mike-jumper commented on a change in pull request #312:
URL: https://github.com/apache/guacamole-server/pull/312#discussion_r532805321
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File path: src/protocols/rdp/channels/audio-input/audio-buffer.c
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@@ -131,6 +137,19 @@ void guac_rdp_audio_buffer_begin(guac_rdp_audio_buffer*
audio_buffer,
* audio_buffer->out_format.channels
* audio_buffer->out_format.bps;
+ /* Calculate the interval to send each packet in ms.
+ * We need to set half of the time duration designated
+ * by packet frames according to Nyquist's theorem.
+ */
+ audio_buffer->packet_interval = packet_frames
+ * 1000
+ / audio_buffer->out_format.rate
+ / 2;
Review comment:
Sure. That being the case:
* That would be the reasoning to cite here, not Nyquist's theorem.
* This sounds like a workaround for a problem that results from not properly
calculating the send interval in the first place. Rather than use a constant
interval, falling back to a hard-coded 5ms if the interval is sufficiently
small, shouldn't the interval be determined dynamically based on (1) how much
time has elapsed since the last time a packet was sent and (2) how much audio
data is about to be sent?
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