Is GSM6.10 audio format block or stream based?

Question

I might be asking the wrong question, but my knowledge in this area is very limited.

I'm using acmStreamConvert to convert PCM to GSM (6.10).

Audio Format: 8khz, 16-bit, mono

For the PCM buffer size I'm using 640 bytes (320 samples). For GSM buffer I'm using 65 bytes. My understanding is that GSM "always" converts 320 samples to 65 bytes.

The reason I ask "block or stream" is I'm wondering if I can safely convert multiple audio streams (real-time) using the same acmStreamConvert handle? I see the function has some flags for ACM_STREAMCONVERTF_START and ACM_STREAMCONVERTF_END and ACM_STREAMCONVERTF_BLOCKALIGN, but is it required I use this start/end sequence for GSM? I understand that might be required for some formats that use head/tails, but I'm hoping this isn't required for GSM format?

I'm working on a group VOIP client, and each client sends GSM format, and then needs to convert to PCM before playing. I'm hoping I don't need one ACM handle per client.

Answer 1

Stream based, or at least the ACM API usage of it is. Trying to use the same ACM objects/handles for multiple streams will produce undesired results. I suspect this also means it doesn't handle lost packets as well as other codecs might (haven't confirmed that part yet).