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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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16
src/fuvst/fuvst.c
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@@ -1259,7 +1259,11 @@ void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length
v27_modem_send(&fuvst->modem, samples, length);
else {
input_num = samplerate_upsample_input_num(&sender->srstate, length);
jitter_load(&sender->dejitter, samples, input_num);
{
int16_t spl[input_num];
jitter_load_samples(&sender->dejitter, (uint8_t *)spl, input_num, sizeof(*spl), jitter_conceal_s16, NULL);
int16_to_samples_speech(samples, spl, input_num);
}
samplerate_upsample(&sender->srstate, samples, input_num, samples, length);
}
}
@@ -1295,7 +1299,7 @@ void sender_receive(sender_t *sender, sample_t *samples, int length, double __at
}
/* 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)
void call_down_audio(void *decoder, void *decoder_priv, int callref, uint16_t sequence, uint8_t marker, uint32_t timestamp, uint32_t ssrc, uint8_t *payload, int payload_len)
{
sender_t *sender;
fuvst_t *fuvst;
@@ -1308,8 +1312,12 @@ void call_down_audio(int callref, uint16_t sequence, uint32_t timestamp, uint32_
if (!sender)
return;
if (fuvst->callref)
jitter_save(&fuvst->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
if (fuvst->callref) {
jitter_frame_t *jf;
jf = jitter_frame_alloc(decoder, decoder_priv, payload, payload_len, marker, sequence, timestamp, ssrc);
if (jf)
jitter_save(&fuvst->sender.dejitter, jf);
}
}
void call_down_clock(void) {}