diff options
Diffstat (limited to 'media-libs/pcaudiolib/files/0002-Fix-latency-related-buffer-sizing.patch')
| -rw-r--r-- | media-libs/pcaudiolib/files/0002-Fix-latency-related-buffer-sizing.patch | 78 |
1 files changed, 78 insertions, 0 deletions
diff --git a/media-libs/pcaudiolib/files/0002-Fix-latency-related-buffer-sizing.patch b/media-libs/pcaudiolib/files/0002-Fix-latency-related-buffer-sizing.patch new file mode 100644 index 000000000000..4af7da5c33a4 --- /dev/null +++ b/media-libs/pcaudiolib/files/0002-Fix-latency-related-buffer-sizing.patch @@ -0,0 +1,78 @@ +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 + |