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Human speech has many measurable (and therefore predictable) characteristics, one of which is a tendency to have embedded pauses. As a rule, people do not spew out a series of uninterrupted sounds; they tend to pause for emphasis, to collect their thoughts, to reword a phrase while the other person listens quietly on the other end of the line. When speech technicians monitor these pauses, they discover that during considerably more than half of the total connect time, the line is silent. Digital Speech Interpolation (DSI) takes advantage of this characteristic silence to drastically reduce the bandwidth required for a single

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In a streaming app, this means you ll stop receiving incoming bytes for a while but that your network connection should eventually pick up and start sending data again It s during this stall time that your audio buffers are most useful There s no guarantee of how long the stall will last, but in my testing I ve found that small buffers won t be up to the task Planning for one to two seconds of downtime is nowhere near sufficient to avoid audio dropouts Make the buffers as large as you think you can afford A side effect of buffering is that, in the case of a live audio stream, your playback may lag compared to the original audio That s a necessary consequence of planning to keep playback going in changing network situations Listing 3-14 shows the initializer for the streaming class.

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This is where the array of PlayQueueData structures is allocated, although the audio buffers they contain are still NULL at this point They can t be allocated until you know something about the audio format, so you ll leave that until you have that information..

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channel Whereas 24 channels can be transported over a typical T1 facility, DSI allows for as many as 120 conversations to be carried over the same circuit The format is proprietary and requires the setting aside of a certain amount of bandwidth for overhead A form of statistical multiplexing lies at the heart of DSI s functionality The standard T-Carrier is a time-division multiplexed scheme in which channel ownership is assured: A user assigned to channel three will always own channel three, regardless of whether they are actually using the line In DSI, channels are not owned Instead, large numbers of users share a pool of available channels When a user starts to talk, the DSI system assigns an available timeslot to that user and notifies the receiving end of the assignment.

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This system works well when the number of users is large, because statistical probabilities are more accurate and indicative of behavior in larger populations than in smaller ones There is a downside to DSI, of course, and it comes in several forms Competitive clipping occurs when more people start to talk than there are available channels, resulting in someone being unable to talk Connection clipping occurs when the receiving end fails to learn what channel a conversation has been assigned within a reasonable amount of time, resulting in signal loss Two approaches have been created to address these problems In the case of competitive clipping, the system intentionally clips off the front end of the initial word of the second person who speaks.

Listing 3-14. Initializing the Audio Streaming Object - (id)initWithURL:(NSURL *)audioUrl { if (self = [super init]) { url = [audioUrl retain]; playQueueDataRecs = (PlayQueueData_t *)malloc(sizeof(PlayQueueData_t) * kNumAQBufs); } return self; }

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