[pulseaudio-commits] [SCM] PulseAudio Sound Server branch, master, updated. v0.9.16-test5-27-g601fb63
Lennart Poettering
gitmailer-noreply at 0pointer.de
Fri Aug 21 15:59:55 PDT 2009
This is an automated email from the git hooks/post-receive script. It was
generated because of a push to the "PulseAudio Sound Server" repository.
The master branch has been updated
from 15eb03a5b39f8c54328caa7516a7870bf977db40 (commit)
- Log -----------------------------------------------------------------
601fb63 Solaris: fixed latency (resent)
87d2dde Solaris: use smoother (resent)
44c7aa5 Solaris: build fixes (resent)
b96390f Solaris: bootstrap portability
-----------------------------------------------------------------------
Summary of changes:
bootstrap.sh | 6 +-
src/modules/module-solaris.c | 100 +++++++++++++++++++++++++++---------------
2 files changed, 67 insertions(+), 39 deletions(-)
-----------------------------------------------------------------------
commit b96390fc9878db5c244256545f36fa14ea1f5276
Author: Finn Thain <fthain at telegraphics.com.au>
Date: Fri Aug 21 18:13:11 2009 +1000
Solaris: bootstrap portability
On Fri, 21 Aug 2009, Colin Guthrie wrote:
>
> Just put an echo statement in there too. Should cover the bases for everyone.
Something like this?
diff --git a/bootstrap.sh b/bootstrap.sh
index 970e884..c7c8582 100755
--- a/bootstrap.sh
+++ b/bootstrap.sh
@@ -47,9 +47,9 @@ case $(uname) in
esac
if [ -f .git/hooks/pre-commit.sample -a ! -f .git/hooks/pre-commit ] ; then
- echo "Activating pre-commit hook."
- cp -pv .git/hooks/pre-commit.sample .git/hooks/pre-commit
- chmod -v +x .git/hooks/pre-commit
+ cp -p .git/hooks/pre-commit.sample .git/hooks/pre-commit && \
+ chmod +x .git/hooks/pre-commit && \
+ echo "Activated pre-commit hook."
fi
if [ -f .tarball-version ]; then
commit 44c7aa55e25334901769b82355c12dee91cb3629
Author: Finn Thain <fthain at telegraphics.com.au>
Date: Fri Aug 21 13:15:38 2009 +1000
Solaris: build fixes (resent)
Fix bit rot due to recent flat volume changes.
diff --git a/src/modules/module-solaris.c b/src/modules/module-solaris.c
index 0920d25..2c878c2 100644
--- a/src/modules/module-solaris.c
+++ b/src/modules/module-solaris.c
@@ -479,7 +479,7 @@ static void sink_set_volume(pa_sink *s) {
if (u->fd >= 0) {
AUDIO_INITINFO(&info);
- info.play.gain = pa_cvolume_max(&s->virtual_volume) * AUDIO_MAX_GAIN / PA_VOLUME_NORM;
+ info.play.gain = pa_cvolume_max(&s->real_volume) * AUDIO_MAX_GAIN / PA_VOLUME_NORM;
assert(info.play.gain <= AUDIO_MAX_GAIN);
if (ioctl(u->fd, AUDIO_SETINFO, &info) < 0) {
@@ -501,8 +501,7 @@ static void sink_get_volume(pa_sink *s) {
if (ioctl(u->fd, AUDIO_GETINFO, &info) < 0)
pa_log("AUDIO_SETINFO: %s", pa_cstrerror(errno));
else
- pa_cvolume_set(&s->virtual_volume, s->sample_spec.channels,
- info.play.gain * PA_VOLUME_NORM / AUDIO_MAX_GAIN);
+ pa_cvolume_set(&s->real_volume, s->sample_spec.channels, info.play.gain * PA_VOLUME_NORM / AUDIO_MAX_GAIN);
}
}
@@ -515,7 +514,7 @@ static void source_set_volume(pa_source *s) {
if (u->fd >= 0) {
AUDIO_INITINFO(&info);
- info.play.gain = pa_cvolume_max(&s->virtual_volume) * AUDIO_MAX_GAIN / PA_VOLUME_NORM;
+ info.play.gain = pa_cvolume_max(&s->volume) * AUDIO_MAX_GAIN / PA_VOLUME_NORM;
assert(info.play.gain <= AUDIO_MAX_GAIN);
if (ioctl(u->fd, AUDIO_SETINFO, &info) < 0) {
@@ -537,8 +536,7 @@ static void source_get_volume(pa_source *s) {
if (ioctl(u->fd, AUDIO_GETINFO, &info) < 0)
pa_log("AUDIO_SETINFO: %s", pa_cstrerror(errno));
else
- pa_cvolume_set(&s->virtual_volume, s->sample_spec.channels,
- info.play.gain * PA_VOLUME_NORM / AUDIO_MAX_GAIN);
+ pa_cvolume_set(&s->volume, s->sample_spec.channels, info.play.gain * PA_VOLUME_NORM / AUDIO_MAX_GAIN);
}
}
@@ -797,7 +795,7 @@ static void sig_callback(pa_mainloop_api *api, pa_signal_event*e, int sig, void
pa_log_debug("caught signal");
if (u->sink) {
- pa_sink_get_volume(u->sink, TRUE, FALSE);
+ pa_sink_get_volume(u->sink, TRUE);
pa_sink_get_mute(u->sink, TRUE);
}
commit 87d2dded9b90331943a6c7b9d8d9b1ac100b6689
Author: Finn Thain <fthain at telegraphics.com.au>
Date: Fri Aug 21 13:17:03 2009 +1000
Solaris: use smoother (resent)
Make use of the smoother, just in case.
diff --git a/src/modules/module-solaris.c b/src/modules/module-solaris.c
index 2c878c2..3bf7c4b 100644
--- a/src/modules/module-solaris.c
+++ b/src/modules/module-solaris.c
@@ -60,6 +60,7 @@
#include <pulsecore/thread-mq.h>
#include <pulsecore/rtpoll.h>
#include <pulsecore/thread.h>
+#include <pulsecore/time-smoother.h>
#include "module-solaris-symdef.h"
@@ -110,6 +111,8 @@ struct userdata {
uint32_t prev_playback_samples, prev_record_samples;
int32_t minimum_request;
+
+ pa_smoother *smoother;
};
static const char* const valid_modargs[] = {
@@ -145,7 +148,12 @@ static uint64_t get_playback_buffered_bytes(struct userdata *u) {
/* Handle wrap-around of the device's sample counter, which is a uint_32. */
if (u->prev_playback_samples > info.play.samples) {
- /* Unfortunately info.play.samples can sometimes go backwards, even before it wraps! */
+ /*
+ * Unfortunately info.play.samples can sometimes go backwards, even before it wraps!
+ * The bug seems to be absent on Solaris x86 nv117 with audio810 driver, at least on this (UP) machine.
+ * The bug is present on a different (SMP) machine running Solaris x86 nv103 with audioens driver.
+ * An earlier revision of this file mentions the same bug independently (unknown configuration).
+ */
if (u->prev_playback_samples + info.play.samples < 240000) {
++u->play_samples_msw;
} else {
@@ -155,6 +163,8 @@ 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;
+ pa_smoother_put(u->smoother, pa_rtclock_now(), pa_bytes_to_usec(played_bytes, &u->sink->sample_spec));
+
return u->written_bytes - played_bytes;
}
@@ -387,6 +397,8 @@ static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offse
pa_assert(PA_SINK_IS_OPENED(u->sink->thread_info.state));
+ pa_smoother_pause(u->smoother, pa_rtclock_now());
+
if (!u->source || u->source_suspended) {
if (suspend(u) < 0)
return -1;
@@ -398,6 +410,8 @@ static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offse
case PA_SINK_RUNNING:
if (u->sink->thread_info.state == PA_SINK_SUSPENDED) {
+ pa_smoother_resume(u->smoother, pa_rtclock_now(), TRUE);
+
if (!u->source || u->source_suspended) {
if (unsuspend(u) < 0)
return -1;
@@ -604,11 +618,13 @@ static void thread_func(void *userdata) {
pa_thread_mq_install(&u->thread_mq);
+ pa_smoother_set_time_offset(u->smoother, pa_rtclock_now());
+
for (;;) {
/* Render some data and write it to the dsp */
if (u->sink && PA_SINK_IS_OPENED(u->sink->thread_info.state)) {
- pa_usec_t xtime0;
+ pa_usec_t xtime0, ysleep_interval, xsleep_interval;
uint64_t buffered_bytes;
if (u->sink->thread_info.rewind_requested)
@@ -627,6 +643,8 @@ static void thread_func(void *userdata) {
info.play.error = 0;
if (ioctl(u->fd, AUDIO_SETINFO, &info) < 0)
pa_log("AUDIO_SETINFO: %s", pa_cstrerror(errno));
+
+ pa_smoother_reset(u->smoother, pa_rtclock_now(), TRUE);
}
for (;;) {
@@ -689,7 +707,9 @@ static void thread_func(void *userdata) {
}
}
- pa_rtpoll_set_timer_absolute(u->rtpoll, xtime0 + pa_bytes_to_usec(buffered_bytes / 2, &u->sink->sample_spec));
+ ysleep_interval = pa_bytes_to_usec(buffered_bytes / 2, &u->sink->sample_spec);
+ xsleep_interval = pa_smoother_translate(u->smoother, xtime0, ysleep_interval);
+ pa_rtpoll_set_timer_absolute(u->rtpoll, xtime0 + PA_MIN(xsleep_interval, ysleep_interval));
} else
pa_rtpoll_set_timer_disabled(u->rtpoll);
@@ -836,6 +856,9 @@ int pa__init(pa_module *m) {
u = pa_xnew0(struct userdata, 1);
+ if (!(u->smoother = pa_smoother_new(PA_USEC_PER_SEC, PA_USEC_PER_SEC * 2, TRUE, TRUE, 10, pa_rtclock_now(), TRUE)))
+ goto fail;
+
/*
* For a process (or several processes) to use the same audio device for both
* record and playback at the same time, the device's mixer must be enabled.
@@ -1073,6 +1096,9 @@ void pa__done(pa_module *m) {
if (u->fd >= 0)
close(u->fd);
+ if (u->smoother)
+ pa_smoother_free(u->smoother);
+
pa_xfree(u->device_name);
pa_xfree(u);
commit 601fb63b0160d3d76083d07dcc1201a123031915
Author: Finn Thain <fthain at telegraphics.com.au>
Date: Fri Aug 21 13:18:40 2009 +1000
Solaris: fixed latency (resent)
Set a fixed latency based on the given buffer size, which is constrained to
the 128 KB limit on buffered writes. Also fix an error path.
diff --git a/src/modules/module-solaris.c b/src/modules/module-solaris.c
index 3bf7c4b..71f1407 100644
--- a/src/modules/module-solaris.c
+++ b/src/modules/module-solaris.c
@@ -136,6 +136,9 @@ static const char* const valid_modargs[] = {
#define MAX_RENDER_HZ (300)
/* This render rate limit imposes a minimum latency, but without it we waste too much CPU time. */
+#define MAX_BUFFER_SIZE (128 * 1024)
+/* An attempt to buffer more than 128 KB causes write() to fail with errno == EAGAIN. */
+
static uint64_t get_playback_buffered_bytes(struct userdata *u) {
audio_info_t info;
uint64_t played_bytes;
@@ -651,6 +654,7 @@ static void thread_func(void *userdata) {
void *p;
ssize_t w;
size_t len;
+ int write_type = 1;
/*
* Since we cannot modify the size of the output buffer we fake it
@@ -668,38 +672,31 @@ static void thread_func(void *userdata) {
break;
if (u->memchunk.length < len)
- pa_sink_render(u->sink, u->sink->thread_info.max_request, &u->memchunk);
+ pa_sink_render(u->sink, len - u->memchunk.length, &u->memchunk);
+
+ len = PA_MIN(u->memchunk.length, len);
p = pa_memblock_acquire(u->memchunk.memblock);
- w = pa_write(u->fd, (uint8_t*) p + u->memchunk.index, u->memchunk.length, NULL);
+ w = pa_write(u->fd, (uint8_t*) p + u->memchunk.index, len, &write_type);
pa_memblock_release(u->memchunk.memblock);
if (w <= 0) {
- switch (errno) {
- case EINTR:
- continue;
- case EAGAIN:
- /* If the buffer_size is too big, we get EAGAIN. Avoiding that limit by trial and error
- * is not ideal, but I don't know how to get the system to tell me what the limit is.
- */
- u->buffer_size = u->buffer_size * 18 / 25;
- u->buffer_size -= u->buffer_size % u->frame_size;
- u->buffer_size = PA_MAX(u->buffer_size, 2 * u->minimum_request);
- pa_sink_set_max_request_within_thread(u->sink, u->buffer_size);
- pa_sink_set_max_rewind_within_thread(u->sink, u->buffer_size);
- pa_log("EAGAIN. Buffer size is now %u bytes (%llu buffered)", u->buffer_size, buffered_bytes);
- break;
- default:
- pa_log("Failed to write data to DSP: %s", pa_cstrerror(errno));
- goto fail;
+ if (errno == EINTR) {
+ continue;
+ } else if (errno == EAGAIN) {
+ /* We may have realtime priority so yield the CPU to ensure that fd can become writable again. */
+ pa_log_debug("EAGAIN with %llu bytes buffered.", buffered_bytes);
+ break;
+ } else {
+ pa_log("Failed to write data to DSP: %s", pa_cstrerror(errno));
+ goto fail;
}
} else {
pa_assert(w % u->frame_size == 0);
u->written_bytes += w;
- u->memchunk.length -= w;
-
u->memchunk.index += w;
+ u->memchunk.length -= w;
if (u->memchunk.length <= 0) {
pa_memblock_unref(u->memchunk.memblock);
pa_memchunk_reset(&u->memchunk);
@@ -830,7 +827,7 @@ int pa__init(pa_module *m) {
pa_channel_map map;
pa_modargs *ma = NULL;
uint32_t buffer_length_msec;
- int fd;
+ int fd = -1;
pa_sink_new_data sink_new_data;
pa_source_new_data source_new_data;
char const *name;
@@ -882,7 +879,13 @@ int pa__init(pa_module *m) {
}
u->buffer_size = pa_usec_to_bytes(1000 * buffer_length_msec, &ss);
if (u->buffer_size < 2 * u->minimum_request) {
- pa_log("supplied buffer size argument is too small");
+ pa_log("buffer_length argument cannot be smaller than %u",
+ (unsigned)(pa_bytes_to_usec(2 * u->minimum_request, &ss) / 1000));
+ goto fail;
+ }
+ if (u->buffer_size > MAX_BUFFER_SIZE) {
+ pa_log("buffer_length argument cannot be greater than %u",
+ (unsigned)(pa_bytes_to_usec(MAX_BUFFER_SIZE, &ss) / 1000));
goto fail;
}
@@ -945,6 +948,7 @@ int pa__init(pa_module *m) {
pa_source_set_asyncmsgq(u->source, u->thread_mq.inq);
pa_source_set_rtpoll(u->source, u->rtpoll);
+ pa_source_set_fixed_latency(u->source, pa_bytes_to_usec(u->buffer_size, &u->source->sample_spec));
u->source->get_volume = source_get_volume;
u->source->set_volume = source_set_volume;
@@ -987,15 +991,15 @@ int pa__init(pa_module *m) {
pa_sink_set_asyncmsgq(u->sink, u->thread_mq.inq);
pa_sink_set_rtpoll(u->sink, u->rtpoll);
+ pa_sink_set_fixed_latency(u->sink, pa_bytes_to_usec(u->buffer_size, &u->sink->sample_spec));
+ pa_sink_set_max_request(u->sink, u->buffer_size);
+ pa_sink_set_max_rewind(u->sink, u->buffer_size);
u->sink->get_volume = sink_get_volume;
u->sink->set_volume = sink_set_volume;
u->sink->get_mute = sink_get_mute;
u->sink->set_mute = sink_set_mute;
u->sink->refresh_volume = u->sink->refresh_muted = TRUE;
-
- pa_sink_set_max_request(u->sink, u->buffer_size);
- pa_sink_set_max_rewind(u->sink, u->buffer_size);
} else
u->sink = NULL;
--
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