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ananace/99-better-pa.patch

Created Mar 22, 2018
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PA patch for qemu 2.11
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99-better-pa.patch
diff --git a/audio/audio.c b/audio/audio.c
index beafed209b..6f42a019b0 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2066,3 +2066,8 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
}
}
}
+
+int64_t audio_get_timer_ticks(void)
+{
+ return conf.period.ticks;
+}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 5bcb1c60e1..2f7fc4f8ac 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -214,6 +214,8 @@ extern struct audio_driver pa_audio_driver;
extern struct audio_driver spice_audio_driver;
extern const struct mixeng_volume nominal_volume;
+int64_t audio_get_timer_ticks(void);
+
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 65beb6f010..b46beeea92 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -1,16 +1,22 @@
/* public domain */
#include "qemu/osdep.h"
-#include "qemu-common.h"
+#include "qemu/timer.h"
#include "audio.h"
#include <pulse/pulseaudio.h>
#define AUDIO_CAP "pulseaudio"
+#define DEBUG
#include "audio_int.h"
-#include "audio_pt_int.h"
typedef struct {
- int samples;
+ int buffer_size_out;
+ int buffer_size_in;
+ int tlength;
+ int fragsize;
+ int maxlength_in;
+ int adjust_latency_out;
+ int adjust_latency_in;
char *server;
char *sink;
char *source;
@@ -24,28 +30,18 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
- int done;
- int live;
- int decr;
- int rpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
paaudio *g;
+ pa_sample_spec ss;
+ pa_buffer_attr ba;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
- int done;
- int dead;
- int incr;
- int wpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
- const void *read_data;
- size_t read_index, read_length;
paaudio *g;
+ pa_sample_spec ss;
+ pa_buffer_attr ba;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -109,182 +105,59 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
} \
} while (0);
-static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
-{
- paaudio *g = p->g;
-
- pa_threaded_mainloop_lock (g->mainloop);
-
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
-
- while (length > 0) {
- size_t l;
-
- while (!p->read_data) {
- int r;
-
- r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
-
- if (!p->read_data) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
- } else {
- p->read_index = 0;
- }
- }
-
- l = p->read_length < length ? p->read_length : length;
- memcpy (data, (const uint8_t *) p->read_data+p->read_index, l);
-
- data = (uint8_t *) data + l;
- length -= l;
-
- p->read_index += l;
- p->read_length -= l;
-
- if (!p->read_length) {
- int r;
-
- r = pa_stream_drop (p->stream);
- p->read_data = NULL;
- p->read_length = 0;
- p->read_index = 0;
-
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
- }
- }
-
- pa_threaded_mainloop_unlock (g->mainloop);
- return 0;
-
-unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
- return -1;
-}
-
-static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
+static int qpa_run_out(HWVoiceOut *hw, int live)
{
- paaudio *g = p->g;
-
- pa_threaded_mainloop_lock (g->mainloop);
-
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
-
- while (length > 0) {
- size_t l;
- int r;
-
- while (!(l = pa_stream_writable_size (p->stream))) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
- }
-
- CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
-
- if (l > length) {
- l = length;
- }
-
- r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
-
- data = (const uint8_t *) data + l;
- length -= l;
- }
-
- pa_threaded_mainloop_unlock (g->mainloop);
- return 0;
-
-unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
- return -1;
-}
-
-static void *qpa_thread_out (void *arg)
-{
- PAVoiceOut *pa = arg;
- HWVoiceOut *hw = &pa->hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
- }
+ PAVoiceOut *pa = (PAVoiceOut *) hw;
+ int rpos, decr, samples;
+ size_t avail_bytes, max_bytes;
+ struct st_sample *src;
+ void *pa_dst;
+ int error = 0;
+ int *rerror = &error;
+ int r;
- for (;;) {
- int decr, to_mix, rpos;
+ decr = 0;
+ rpos = hw->rpos;
- for (;;) {
- if (pa->done) {
- goto exit;
- }
+ pa_threaded_mainloop_lock(pa->g->mainloop);
+ CHECK_DEAD_GOTO(pa->g, pa->stream, rerror, fail);
- if (pa->live > 0) {
- break;
- }
+ avail_bytes = (size_t) live << hw->info.shift;
- if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
- goto exit;
- }
- }
+ max_bytes = pa_stream_writable_size(pa->stream);
+ CHECK_SUCCESS_GOTO(pa->g, rerror, max_bytes != -1, fail);
- decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
- rpos = pa->rpos;
+ samples = (int)(audio_MIN(avail_bytes, max_bytes)) >> hw->info.shift;
+ while (samples) {
+ int convert_samples = audio_MIN(samples, hw->samples - rpos);
+ size_t b_wanted = (size_t) convert_samples << hw->info.shift;
+ size_t b_effective = b_wanted;
- if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
- }
+ r = pa_stream_begin_write(pa->stream, &pa_dst, &b_effective);
+ CHECK_SUCCESS_GOTO(pa->g, rerror, r == 0, fail);
+ CHECK_SUCCESS_GOTO(pa->g, (int *)0, b_effective == b_wanted, fail);
- while (to_mix) {
- int error;
- int chunk = audio_MIN (to_mix, hw->samples - rpos);
- struct st_sample *src = hw->mix_buf + rpos;
+ src = hw->mix_buf + rpos;
+ hw->clip(pa_dst, src, convert_samples);
- hw->clip (pa->pcm_buf, src, chunk);
-
- if (qpa_simple_write (pa, pa->pcm_buf,
- chunk << hw->info.shift, &error) < 0) {
- qpa_logerr (error, "pa_simple_write failed\n");
- return NULL;
- }
+ r = pa_stream_write(pa->stream, pa_dst, b_effective,
+ NULL, 0LL, PA_SEEK_RELATIVE);
+ CHECK_SUCCESS_GOTO(pa->g, rerror, r >= 0, fail);
- rpos = (rpos + chunk) % hw->samples;
- to_mix -= chunk;
- }
-
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- pa->rpos = rpos;
- pa->live -= decr;
- pa->decr += decr;
+ rpos = (rpos + convert_samples) % hw->samples;
+ samples -= convert_samples;
+ decr += convert_samples;
}
- exit:
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qpa_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- PAVoiceOut *pa = (PAVoiceOut *) hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return 0;
- }
+ bail:
+ pa_threaded_mainloop_unlock(pa->g->mainloop);
- decr = audio_MIN (live, pa->decr);
- pa->decr -= decr;
- pa->live = live - decr;
- hw->rpos = pa->rpos;
- if (pa->live > 0) {
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- }
+ hw->rpos = rpos;
return decr;
+
+fail:
+ qpa_logerr(error, "qpa_run_out failed\n");
+ goto bail;
}
static int qpa_write (SWVoiceOut *sw, void *buf, int len)
@@ -292,92 +165,68 @@ static int qpa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-/* capture */
-static void *qpa_thread_in (void *arg)
+static int qpa_run_in(HWVoiceIn *hw)
{
- PAVoiceIn *pa = arg;
- HWVoiceIn *hw = &pa->hw;
+ PAVoiceIn *pa = (PAVoiceIn *) hw;
+ int wpos, incr;
+ char *pa_src;
+ int error = 0;
+ int *rerror = &error;
+ int r;
+ size_t pa_avail;
+ incr = 0;
+ wpos = hw->wpos;
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
- }
+ pa_threaded_mainloop_lock(pa->g->mainloop);
+ CHECK_DEAD_GOTO(pa->g, pa->stream, rerror, fail);
- for (;;) {
- int incr, to_grab, wpos;
+ size_t bytes_wanted = ((unsigned int)
+ (hw->samples - audio_pcm_hw_get_live_in(hw)) << hw->info.shift);
- for (;;) {
- if (pa->done) {
- goto exit;
- }
+ if (bytes_wanted == 0) {
+ /* no room */
+ goto bail;
+ }
- if (pa->dead > 0) {
- break;
- }
+ size_t bytes_avail = pa_stream_readable_size(pa->stream);
- if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
- goto exit;
- }
- }
+ if (bytes_wanted > bytes_avail) {
+ bytes_wanted = bytes_avail;
+ }
- incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
- wpos = pa->wpos;
+ while (bytes_wanted) {
+ r = pa_stream_peek(pa->stream, (const void **)&pa_src, &pa_avail);
+ CHECK_SUCCESS_GOTO(pa->g, rerror, r == 0, fail);
- if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
+ if (pa_avail == 0 || pa_avail > bytes_wanted) {
+ break;
}
- while (to_grab) {
- int error;
- int chunk = audio_MIN (to_grab, hw->samples - wpos);
- void *buf = advance (pa->pcm_buf, wpos);
+ bytes_wanted -= pa_avail;
- if (qpa_simple_read (pa, buf,
- chunk << hw->info.shift, &error) < 0) {
- qpa_logerr (error, "pa_simple_read failed\n");
- return NULL;
- }
-
- hw->conv (hw->conv_buf + wpos, buf, chunk);
+ while (pa_avail) {
+ int chunk = audio_MIN(
+ (int)(pa_avail >> hw->info.shift), hw->samples - wpos);
+ hw->conv(hw->conv_buf + wpos, pa_src, chunk);
wpos = (wpos + chunk) % hw->samples;
- to_grab -= chunk;
- }
-
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return NULL;
+ pa_src += chunk << hw->info.shift;
+ pa_avail -= chunk << hw->info.shift;
+ incr += chunk;
}
- pa->wpos = wpos;
- pa->dead -= incr;
- pa->incr += incr;
+ r = pa_stream_drop(pa->stream);
+ CHECK_SUCCESS_GOTO(pa->g, rerror, r == 0, fail);
}
- exit:
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qpa_run_in (HWVoiceIn *hw)
-{
- int live, incr, dead;
- PAVoiceIn *pa = (PAVoiceIn *) hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
- return 0;
- }
+bail:
+ pa_threaded_mainloop_unlock(pa->g->mainloop);
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- incr = audio_MIN (dead, pa->incr);
- pa->incr -= incr;
- pa->dead = dead - incr;
- hw->wpos = pa->wpos;
- if (pa->dead > 0) {
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- }
+ hw->wpos = wpos;
return incr;
+
+fail:
+ qpa_logerr(error, "qpa_run_in failed\n");
+ goto bail;
}
static int qpa_read (SWVoiceIn *sw, void *buf, int len)
@@ -470,13 +319,6 @@ static void stream_state_cb (pa_stream *s, void * userdata)
}
}
-static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
-{
- paaudio *g = userdata;
-
- pa_threaded_mainloop_signal (g->mainloop, 0);
-}
-
static pa_stream *qpa_simple_new (
paaudio *g,
const char *name,
@@ -498,23 +340,17 @@ static pa_stream *qpa_simple_new (
}
pa_stream_set_state_callback (stream, stream_state_cb, g);
- pa_stream_set_read_callback (stream, stream_request_cb, g);
- pa_stream_set_write_callback (stream, stream_request_cb, g);
if (dir == PA_STREAM_PLAYBACK) {
- r = pa_stream_connect_playback (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+ r = pa_stream_connect_playback(stream, dev, attr,
+ PA_STREAM_INTERPOLATE_TIMING
+ | (g->conf.adjust_latency_out ? PA_STREAM_ADJUST_LATENCY : 0)
+ | PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
} else {
- r = pa_stream_connect_record (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE);
+ r = pa_stream_connect_record(stream, dev, attr,
+ PA_STREAM_INTERPOLATE_TIMING
+ | (g->conf.adjust_latency_in ? PA_STREAM_ADJUST_LATENCY : 0)
+ | PA_STREAM_AUTO_TIMING_UPDATE);
}
if (r < 0) {
@@ -541,165 +377,167 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
int error;
- pa_sample_spec ss;
- pa_buffer_attr ba;
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
- ss.format = audfmt_to_pa (as->fmt, as->endianness);
- ss.channels = as->nchannels;
- ss.rate = as->freq;
-
- /*
- * qemu audio tick runs at 100 Hz (by default), so processing
- * data chunks worth 10 ms of sound should be a good fit.
- */
- ba.tlength = pa_usec_to_bytes (10 * 1000, &ss);
- ba.minreq = pa_usec_to_bytes (5 * 1000, &ss);
- ba.maxlength = -1;
- ba.prebuf = -1;
-
- obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
-
- pa->stream = qpa_simple_new (
- g,
- "qemu",
- PA_STREAM_PLAYBACK,
- g->conf.sink,
- &ss,
- NULL, /* channel map */
- &ba, /* buffering attributes */
- &error
- );
+ int64_t timer_tick_duration =
+ audio_MAX(audio_get_timer_ticks(), 1 * SCALE_MS);
+ int64_t frames_per_tick_x1000 =
+ ((timer_tick_duration * as->freq * 1000LL) / NANOSECONDS_PER_SECOND);
+
+ int64_t tlength = g->conf.tlength;
+ if (tlength == 0) {
+ tlength = (frames_per_tick_x1000) / 400;
+ }
+ int64_t buflen = g->conf.buffer_size_out;
+ if (buflen == 0) {
+ buflen = frames_per_tick_x1000 / 400;
+ }
+
+ ldebug("tick duration: %.2f ms (%.3f frames)\n",
+ ((float)timer_tick_duration) / SCALE_MS,
+ (float)frames_per_tick_x1000 / 1000.0f);
+
+ ldebug("OUT internal buffer: %.2f ms (%"PRId64" frames)\n",
+ buflen * (1000.0f / as->freq),
+ buflen);
+
+ ldebug("OUT tlength: %.2f ms (%"PRId64" frames)\n",
+ tlength * (1000.0f / as->freq),
+ tlength);
+
+ ldebug("OUT adjust latency: %s\n",
+ g->conf.adjust_latency_out ? "yes" : "no");
+
+ pa->ss.format = audfmt_to_pa(as->fmt, as->endianness);
+ pa->ss.channels = as->nchannels;
+ pa->ss.rate = as->freq;
+
+ pa->ba.tlength = tlength * pa_frame_size(&pa->ss);
+ pa->ba.maxlength = -1;
+ pa->ba.minreq = -1;
+ pa->ba.prebuf = -1;
+
+ obt_as.fmt = pa_to_audfmt(pa->ss.format, &obt_as.endianness);
+
+ pa->stream = qpa_simple_new(
+ g,
+ "qemu",
+ PA_STREAM_PLAYBACK,
+ g->conf.sink,
+ &pa->ss,
+ NULL, /* channel map */
+ &pa->ba, /* buffering attributes */
+ &error
+ );
if (!pa->stream) {
qpa_logerr (error, "pa_simple_new for playback failed\n");
goto fail1;
}
- audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
- pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
- pa->rpos = hw->rpos;
- if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {
- goto fail3;
- }
+ audio_pcm_init_info(&hw->info, &obt_as);
+ hw->samples = buflen;
return 0;
- fail3:
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
- fail2:
- if (pa->stream) {
- pa_stream_unref (pa->stream);
- pa->stream = NULL;
- }
- fail1:
+fail1:
return -1;
}
static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
- pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
- ss.format = audfmt_to_pa (as->fmt, as->endianness);
- ss.channels = as->nchannels;
- ss.rate = as->freq;
-
- obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
-
- pa->stream = qpa_simple_new (
- g,
- "qemu",
- PA_STREAM_RECORD,
- g->conf.source,
- &ss,
- NULL, /* channel map */
- NULL, /* buffering attributes */
- &error
- );
+ int64_t timer_tick_duration =
+ audio_MAX(audio_get_timer_ticks(), 1 * SCALE_MS);
+ int64_t frames_per_tick_x1000 =
+ ((timer_tick_duration * as->freq * 1000LL) / NANOSECONDS_PER_SECOND);
+
+ int64_t fragsize = g->conf.fragsize;
+ if (fragsize == 0) {
+ fragsize = frames_per_tick_x1000 / 1000;
+ }
+ int64_t buflen = g->conf.buffer_size_in;
+ if (buflen == 0) {
+ buflen = frames_per_tick_x1000 / 400;
+ }
+ int64_t maxlength = g->conf.maxlength_in;
+ if (maxlength == 0) {
+ maxlength = fragsize * 2;
+ }
+
+ ldebug("IN internal buffer: %.2f ms (%"PRId64" frames)\n",
+ buflen * (1000.0f / as->freq),
+ buflen);
+
+ ldebug("IN fragsize: %.2f ms (%"PRId64" frames)\n",
+ fragsize * (1000.0f / as->freq),
+ fragsize);
+
+ ldebug("IN maxlength: %.2f ms (%"PRId64" frames)\n",
+ maxlength * (1000.0f / as->freq),
+ maxlength);
+
+ ldebug("IN adjust latency: %s\n",
+ g->conf.adjust_latency_in ? "yes" : "no");
+
+ pa->ss.format = audfmt_to_pa(as->fmt, as->endianness);
+ pa->ss.channels = as->nchannels;
+ pa->ss.rate = as->freq;
+
+ pa->ba.fragsize = fragsize * pa_frame_size(&pa->ss);
+ pa->ba.maxlength = maxlength * pa_frame_size(&pa->ss);
+ pa->ba.minreq = -1;
+ pa->ba.prebuf = -1;
+
+ obt_as.fmt = pa_to_audfmt(pa->ss.format, &obt_as.endianness);
+
+ pa->stream = qpa_simple_new(
+ g,
+ "qemu",
+ PA_STREAM_RECORD,
+ g->conf.source,
+ &pa->ss,
+ NULL, /* channel map */
+ &pa->ba, /* buffering attributes */
+ &error
+ );
if (!pa->stream) {
qpa_logerr (error, "pa_simple_new for capture failed\n");
goto fail1;
}
- audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
- pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
- pa->wpos = hw->wpos;
- if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
- goto fail3;
- }
+ audio_pcm_init_info(&hw->info, &obt_as);
+ hw->samples = buflen;
return 0;
- fail3:
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
- fail2:
- if (pa->stream) {
- pa_stream_unref (pa->stream);
- pa->stream = NULL;
- }
- fail1:
+ fail1:
return -1;
}
static void qpa_fini_out (HWVoiceOut *hw)
{
- void *ret;
PAVoiceOut *pa = (PAVoiceOut *) hw;
- audio_pt_lock (&pa->pt, AUDIO_FUNC);
- pa->done = 1;
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
-
if (pa->stream) {
pa_stream_unref (pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini (&pa->pt, AUDIO_FUNC);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static void qpa_fini_in (HWVoiceIn *hw)
{
- void *ret;
PAVoiceIn *pa = (PAVoiceIn *) hw;
- audio_pt_lock (&pa->pt, AUDIO_FUNC);
- pa->done = 1;
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
-
if (pa->stream) {
pa_stream_unref (pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini (&pa->pt, AUDIO_FUNC);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -809,7 +647,8 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
/* common */
static PAConf glob_conf = {
- .samples = 4096,
+ .adjust_latency_out = 0,
+ .adjust_latency_in = 1,
};
static void *qpa_audio_init (void)
@@ -897,10 +736,46 @@ static void qpa_audio_fini (void *opaque)
struct audio_option qpa_options[] = {
{
- .name = "SAMPLES",
+ .name = "BUFFER_SIZE_OUT",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_out,
+ .descr = "internal buffer size in frames for playback device"
+ },
+ {
+ .name = "BUFFER_SIZE_IN",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.buffer_size_in,
+ .descr = "internal buffer size in frames for recording device"
+ },
+ {
+ .name = "TLENGTH",
.tag = AUD_OPT_INT,
- .valp = &glob_conf.samples,
- .descr = "buffer size in samples"
+ .valp = &glob_conf.tlength,
+ .descr = "playback buffer target length in frames"
+ },
+ {
+ .name = "FRAGSIZE",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.fragsize,
+ .descr = "fragment length of recording device in frames"
+ },
+ {
+ .name = "MAXLENGTH_IN",
+ .tag = AUD_OPT_INT,
+ .valp = &glob_conf.maxlength_in,
+ .descr = "maximum length of PA recording buffer in frames"
+ },
+ {
+ .name = "ADJUST_LATENCY_OUT",
+ .tag = AUD_OPT_BOOL,
+ .valp = &glob_conf.adjust_latency_out,
+ .descr = "instruct PA to adjust latency for playback device"
+ },
+ {
+ .name = "ADJUST_LATENCY_IN",
+ .tag = AUD_OPT_BOOL,
+ .valp = &glob_conf.adjust_latency_in,
+ .descr = "instruct PA to adjust latency for recording device"
},
{
.name = "SERVER",
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 5402cd196c..ab89158bfc 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -18,6 +18,7 @@
*/
#include "qemu/osdep.h"
+#include "qemu/atomic.h"
#include "hw/hw.h"
#include "hw/pci/pci.h"
#include "intel-hda.h"
@@ -126,6 +127,11 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
#define PARAM nomixemu
#include "hda-codec-common.h"
+#define HDA_TIMER_TICKS (SCALE_MS)
+#define MAX_CORR (SCALE_US * 100)
+#define B_SIZE sizeof(st->buf)
+#define B_MASK (sizeof(st->buf) - 1)
+
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
@@ -154,8 +160,13 @@ struct HDAAudioStream {
SWVoiceIn *in;
SWVoiceOut *out;
} voice;
- uint8_t buf[HDA_BUFFER_SIZE];
- uint32_t bpos;
+ uint8_t compat_buf[HDA_BUFFER_SIZE];
+ uint32_t compat_bpos;
+ uint8_t buf[8192]; /* size must be power of two */
+ int64_t rpos;
+ int64_t wpos;
+ QEMUTimer *buft;
+ int64_t buft_start;
};
#define TYPE_HDA_AUDIO "hda-audio"
@@ -176,53 +187,146 @@ struct HDAAudioState {
bool mixer;
};
+static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
+{
+ return 2 * st->as.nchannels * st->as.freq;
+}
+
+static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
+{
+ int64_t corr =
+ NANOSECONDS_PER_SECOND * target_pos / hda_bytes_per_second(st);
+ if (corr > MAX_CORR) {
+ corr = MAX_CORR;
+ } else if (corr < -MAX_CORR) {
+ corr = -MAX_CORR;
+ }
+ atomic_fetch_add(&st->buft_start, corr);
+}
+
+static void hda_audio_input_timer(void *opaque)
+{
+ HDAAudioStream *st = opaque;
+
+ int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+
+ int64_t buft_start = atomic_fetch_add(&st->buft_start, 0);
+ int64_t wpos = atomic_fetch_add(&st->wpos, 0);
+ int64_t rpos = atomic_fetch_add(&st->rpos, 0);
+
+ int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
+ / NANOSECONDS_PER_SECOND;
+ wanted_rpos &= -4; /* IMPORTANT! clip to frames */
+
+ if (wanted_rpos <= rpos) {
+ /* we already transmitted the data */
+ goto out_timer;
+ }
+
+ int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
+ while (to_transfer) {
+ uint32_t start = (rpos & B_MASK);
+ uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+ int rc = hda_codec_xfer(
+ &st->state->hda, st->stream, false, st->buf + start, chunk);
+ if (!rc) {
+ break;
+ }
+ rpos += chunk;
+ to_transfer -= chunk;
+ atomic_fetch_add(&st->rpos, chunk);
+ }
+
+out_timer:
+
+ if (st->running) {
+ timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
+ }
+}
+
static void hda_audio_input_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
- int recv = 0;
- int len;
- bool rc;
-
- while (avail - recv >= sizeof(st->buf)) {
- if (st->bpos != sizeof(st->buf)) {
- len = AUD_read(st->voice.in, st->buf + st->bpos,
- sizeof(st->buf) - st->bpos);
- st->bpos += len;
- recv += len;
- if (st->bpos != sizeof(st->buf)) {
- break;
- }
+
+ int64_t wpos = atomic_fetch_add(&st->wpos, 0);
+ int64_t rpos = atomic_fetch_add(&st->rpos, 0);
+
+ int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
+
+ hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
+
+ while (to_transfer) {
+ uint32_t start = (uint32_t) (wpos & B_MASK);
+ uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
+ wpos += read;
+ to_transfer -= read;
+ atomic_fetch_add(&st->wpos, read);
+ if (chunk != read) {
+ break;
}
- rc = hda_codec_xfer(&st->state->hda, st->stream, false,
- st->buf, sizeof(st->buf));
+ }
+}
+
+static void hda_audio_output_timer(void *opaque)
+{
+ HDAAudioStream *st = opaque;
+
+ int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+
+ int64_t buft_start = atomic_fetch_add(&st->buft_start, 0);
+ int64_t wpos = atomic_fetch_add(&st->wpos, 0);
+ int64_t rpos = atomic_fetch_add(&st->rpos, 0);
+
+ int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
+ / NANOSECONDS_PER_SECOND;
+ wanted_wpos &= -4; /* IMPORTANT! clip to frames */
+
+ if (wanted_wpos <= wpos) {
+ /* we already received the data */
+ goto out_timer;
+ }
+
+ int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
+ while (to_transfer) {
+ uint32_t start = (wpos & B_MASK);
+ uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+ int rc = hda_codec_xfer(
+ &st->state->hda, st->stream, true, st->buf + start, chunk);
if (!rc) {
break;
}
- st->bpos = 0;
+ wpos += chunk;
+ to_transfer -= chunk;
+ atomic_fetch_add(&st->wpos, chunk);
+ }
+
+out_timer:
+
+ if (st->running) {
+ timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
}
}
static void hda_audio_output_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
- int sent = 0;
- int len;
- bool rc;
-
- while (avail - sent >= sizeof(st->buf)) {
- if (st->bpos == sizeof(st->buf)) {
- rc = hda_codec_xfer(&st->state->hda, st->stream, true,
- st->buf, sizeof(st->buf));
- if (!rc) {
- break;
- }
- st->bpos = 0;
- }
- len = AUD_write(st->voice.out, st->buf + st->bpos,
- sizeof(st->buf) - st->bpos);
- st->bpos += len;
- sent += len;
- if (st->bpos != sizeof(st->buf)) {
+
+ int64_t wpos = atomic_fetch_add(&st->wpos, 0);
+ int64_t rpos = atomic_fetch_add(&st->rpos, 0);
+
+ int64_t to_transfer = audio_MIN(wpos - rpos, avail);
+
+ hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1));
+
+ while (to_transfer) {
+ uint32_t start = (uint32_t) (rpos & B_MASK);
+ uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
+ rpos += written;
+ to_transfer -= written;
+ atomic_fetch_add(&st->rpos, written);
+ if (chunk != written) {
break;
}
}
@@ -239,6 +343,15 @@ static void hda_audio_set_running(HDAAudioStream *st, bool running)
st->running = running;
dprint(st->state, 1, "%s: %s (stream %d)\n", st->node->name,
st->running ? "on" : "off", st->stream);
+ if (running) {
+ int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ st->rpos = 0;
+ st->wpos = 0;
+ st->buft_start = now;
+ timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
+ } else {
+ timer_del(st->buft);
+ }
if (st->output) {
AUD_set_active_out(st->voice.out, st->running);
} else {
@@ -286,10 +399,12 @@ static void hda_audio_setup(HDAAudioStream *st)
st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
st->node->name, st,
hda_audio_output_cb, &st->as);
+ st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL, hda_audio_output_timer, st);
} else {
st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
st->node->name, st,
hda_audio_input_cb, &st->as);
+ st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL, hda_audio_input_timer, st);
}
}
@@ -505,7 +620,6 @@ static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
/* unmute output by default */
st->gain_left = QEMU_HDA_AMP_STEPS;
st->gain_right = QEMU_HDA_AMP_STEPS;
- st->bpos = sizeof(st->buf);
st->output = true;
} else {
st->output = false;
@@ -532,6 +646,7 @@ static void hda_audio_exit(HDACodecDevice *hda)
if (st->node == NULL) {
continue;
}
+ timer_del(st->buft);
if (st->output) {
AUD_close_out(&a->card, st->voice.out);
} else {
@@ -592,8 +707,8 @@ static const VMStateDescription vmstate_hda_audio_stream = {
VMSTATE_UINT32(gain_right, HDAAudioStream),
VMSTATE_BOOL(mute_left, HDAAudioStream),
VMSTATE_BOOL(mute_right, HDAAudioStream),
- VMSTATE_UINT32(bpos, HDAAudioStream),
- VMSTATE_BUFFER(buf, HDAAudioStream),
+ VMSTATE_UINT32(compat_bpos, HDAAudioStream),
+ VMSTATE_BUFFER(compat_buf, HDAAudioStream),
VMSTATE_END_OF_LIST()
}
};
diff --git a/hw/audio/intel-hda.c b/hw/audio/intel-hda.c
index a3e670c188..55f94f9afe 100644
--- a/hw/audio/intel-hda.c
+++ b/hw/audio/intel-hda.c
@@ -407,13 +407,6 @@ static bool intel_hda_xfer(HDACodecDevice *dev, uint32_t stnr, bool output,
if (st->bpl == NULL) {
return false;
}
- if (st->ctl & (1 << 26)) {
- /*
- * Wait with the next DMA xfer until the guest
- * has acked the buffer completion interrupt
- */
- return false;
- }
left = len;
s = st->bentries;
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