Refactoring jitter buffer
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
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+3
-6
@@ -513,7 +513,7 @@ void call_down_release(int callref, __attribute__((unused)) int cause)
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}
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/* Receive audio from call instance. */
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void call_down_audio(int callref, sample_t *samples, int count)
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void call_down_audio(int callref, uint16_t sequence, uint32_t timestamp, uint32_t ssrc, sample_t *samples, int count)
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{
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sender_t *sender;
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anetz_t *anetz;
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@@ -526,11 +526,8 @@ void call_down_audio(int callref, sample_t *samples, int count)
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if (!sender)
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return;
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if (anetz->dsp_mode == DSP_MODE_AUDIO) {
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sample_t up[(int)((double)count * anetz->sender.srstate.factor + 0.5) + 10];
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count = samplerate_upsample(&anetz->sender.srstate, samples, count, up);
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jitter_save(&anetz->sender.dejitter, up, count);
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}
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if (anetz->dsp_mode == DSP_MODE_AUDIO)
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jitter_save(&anetz->sender.dejitter, samples, count, 1, sequence, timestamp, ssrc);
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}
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void call_down_clock(void) {}
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