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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/amps/amps.c
View File

@@ -1065,7 +1065,7 @@ void call_down_release(int callref, int 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)
{
sender_t *sender;
amps_t *amps;
@@ -1079,10 +1079,8 @@ void call_down_audio(int callref, sample_t *samples, int count)
return;
if (amps->dsp_mode == DSP_MODE_AUDIO_RX_AUDIO_TX) {
sample_t up[(int)((double)count * amps->sender.srstate.factor + 0.5) + 10];
compress_audio(&amps->cstate, samples, count);
count = samplerate_upsample(&amps->sender.srstate, samples, count, up);
jitter_save(&amps->sender.dejitter, up, count);
jitter_save(&amps->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
}
}

6
src/amps/dsp.c
View File

@@ -473,7 +473,7 @@ static void sat_encode(amps_t *amps, sample_t *samples, int length)
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
{
amps_t *amps = (amps_t *) sender;
int count;
int count, input_num;
again:
switch (amps->dsp_mode) {
@@ -483,7 +483,9 @@ again:
break;
case DSP_MODE_AUDIO_RX_AUDIO_TX:
memset(power, 1, length);
jitter_load(&amps->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);
/* pre-emphasis */
if (amps->pre_emphasis)
pre_emphasis(&amps->estate, samples, length);