diff options
Diffstat (limited to 'src/modules/module-solaris.c')
-rw-r--r-- | src/modules/module-solaris.c | 100 |
1 files changed, 64 insertions, 36 deletions
diff --git a/src/modules/module-solaris.c b/src/modules/module-solaris.c index 0920d25e..71f14071 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[] = { @@ -133,6 +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; @@ -145,7 +151,12 @@ 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 +166,8 @@ 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 +400,8 @@ 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 +413,8 @@ 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; @@ -479,7 +496,7 @@ 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 +518,7 @@ 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 +531,7 @@ 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 +553,7 @@ 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); } } @@ -606,11 +621,13 @@ 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) @@ -629,12 +646,15 @@ 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 (;;) { 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 @@ -652,38 +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); @@ -691,7 +704,9 @@ 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); @@ -797,7 +812,7 @@ 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); } @@ -812,7 +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; @@ -838,6 +853,9 @@ 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. @@ -861,7 +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; } @@ -924,6 +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; @@ -966,15 +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; @@ -1075,6 +1100,9 @@ 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); |