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Refactoring jitter buffer

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Features are:
 * Packet based buffer
 * Random in, first out
 * Adaptive delay compensation (voice)
 * Fixed delay (data, optionally MODEM/FAX)
 * Interpolation of missing frames
 * Any sample size
This commit is contained in:
Andreas Eversberg
2022-07-31 07:55:14 +02:00
parent 4fc92eba45
commit 2b7efedc48
42 changed files with 544 additions and 206 deletions
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6
src/nmt/dsp.c
View File

@@ -446,7 +446,7 @@ static void dial_tone(nmt_t *nmt, sample_t *samples, int length)
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
{
nmt_t *nmt = (nmt_t *) sender;
int count;
int count, input_num;
memset(power, 1, length);
@@ -454,7 +454,9 @@ again:
switch (nmt->dsp_mode) {
case DSP_MODE_AUDIO:
case DSP_MODE_DTMF:
jitter_load(&nmt->sender.dejitter, samples, length);
input_num = samplerate_upsample_input_num(&sender->srstate, length);
jitter_load(&sender->dejitter, samples, input_num);
samplerate_upsample(&sender->srstate, samples, input_num, samples, length);
/* send after dejitter, so audio is flushed */
if (nmt->dms.tx_frame_valid) {
fsk_mod_send(&nmt->fsk_mod, samples, length, 0);

6
src/nmt/nmt.c
View File

@@ -1954,7 +1954,7 @@ void call_down_release(int callref, int __attribute__((unused)) cause)
}
/* Receive audio from call instance. */
void call_down_audio(int callref, sample_t *samples, int count)
void call_down_audio(int callref, uint16_t sequence, uint32_t timestamp, uint32_t ssrc, sample_t *samples, int count)
{
transaction_t *trans;
nmt_t *nmt;
@@ -1967,11 +1967,9 @@ void call_down_audio(int callref, sample_t *samples, int count)
return;
if (nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF) {
sample_t up[(int)((double)count * nmt->sender.srstate.factor + 0.5) + 10];
if (nmt->compandor)
compress_audio(&nmt->cstate, samples, count);
count = samplerate_upsample(&nmt->sender.srstate, samples, count, up);
jitter_save(&nmt->sender.dejitter, up, count);
jitter_save(&nmt->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
}
}