM waveform associated with standard audio workstations is only useful if decoded [15]. The resolution of the compressed data may or may not be adequate to allow direct editing of the audio signal. The minimum audio sample that can be removed or edited from a transformcoded signal will be determined by the size of the time block of the PCM signal being analyzed. The larger the time block, the more difficult the editing of the compressed data becomes.
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Audio Compression Schemes Important to Streaming
Subband APCM coding has found numerous applications in the professional audio industry and the streaming media industry. We will examine ISO/MPEG-1 Layer 2 (MUSICAM by another name) and ISO/MPEG-1 Layer 3 (the popular MP3 format used by such applications as Winamp, MusicMatch, Napster, and leading streaming media formats). We will also examine Dolby AC-3, which is used in DVD players and the ATSC DTV system, and finally, the MPEG AAC system favored by MPEG-4 streaming media implementations. ISO/MPEG-1 Layer 2 (MUSICAM). This algorithm differs from Layer 1 by adopting more accurate quantizing procedures and by additionally removing redundancy and irrelevancy on the generated scale factors [15]. The ISO/MPEG-1 Layer 2 scheme operates on a block of 1152 PCM samples, which at 48 kHz sampling represents a 24 ms time block of the input audio signal. Simplified block diagrams of the encoding/decoding systems are given in Figure E.16. The incoming linear PCM signal block is divided into 32 equally spaced subbands using a polyphase analysis filter bank (Figure E.16a). At 48 kHz sampling, this equates to the bandwidth of each subband being 750 Hz. The bit allocation for the requantizing of these subband samples is then dynamically controlled by information derived from analyzing the audio signal, measured against a preset psychoacoustic model. The filter bank, which displays manageable delay and minimal complexity, optimally adapts each block of audio to balance between the effects of temporal masking and inaudible pre-echoes.
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