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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/fuenf/dsp.c
View File

@@ -595,15 +595,17 @@ void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length
fuenf_t *fuenf = (fuenf_t *) sender;
sample_t *orig_samples = samples;
int orig_length = length;
int count;
int count, input_num;
sample_t *spl;
int pos;
int i;
/* speak through */
if (fuenf->state == FUENF_STATE_DURCHSAGE && fuenf->callref) {
jitter_load(&fuenf->sender.dejitter, samples, length);
memset(power, 1, 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);
} else {
/* send if something has to be sent. else turn transmitter off */
while ((count = encode(fuenf, samples, length))) {

9
src/fuenf/fuenf.c
View File

@@ -390,7 +390,7 @@ 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;
fuenf_t *fuenf;
@@ -403,11 +403,8 @@ void call_down_audio(int callref, sample_t *samples, int count)
if (!sender)
return;
if (fuenf->state == FUENF_STATE_DURCHSAGE) {
sample_t up[(int)((double)count * fuenf->sender.srstate.factor + 0.5) + 10];
count = samplerate_upsample(&fuenf->sender.srstate, samples, count, up);
jitter_save(&fuenf->sender.dejitter, up, count);
}
if (fuenf->state == FUENF_STATE_DURCHSAGE)
jitter_save(&fuenf->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
}
void dump_info(void) {}