77 lines
2.5 KiB
Diff
77 lines
2.5 KiB
Diff
commit a41d46e816d2cbcd93564c42b65a87af547bcb2d
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Author: Samuel Thibault <samuel.thibault@ens-lyon.org>
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Date: Sun Oct 11 17:34:57 2020 +0200
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Fix cancellation snappiness
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Screen reader users report getting late cancellation, or even mixtures
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of speech. This is because the default buffering parameters of alsa or
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pulseaudio are relatively large.
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This change sets alsa and pulseaudio buffer sizes to 10ms worth of audio,
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which is the human snappiness perception limit.
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diff --git src/alsa.c src/alsa.c
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index 64d5a90..c856788 100644
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--- src/alsa.c
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+++ src/alsa.c
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@@ -99,6 +99,7 @@ alsa_object_open(struct audio_object *object,
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snd_pcm_hw_params_t *params = NULL;
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snd_pcm_hw_params_malloc(¶ms);
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+ snd_pcm_uframes_t bufsize = (rate * channels * LATENCY) / 1000;
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int err = 0;
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if ((err = snd_pcm_open(&self->handle, self->device ? self->device : "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
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@@ -113,6 +114,8 @@ alsa_object_open(struct audio_object *object,
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goto error;
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if ((err = snd_pcm_hw_params_set_channels(self->handle, params, channels)) < 0)
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goto error;
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+ if ((err = snd_pcm_hw_params_set_buffer_size_near(self->handle, params, &bufsize)) < 0)
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+ goto error;
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if ((err = snd_pcm_hw_params(self->handle, params)) < 0)
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goto error;
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if ((err = snd_pcm_prepare(self->handle)) < 0)
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diff --git src/audio_priv.h src/audio_priv.h
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index 9526138..669a037 100644
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--- src/audio_priv.h
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+++ src/audio_priv.h
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@@ -52,6 +52,10 @@ struct audio_object
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int error);
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};
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+/* We try to aim for 10ms cancelation latency, which will be perceived as
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+ * "snappy" by users */
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+#define LATENCY 10
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+
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#if defined(_WIN32) || defined(_WIN64)
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#include <windows.h>
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diff --git src/pulseaudio.c src/pulseaudio.c
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index d23366d..2f80c62 100644
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--- src/pulseaudio.c
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+++ src/pulseaudio.c
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@@ -74,6 +74,13 @@ pulseaudio_object_open(struct audio_object *object,
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}
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int error = 0;
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+ pa_buffer_attr battr;
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+
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+ battr.fragsize = (uint32_t) -1;
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+ battr.maxlength = (uint32_t) -1;
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+ battr.minreq = (uint32_t) -1;
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+ battr.prebuf = (uint32_t) -1;
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+ battr.tlength = pa_bytes_per_second(&self->ss) * LATENCY / 1000;
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self->s = pa_simple_new(NULL,
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self->application_name,
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PA_STREAM_PLAYBACK,
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@@ -81,7 +88,7 @@ pulseaudio_object_open(struct audio_object *object,
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self->description,
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&self->ss,
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NULL,
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- NULL,
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+ &battr,
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&error);
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return error;
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}
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