From febd081202ce0d64c6698d11f17fa14a93d84d17 Mon Sep 17 00:00:00 2001
From: Geoffrey McRae <geoff@hostfission.com>
Date: Fri, 28 Jan 2022 12:11:56 +1100
Subject: [PATCH] [client] audio: tune the target latency based on the latency
jitter
---
client/src/audio.c | 40 +++++++++++++++++++++++++++++++++++-----
1 file changed, 35 insertions(+), 5 deletions(-)
diff --git a/client/src/audio.c b/client/src/audio.c
index 827ba38d..69a58400 100644
--- a/client/src/audio.c
+++ b/client/src/audio.c
@@ -102,6 +102,7 @@ typedef struct
RingBuffer timings;
GraphHandle graph;
+ float jitter;
// These two structs contain data specifically for use in the device and
// Spice data threads respectively. Keep them on separate cache lines to
@@ -172,8 +173,8 @@ static const char * audioGraphFormatFn(const char * name,
{
static char title[64];
snprintf(title, sizeof(title),
- "%s: min:%4.2f max:%4.2f avg:%4.2f now:%4.2f",
- name, min, max, avg, last);
+ "%s: min:%4.2f max:%4.2f avg:%4.2f now:%4.2f jitter:%4.2f",
+ name, min, max, avg, last, max - min);
return title;
}
@@ -305,6 +306,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.sampleRate = sampleRate;
audio.playback.stride = channels * sizeof(float);
audio.playback.state = STREAM_STATE_SETUP;
+ audio.playback.jitter = 60.0f; //assume 60ms of jitter initially
audio.playback.deviceData.periodFrames = 0;
audio.playback.deviceData.nextPosition = 0;
@@ -329,8 +331,10 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
if (audio.audioDev->playback.mute)
audio.audioDev->playback.mute(audio.playback.mute);
- // if the audio dev can report it's latency setup a timing graph
- audio.playback.timings = ringbuffer_new(1200, sizeof(float));
+ // timings for jitter calculations and display graph
+ // spice operates on a period size of (sampleRate / 100), so allocate enough
+ // room for 4 seconds of timing samples.
+ audio.playback.timings = ringbuffer_new(sampleRate / 100, sizeof(float));
audio.playback.graph = app_registerGraph("PLAYBACK",
audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
@@ -375,6 +379,22 @@ void audio_playbackMute(bool mute)
audio.audioDev->playback.mute(mute);
}
+static bool getMinMax(int index, void * value, void * udata)
+{
+ float ms = *(float *)value;
+ float * minMax = (float *)udata;
+
+ if (index == 0)
+ {
+ minMax[0] = minMax[1] = ms;
+ return true;
+ }
+
+ minMax[0] = min(minMax[0], ms);
+ minMax[1] = max(minMax[1], ms);
+ return true;
+}
+
void audio_playbackData(uint8_t * data, size_t size)
{
if (!audio.audioDev || size == 0)
@@ -498,7 +518,7 @@ void audio_playbackData(uint8_t * data, size_t size)
// device period size, but that would result in underruns if the period size
// suddenly increases. It may be better instead to just reduce the maximum
// latency on the audio devices, which currently is set quite high
- int targetLatencyMs = 70;
+ int targetLatencyMs = ceil(audio.playback.jitter);
int targetLatencyFrames =
targetLatencyMs * audio.playback.sampleRate / 1000;
@@ -565,6 +585,16 @@ void audio_playbackData(uint8_t * data, size_t size)
const float latency = latencyFrames /
(float)(audio.playback.sampleRate / 1000);
ringbuffer_push(audio.playback.timings, &latency);
+
+ // if the ringbuffer is full calculate the jitter
+ if (ringbuffer_getCount(audio.playback.timings) ==
+ ringbuffer_getLength(audio.playback.timings))
+ {
+ float minMax[2];
+ ringbuffer_forEach(audio.playback.timings, getMinMax, minMax, false);
+ audio.playback.jitter = minMax[1] - minMax[0];
+ }
+
app_invalidateGraph(audio.playback.graph);
}