How can one avoid data throttling when streaming with an audio codec?

I’m working with an incoming audio stream sampled at 8Kps. My codec encodes 80 samples into a packet (5 samples) in about 1.6ms. Consequently, I get approximately 13 samples per encoding cycle, while I require 80.
How can I ensure a seamless stream?
My initial thought is to reduce the outgoing encode stream’s bitrate, yet I’m unsure about the precise calculation to prevent buffer overflow on the input side and buffer starvation on the output side. This appears to be a fundamental aspect of streaming, but I’m struggling to find relevant information.

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To address your issue, it seems like there’s a mismatch between the rate at which your audio stream is being sampled and the encoding capacity of your codec. Since your codec is encoding at a slower rate than the incoming sample rate, you’re naturally running into a bottleneck.
One approach to consider is buffer management. By implementing a buffer, you can temporarily store incoming audio samples and then feed them to your codec at a rate it can handle.

How would buffer management help in this case, considering the continuous nature of the incoming stream?

The idea behind using a buffer is to balance the incoming and outgoing data rates. The buffer acts as a temporary holding area for the incoming audio samples. While your codec is encoding the data at its own pace, the buffer ensures that the incoming samples are not lost.
However, the size of the buffer and the strategy for managing it are crucial. If the buffer is too small, it might overflow. Conversely, if it’s too large, it might add unwanted latency to your stream.

I understand the concept of buffering, but how do I calculate the optimal buffer size and manage it effectively?

To calculate the optimal buffer size, you need to consider the maximum expected difference between the incoming data rate and the codec’s encoding rate. This can be determined by the sampling rate, the number of samples per encoding cycle, and the codec’s encoding time per packet. A good starting point is to set the buffer size to handle at least a few seconds of audio data.
As for buffer management, you need to implement a system that monitors the buffer’s fill level. When it reaches a certain threshold, you can start encoding the data to prevent overflow. Additionally, you should have a mechanism to handle scenarios where the buffer is underutilized, to avoid data starvation on the receiving end.

Makes sense. But what about reducing the bitrate of the outgoing encoded stream? How would that factor into the solution?

Reducing the bitrate of the outgoing stream is indeed a viable solution. This can be done by either changing the codec settings to produce a smaller output size per cycle or by deliberately reducing the data rate. However, reducing the bitrate often comes with a trade-off in audio quality.
It’s a balance between maintaining a smooth stream and preserving audio fidelity. You’ll need to experiment with different bitrate settings to find a sweet spot where the quality is acceptable, and the stream remains continuous without overwhelming your buffer.
Remember, any adjustments to the bitrate should be aligned with the capabilities of the receiving end to decode and process the stream effectively.