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Audio rework, new jitter buffer

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Jitter buffer is now based on packets, not on samples. The frames are
dejittered in received form. After reading from jitter buffer, they are
decoded in correct order. If a frame is missing, it is concealed by
repeating audio.
This commit is contained in:
Andreas Eversberg
2024-03-10 13:45:08 +01:00
parent e7efcee289
commit a20637825a
40 changed files with 865 additions and 572 deletions
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22
src/amps/amps.c
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@@ -1080,28 +1080,6 @@ void call_down_release(int callref, int cause)
}
}
/* Receive audio from call instance. */
void call_down_audio(int callref, uint16_t sequence, uint32_t timestamp, uint32_t ssrc, sample_t *samples, int count)
{
sender_t *sender;
amps_t *amps;
for (sender = sender_head; sender; sender = sender->next) {
amps = (amps_t *) sender;
if (amps->trans_list && amps->trans_list->callref == callref)
break;
}
if (!sender)
return;
if (amps->dsp_mode == DSP_MODE_AUDIO_RX_AUDIO_TX) {
compress_audio(&amps->cstate, samples, count);
jitter_save(&amps->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
}
}
void call_down_clock(void) {}
/* Timeout handling */
void transaction_timeout(void *data)
{