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solaris: Allow module-solaris to use alternative smoother code

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This is untested.

Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/55>
This commit is contained in:
Georg Chini 2020-01-15 20:52:31 +01:00 • committed by PulseAudio Marge Bot
parent a92f566e36
commit 0497821afc
1 changed files with 43 additions and 0 deletions
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43
src/modules/module-solaris.c
View file

@ -60,7 +60,12 @@
#include <pulsecore/thread-mq.h>
#include <pulsecore/rtpoll.h>
#include <pulsecore/thread.h>
#ifdef USE_SMOOTHER_2
#include <pulsecore/time-smoother_2.h>
#else
#include <pulsecore/time-smoother.h>
#endif
PA_MODULE_AUTHOR("Pierre Ossman");
PA_MODULE_DESCRIPTION("Solaris Sink/Source");
@ -110,7 +115,11 @@ struct userdata {
int32_t minimum_request;
#ifdef USE_SMOOTHER_2
pa_smoother_2 *smoother;
#else
pa_smoother *smoother;
#endif
};
static const char* const valid_modargs[] = {
@ -164,7 +173,11 @@ static uint64_t get_playback_buffered_bytes(struct userdata *u) {
u->prev_playback_samples = info.play.samples;
played_bytes = (((uint64_t)u->play_samples_msw << 32) + info.play.samples) * u->frame_size;
#ifdef USE_SMOOTHER_2
pa_smoother_2_put(u->smoother, pa_rtclock_now(), played_bytes);
#else
pa_smoother_put(u->smoother, pa_rtclock_now(), pa_bytes_to_usec(played_bytes, &u->sink->sample_spec));
#endif
if (u->written_bytes > played_bytes)
return u->written_bytes - played_bytes;
@ -413,7 +426,11 @@ static int sink_set_state_in_io_thread_cb(pa_sink *s, pa_sink_state_t new_state,
pa_assert(PA_SINK_IS_OPENED(s->thread_info.state));
#ifdef USE_SMOOTHER_2
pa_smoother_2_pause(u->smoother, pa_rtclock_now());
#else
pa_smoother_pause(u->smoother, pa_rtclock_now());
#endif
if (!u->source || u->source_suspended)
suspend(u);
@ -425,7 +442,11 @@ static int sink_set_state_in_io_thread_cb(pa_sink *s, pa_sink_state_t new_state,
case PA_SINK_RUNNING:
if (s->thread_info.state == PA_SINK_SUSPENDED) {
#ifdef USE_SMOOTHER_2
pa_smoother_2_resume(u->smoother, pa_rtclock_now());
#else
pa_smoother_resume(u->smoother, pa_rtclock_now(), true);
#endif
if (!u->source || u->source_suspended) {
bool mute;
@ -654,7 +675,11 @@ static void thread_func(void *userdata) {
pa_thread_mq_install(&u->thread_mq);
#ifdef USE_SMOOTHER_2
pa_smoother_2_reset(u->smoother, pa_rtclock_now());
#else
pa_smoother_set_time_offset(u->smoother, pa_rtclock_now());
#endif
for (;;) {
/* Render some data and write it to the dsp */
@ -680,7 +705,11 @@ static void thread_func(void *userdata) {
if (ioctl(u->fd, AUDIO_SETINFO, &info) < 0)
pa_log("AUDIO_SETINFO: %s", pa_cstrerror(errno));
#ifdef USE_SMOOTHER_2
pa_smoother_2_reset(u->smoother, pa_rtclock_now());
#else
pa_smoother_reset(u->smoother, pa_rtclock_now(), true);
#endif
}
for (;;) {
@ -736,7 +765,11 @@ static void thread_func(void *userdata) {
}
ysleep_interval = pa_bytes_to_usec(buffered_bytes / 2, &u->sink->sample_spec);
#ifdef USE_SMOOTHER_2
xsleep_interval = pa_smoother_2_translate(u->smoother, ysleep_interval);
#else
xsleep_interval = pa_smoother_translate(u->smoother, xtime0, ysleep_interval);
#endif
pa_rtpoll_set_timer_absolute(u->rtpoll, xtime0 + PA_MIN(xsleep_interval, ysleep_interval));
} else
pa_rtpoll_set_timer_disabled(u->rtpoll);
@ -884,8 +917,10 @@ int pa__init(pa_module *m) {
u = pa_xnew0(struct userdata, 1);
#ifndef USE_SMOOTHER_2
if (!(u->smoother = pa_smoother_new(PA_USEC_PER_SEC, PA_USEC_PER_SEC * 2, true, true, 10, pa_rtclock_now(), true)))
goto fail;
#endif
/*
* For a process (or several processes) to use the same audio device for both
@ -901,6 +936,10 @@ int pa__init(pa_module *m) {
}
u->frame_size = pa_frame_size(&ss);
#ifdef USE_SMOOTHER_2
u->smoother = pa_smoother_2_new(5*PA_USEC_PER_SEC, pa_rtclock_now(), u->frame_size, ss.rate);
#endif
u->minimum_request = pa_usec_to_bytes(PA_USEC_PER_SEC / MAX_RENDER_HZ, &ss);
buffer_length_msec = 100;
@ -1142,7 +1181,11 @@ void pa__done(pa_module *m) {
close(u->fd);
if (u->smoother)
#ifdef USE_SMOOTHER_2
pa_smoother_2_free(u->smoother);
#else
pa_smoother_free(u->smoother);
#endif
pa_xfree(u->device_name);