commit: db98333db476343854d4abf5cfe7c86a6f802399 Author: William Hubbs <williamh <AT> gentoo <DOT> org> AuthorDate: Fri Sep 9 05:10:27 2022 +0000 Commit: William Hubbs <williamh <AT> gentoo <DOT> org> CommitDate: Fri Sep 9 05:13:13 2022 +0000 URL: https://gitweb.gentoo.org/repo/gentoo.git/commit/?id=db98333d
media-libs/pcaudiolib: 1.2-r1 revbump update EAPI 7 -> 8 add patches from the bug. Bug: https://bugs.gentoo.org/867793 Signed-off-by: William Hubbs <williamh <AT> gentoo.org> ...0001-Fix-audio-choppiness-on-some-systems.patch | 36 +++++++ .../0002-Fix-latency-related-buffer-sizing.patch | 78 ++++++++++++++ ...buffer-and-send-for-playback-without-bloc.patch | 118 +++++++++++++++++++++ media-libs/pcaudiolib/pcaudiolib-1.2-r1.ebuild | 51 +++++++++ 4 files changed, 283 insertions(+) diff --git a/media-libs/pcaudiolib/files/0001-Fix-audio-choppiness-on-some-systems.patch b/media-libs/pcaudiolib/files/0001-Fix-audio-choppiness-on-some-systems.patch new file mode 100644 index 000000000000..b8c147454df9 --- /dev/null +++ b/media-libs/pcaudiolib/files/0001-Fix-audio-choppiness-on-some-systems.patch @@ -0,0 +1,36 @@ +From 478bb69d51a945123a18dedf4bee7a3bede67f40 Mon Sep 17 00:00:00 2001 +From: Nicolas Pitre <n...@fluxnic.net> +Date: Sat, 2 Apr 2022 22:04:30 -0400 +Subject: [PATCH] Fix audio choppiness on some systems + +Commit a41d46e816d2 ("Fix cancellation snappiness") made espeak unusable +on the RaspberryPi due to extreme audio choppiness. This can sometimes +be observed on some PC-type systems as well, albeit much less +prominently. + +Relax the timing to the smallest value that makes it work again on the +RaspberryPi. +--- + src/audio_priv.h | 6 ++++-- + 1 file changed, 4 insertions(+), 2 deletions(-) + +diff --git a/src/audio_priv.h b/src/audio_priv.h +index 669a037..0c2ce3c 100644 +--- a/src/audio_priv.h ++++ b/src/audio_priv.h +@@ -53,8 +53,10 @@ struct audio_object + }; + + /* We try to aim for 10ms cancelation latency, which will be perceived as +- * "snappy" by users */ +-#define LATENCY 10 ++ * "snappy" by users. However, some systems (e.g. RPi) do produce chopped ++ * audio when this value is smaller than 60. ++ */ ++#define LATENCY 60 + + #if defined(_WIN32) || defined(_WIN64) + +-- +2.35.1 + 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 <n...@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 + diff --git a/media-libs/pcaudiolib/files/0003-Copy-audio-buffer-and-send-for-playback-without-bloc.patch b/media-libs/pcaudiolib/files/0003-Copy-audio-buffer-and-send-for-playback-without-bloc.patch new file mode 100644 index 000000000000..5bc4eddc7edd --- /dev/null +++ b/media-libs/pcaudiolib/files/0003-Copy-audio-buffer-and-send-for-playback-without-bloc.patch @@ -0,0 +1,118 @@ +From e0ff53614074feabc637598000f5799b480179c3 Mon Sep 17 00:00:00 2001 +From: zment <jani.karkkai...@gmail.com> +Date: Tue, 16 Aug 2022 20:18:31 +0300 +Subject: [PATCH] Copy audio buffer and send for playback without blocking, + fixes Windows audio choppyness + +--- + src/xaudio2.cpp | 60 +++++++++++++++++++++++++++++++++++++++++-------- + 1 file changed, 51 insertions(+), 9 deletions(-) + +diff --git a/src/xaudio2.cpp b/src/xaudio2.cpp +index 72190b4..bb9d5be 100644 +--- a/src/xaudio2.cpp ++++ b/src/xaudio2.cpp +@@ -35,6 +35,25 @@ struct xaudio2_object + LPWSTR devicename; + }; + ++class VoiceCallbacks : public IXAudio2VoiceCallback ++{ ++public: ++ void OnBufferEnd(void* pBufferContext) { ++ if (pBufferContext != NULL) ++ { ++ free((void*)pBufferContext); ++ } ++ } ++ ++ // Stubs for all interface callbacks ++ void OnStreamEnd() { } ++ void OnVoiceProcessingPassEnd() { } ++ void OnVoiceProcessingPassStart(UINT32 SamplesRequired) { } ++ void OnBufferStart(void* pBufferContext) { } ++ void OnLoopEnd(void* pBufferContext) { } ++ void OnVoiceError(void* pBufferContext, HRESULT Error) { } ++} voiceCallbacks; ++ + void + xaudio2_object_close(struct audio_object *object); + +@@ -59,7 +78,7 @@ xaudio2_object_open(struct audio_object *object, + if (FAILED(hr)) + goto error; + +- hr = self->audio->CreateSourceVoice(&self->source, self->format); ++ hr = self->audio->CreateSourceVoice(&self->source, self->format, 0, 2.0f, &voiceCallbacks); + if (FAILED(hr)) + goto error; + +@@ -110,6 +129,16 @@ xaudio2_object_drain(struct audio_object *object) + { + struct xaudio2_object *self = to_xaudio2_object(object); + ++ while (true) ++ { ++ Sleep(10); ++ ++ XAUDIO2_VOICE_STATE state = { 0 }; ++ self->source->GetState(&state); ++ if (state.pCurrentBufferContext == NULL && state.BuffersQueued == 0) ++ break; ++ } ++ + return S_OK; + } + +@@ -128,27 +157,40 @@ xaudio2_object_write(struct audio_object *object, + { + struct xaudio2_object *self = to_xaudio2_object(object); + +- XAUDIO2_BUFFER buffer = {0}; ++ BYTE* buf_data = (BYTE *)malloc(bytes); ++ memcpy(buf_data, data, bytes); ++ ++ XAUDIO2_BUFFER buffer = { 0 }; + buffer.AudioBytes = bytes; +- buffer.pAudioData = (const BYTE *)data; ++ buffer.pAudioData = buf_data; ++ buffer.pContext = buf_data; + + HRESULT hr = S_OK; + if (SUCCEEDED(hr)) + hr = self->source->SubmitSourceBuffer(&buffer); + +- if (SUCCEEDED(hr)) +- hr = self->source->Start(0); ++ XAUDIO2_VOICE_STATE state = { 0 }; ++ self->source->GetState(&state); ++ UINT32 buffersQueued = state.BuffersQueued; + +- if (SUCCEEDED(hr)) while (true) ++ while (FAILED(hr)) + { + Sleep(10); + +- XAUDIO2_VOICE_STATE state = { 0 }; + self->source->GetState(&state); +- if (state.pCurrentBufferContext == NULL && state.BuffersQueued == 0) +- return hr; ++ if (state.BuffersQueued < buffersQueued) ++ { ++ hr = self->source->SubmitSourceBuffer(&buffer); ++ ++ self->source->GetState(&state); ++ buffersQueued = state.BuffersQueued; ++ } + } + ++ ++ if (SUCCEEDED(hr)) ++ hr = self->source->Start(0); ++ + return hr; + } + +-- +2.35.1 + diff --git a/media-libs/pcaudiolib/pcaudiolib-1.2-r1.ebuild b/media-libs/pcaudiolib/pcaudiolib-1.2-r1.ebuild new file mode 100644 index 000000000000..6525aa93e2a9 --- /dev/null +++ b/media-libs/pcaudiolib/pcaudiolib-1.2-r1.ebuild @@ -0,0 +1,51 @@ +# Copyright 1999-2022 Gentoo Authors +# Distributed under the terms of the GNU General Public License v2 + +EAPI=8 + +inherit autotools + +DESCRIPTION="Portable C Audio Library" +HOMEPAGE="https://github.com/espeak-ng/pcaudiolib"; +SRC_URI="https://github.com/espeak-ng/pcaudiolib/archive/${PV}.tar.gz -> ${P}.tar.gz" + +LICENSE="GPL-3+ ZLIB" +SLOT="0" +KEYWORDS="~alpha ~amd64 ~arm ~arm64 ~hppa ~ia64 ~loong ~ppc ~ppc64 ~riscv ~sparc ~x86" +IUSE="+alsa oss pulseaudio" + +REQUIRED_USE="|| ( alsa oss pulseaudio )" + +RDEPEND=" + alsa? ( >=media-libs/alsa-lib-1.0.18 ) + pulseaudio? ( media-sound/pulseaudio ) +" +DEPEND="${RDEPEND}" +BDEPEND="virtual/pkgconfig" + +PATCHES=( + "${FILESDIR}/0001-Fix-audio-choppiness-on-some-systems.patch" + "${FILESDIR}/0002-Fix-latency-related-buffer-sizing.patch" + "${FILESDIR}/0003-Copy-audio-buffer-and-send-for-playback-without-bloc.patch" + ) + +src_prepare() { + default + eautoreconf +} + +src_configure() { + local econf_args + econf_args=( + $(use_with oss) + $(use_with alsa) + $(use_with pulseaudio) + --disable-static + ) + econf "${econf_args[@]}" +} + +src_install() { + default + find "${ED}" -name '*.la' -delete || die +}
