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From 72da4a54a5afbbdadfa6d8131e0f4a9f08cf4394 Mon Sep 17 00:00:00 2001
From: Nicolas Pitre <nico@fluxnic.net>
Date: Wed, 6 Jul 2022 00:30:42 -0400
Subject: [PATCH] Fix latency-related buffer sizing
Turns out that eSpeak-NG (the main user of this lib) enforces a minimum
buffer size of 60ms which is also the default size. This explains why
smaller LATENCY values were inducing choppiness in the audio on some
systems. Adjust the comment accordingly,.
Also make sure computed buffer sizes don't land in the middle of a
sample frame. Doing (samplerate * channels * LATENCY) / 1000 is wrong.
Both ALSA and PulseAudio provide nice abstractions for buffer sizing
so let's use them directly. In the ALSA case in particular, we want the
period to be 60ms, not the whole buffer, so to interleave speech audio
computation and audio playback.
---
src/alsa.c | 5 +++--
src/audio_priv.h | 5 +----
src/pulseaudio.c | 2 +-
3 files changed, 5 insertions(+), 7 deletions(-)
diff --git a/src/alsa.c b/src/alsa.c
index c856788..a0da0f0 100644
--- a/src/alsa.c
+++ b/src/alsa.c
@@ -99,7 +99,8 @@ alsa_object_open(struct audio_object *object,
snd_pcm_hw_params_t *params = NULL;
snd_pcm_hw_params_malloc(¶ms);
- snd_pcm_uframes_t bufsize = (rate * channels * LATENCY) / 1000;
+ unsigned int period_time = LATENCY * 1000;
+ int dir = 0;
int err = 0;
if ((err = snd_pcm_open(&self->handle, self->device ? self->device : "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
@@ -114,7 +115,7 @@ alsa_object_open(struct audio_object *object,
goto error;
if ((err = snd_pcm_hw_params_set_channels(self->handle, params, channels)) < 0)
goto error;
- if ((err = snd_pcm_hw_params_set_buffer_size_near(self->handle, params, &bufsize)) < 0)
+ if ((err = snd_pcm_hw_params_set_period_time_near(self->handle, params, &period_time, &dir)) < 0)
goto error;
if ((err = snd_pcm_hw_params(self->handle, params)) < 0)
goto error;
diff --git a/src/audio_priv.h b/src/audio_priv.h
index 0c2ce3c..dbccb1c 100644
--- a/src/audio_priv.h
+++ b/src/audio_priv.h
@@ -52,10 +52,7 @@ struct audio_object
int error);
};
-/* We try to aim for 10ms cancelation latency, which will be perceived as
- * "snappy" by users. However, some systems (e.g. RPi) do produce chopped
- * audio when this value is smaller than 60.
- */
+/* 60ms is the minimum and default buffer size used by eSpeak */
#define LATENCY 60
#if defined(_WIN32) || defined(_WIN64)
diff --git a/src/pulseaudio.c b/src/pulseaudio.c
index 2f80c62..da6c49f 100644
--- a/src/pulseaudio.c
+++ b/src/pulseaudio.c
@@ -80,7 +80,7 @@ pulseaudio_object_open(struct audio_object *object,
battr.maxlength = (uint32_t) -1;
battr.minreq = (uint32_t) -1;
battr.prebuf = (uint32_t) -1;
- battr.tlength = pa_bytes_per_second(&self->ss) * LATENCY / 1000;
+ battr.tlength = pa_usec_to_bytes(LATENCY * 1000, &self->ss);
self->s = pa_simple_new(NULL,
self->application_name,
PA_STREAM_PLAYBACK,
--
2.35.1
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