CODEWITHSUNDEEP
ioscore-audio
Julia K.
Julia K.

Reputation: 123

iOS: Playing RTP packets using audio unit

I am developing VoIP app and need to play data from RTP packets which are sent by server every 20 ms.

I have a buffer which accumulates samples from RTP packets. Audio unit render callback reads data from this buffer.

The problem is that I cannot synchronise audio unit with RTP stream. Preferred IO buffer duration cannot be set to exactly 20 ms. And number of frames requested by render callback also cannot be set to the packet's number of samples.

As a result, there are two possible situations (depending on sample rate and IO buffer duration):

a) audio unit reads from my buffer faster than it is filled from RTP packets; in this case buffer periodically doesn't contain the requested number of samples and I get distorted sound;

b) buffer is filled faster than audio unit reads from it; in this case buffer periodically is overflowed and samples from new RTP packets are lost.

What should I do to avoid this issue?

Upvotes: 4

Views: 481

Answers (2)

hotpaw2
hotpaw2

Reputation: 70663

If you have control over the packet rate, this is typically done via a "leaky bucket" algorithm. A circular FIFO/buffer can hold the "bucket" of incoming data, and a certain amount of padding needs to be kept in the FIFO/buffer to cover variations in network rate and latency. If the bucket gets too full, you ask the packet sender to slow down, etc.

On the audio playback end, various audio concealment methods (PSOLA time-pitch modification, etc.) can be used to slightly stretch or shrink the data to fit, if adequate buffer fill thresholds are exceeded.

Upvotes: 1

Coder-256
Coder-256

Reputation: 5618

If you are receiving audio

Try having the client automatically request periodically (ex. every second) that the server sends audio of a certain bitrate dependent on the buffer size and connection speed.

For example, have each audio sample be 300kbits large if there are, for example, 20 samples in the buffer and a 15000kbit/s speed and increase/decrease the audio sample bitrate dynamically as necessary.

If you are sending audio

Do the same, but in reverse. Have the server request periodically that the client changes the audio bitrate.

Upvotes: 1

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