commit a41d46e816d2cbcd93564c42b65a87af547bcb2d
Author: Samuel Thibault <samuel.thibault@ens-lyon.org>
Date:   Sun Oct 11 17:34:57 2020 +0200

    Fix cancellation snappiness
    
    Screen reader users report getting late cancellation, or even mixtures
    of speech. This is because the default buffering parameters of alsa or
    pulseaudio are relatively large.
    
    This change sets alsa and pulseaudio buffer sizes to 10ms worth of audio,
    which is the human snappiness perception limit.

diff --git src/alsa.c src/alsa.c
index 64d5a90..c856788 100644
--- a/src/alsa.c
+++ b/src/alsa.c
@@ -99,6 +99,7 @@ alsa_object_open(struct audio_object *object,
 
 	snd_pcm_hw_params_t *params = NULL;
 	snd_pcm_hw_params_malloc(&params);
+	snd_pcm_uframes_t bufsize = (rate * channels * LATENCY) / 1000;
 
 	int err = 0;
 	if ((err = snd_pcm_open(&self->handle, self->device ? self->device : "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
@@ -113,6 +114,8 @@ 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)
+		goto error;
 	if ((err = snd_pcm_hw_params(self->handle, params)) < 0)
 		goto error;
 	if ((err = snd_pcm_prepare(self->handle)) < 0)
diff --git src/audio_priv.h src/audio_priv.h
index 9526138..669a037 100644
--- a/src/audio_priv.h
+++ b/src/audio_priv.h
@@ -52,6 +52,10 @@ struct audio_object
 	                         int error);
 };
 
+/* We try to aim for 10ms cancelation latency, which will be perceived as
+ * "snappy" by users */
+#define LATENCY 10
+
 #if defined(_WIN32) || defined(_WIN64)
 
 #include <windows.h>
diff --git src/pulseaudio.c src/pulseaudio.c
index d23366d..2f80c62 100644
--- a/src/pulseaudio.c
+++ b/src/pulseaudio.c
@@ -74,6 +74,13 @@ pulseaudio_object_open(struct audio_object *object,
 	}
 
 	int error = 0;
+	pa_buffer_attr battr;
+
+	battr.fragsize = (uint32_t) -1;
+	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;
 	self->s = pa_simple_new(NULL,
 	                        self->application_name,
 	                        PA_STREAM_PLAYBACK,
@@ -81,7 +88,7 @@ pulseaudio_object_open(struct audio_object *object,
 	                        self->description,
 	                        &self->ss,
 	                        NULL,
-	                        NULL,
+	                        &battr,
 	                        &error);
 	return error;
 }
