pcaudiolib: update to 1.2.

This commit is contained in:
Christopher Brannon 2022-11-01 10:08:44 -07:00 committed by classabbyamp
parent 302febbf6b
commit 1c5b5ce14c
3 changed files with 3 additions and 97 deletions

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@ -1,19 +0,0 @@
commit d6a6b00aa4334b461c1a09c27b6c681eaac8da8a
Author: Martin Schreiber <mse00000@gmail.com>
Date: Sat Mar 10 06:55:58 2018 +0100
* Alsa: fixed sample_size calculation, multiply with channel count.
diff --git src/alsa.c src/alsa.c
index 34e39be..6af1941 100644
--- a/src/alsa.c
+++ b/src/alsa.c
@@ -53,7 +53,7 @@ alsa_object_open(struct audio_object *object,
return -EEXIST;
snd_pcm_format_t pcm_format;
-#define FORMAT(srcfmt, dstfmt, size) case srcfmt: pcm_format = dstfmt; self->sample_size = size; break;
+#define FORMAT(srcfmt, dstfmt, size) case srcfmt: pcm_format = dstfmt; self->sample_size = size*channels; break;
switch (format)
{
FORMAT(AUDIO_OBJECT_FORMAT_ALAW, SND_PCM_FORMAT_A_LAW, 1)

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@ -1,76 +0,0 @@
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;
}

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@ -1,6 +1,6 @@
# Template file for 'pcaudiolib'
pkgname=pcaudiolib
version=1.1
version=1.2
revision=1
build_style=gnu-configure
hostmakedepends="automake libtool which pkg-config"
@ -9,8 +9,9 @@ short_desc="Portable C Audio Library"
maintainer="Joey <joey@imap.cc>"
license="GPL-3.0-or-later"
homepage="https://github.com/espeak-ng/pcaudiolib"
changelog="https://raw.githubusercontent.com/espeak-ng/pcaudiolib/master/ChangeLog.md"
distfiles="https://github.com/espeak-ng/pcaudiolib/archive/${version}.tar.gz"
checksum=699a5a347b1e12dc5b122e192e19f4db01621826bf41b9ebefb1cbc63ae2180b
checksum=44b9d509b9eac40a0c61585f756d76a7b555f732e8b8ae4a501c8819c59c6619
pre_configure() {
./autogen.sh