* [PATCH v4 01/24] audio: api for mixeng code free backends
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 02/24] alsaaudio: port to the new audio backend api Kővágó, Zoltán
` (19 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
Changes from v2:
* audio_pcm_hw_run_out: handle NULL result from get_buffer_out
audio/audio.c | 216 ++++++++++++++++++++++++++++++++++++++++-
audio/audio_int.h | 43 ++++++--
audio/audio_template.h | 1 +
3 files changed, 251 insertions(+), 9 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e99fcd0694..e29a1e15eb 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -573,6 +573,25 @@ size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
return clipped;
}
+static void audio_pcm_hw_clip_out2(HWVoiceOut *hw, void *pcm_buf, size_t len)
+{
+ size_t clipped = 0;
+ size_t pos = hw->rpos;
+
+ while (len) {
+ st_sample *src = hw->mix_buf + pos;
+ uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+ size_t samples_till_end_of_buf = hw->samples - pos;
+ size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
+
+ hw->clip(dst, src, samples_to_clip);
+
+ pos = (pos + samples_to_clip) % hw->samples;
+ len -= samples_to_clip;
+ clipped += samples_to_clip;
+ }
+}
+
/*
* Soft voice (capture)
*/
@@ -1050,6 +1069,36 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
mixeng_clear(hw->mix_buf, samples - n);
}
+static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
+{
+ size_t clipped = 0;
+
+ while (live) {
+ size_t size, decr, proc;
+ void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
+ if (!buf) {
+ /* retrying will likely won't help, drop everything. */
+ hw->rpos = (hw->rpos + live) % hw->samples;
+ return clipped + live;
+ }
+
+ decr = MIN(size >> hw->info.shift, live);
+ audio_pcm_hw_clip_out2(hw, buf, decr);
+ proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
+ hw->info.shift;
+
+ live -= proc;
+ clipped += proc;
+ hw->rpos = (hw->rpos + proc) % hw->samples;
+
+ if (proc == 0 || proc < decr) {
+ break;
+ }
+ }
+
+ return clipped;
+}
+
static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
@@ -1097,7 +1146,11 @@ static void audio_run_out (AudioState *s)
}
prev_rpos = hw->rpos;
- played = hw->pcm_ops->run_out (hw, live);
+ if (hw->pcm_ops->run_out) {
+ played = hw->pcm_ops->run_out(hw, live);
+ } else {
+ played = audio_pcm_hw_run_out(hw, live);
+ }
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
@@ -1156,6 +1209,35 @@ static void audio_run_out (AudioState *s)
}
}
+static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
+{
+ size_t conv = 0;
+
+ while (samples) {
+ size_t proc;
+ size_t size = samples << hw->info.shift;
+ void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
+
+ assert((size & hw->info.align) == 0);
+ if (size == 0) {
+ hw->pcm_ops->put_buffer_in(hw, buf, size);
+ break;
+ }
+
+ proc = MIN(size >> hw->info.shift,
+ hw->samples - hw->wpos);
+
+ hw->conv(hw->conv_buf + hw->wpos, buf, proc);
+ hw->wpos = (hw->wpos + proc) % hw->samples;
+
+ samples -= proc;
+ conv += proc;
+ hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+ }
+
+ return conv;
+}
+
static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
@@ -1165,7 +1247,12 @@ static void audio_run_in (AudioState *s)
size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
- captured = hw->pcm_ops->run_in(hw);
+ if (hw->pcm_ops->run_in) {
+ captured = hw->pcm_ops->run_in(hw);
+ } else {
+ captured = audio_pcm_hw_run_in(
+ hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+ }
}
replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
@@ -1259,12 +1346,137 @@ void audio_run(AudioState *s, const char *msg)
#endif
}
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+ ssize_t start;
+
+ if (unlikely(!hw->buf_emul)) {
+ size_t calc_size = hw->samples << hw->info.shift;
+ hw->buf_emul = g_malloc(calc_size);
+ hw->size_emul = calc_size;
+ hw->pos_emul = hw->pending_emul = 0;
+ }
+
+ while (hw->pending_emul < hw->size_emul) {
+ size_t read_len = MIN(hw->size_emul - hw->pos_emul,
+ hw->size_emul - hw->pending_emul);
+ size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
+ read_len);
+ hw->pending_emul += read;
+ if (read < read_len) {
+ break;
+ }
+ }
+
+ start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
+
+ *size = MIN(hw->pending_emul, hw->size_emul - start);
+ return hw->buf_emul + start;
+}
+
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+ assert(size <= hw->pending_emul);
+ hw->pending_emul -= size;
+}
+
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ if (unlikely(!hw->buf_emul)) {
+ size_t calc_size = hw->samples << hw->info.shift;
+
+ hw->buf_emul = g_malloc(calc_size);
+ hw->size_emul = calc_size;
+ hw->pos_emul = hw->pending_emul = 0;
+ }
+
+ *size = MIN(hw->size_emul - hw->pending_emul,
+ hw->size_emul - hw->pos_emul);
+ return hw->buf_emul + hw->pos_emul;
+}
+
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+ size_t size)
+{
+ assert(buf == hw->buf_emul + hw->pos_emul &&
+ size + hw->pending_emul <= hw->size_emul);
+
+ hw->pending_emul += size;
+ hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+
+ return size;
+}
+
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+ audio_generic_put_buffer_out_nowrite(hw, buf, size);
+
+ while (hw->pending_emul) {
+ size_t write_len, written;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
+
+ write_len = MIN(hw->pending_emul, hw->size_emul - start);
+
+ written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
+ hw->pending_emul -= written;
+
+ if (written < write_len) {
+ break;
+ }
+ }
+
+ /*
+ * fake we have written everything. non-written data remain in pending_emul,
+ * so we do not have to clip them multiple times
+ */
+ return size;
+}
+
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
+{
+ size_t dst_size, copy_size;
+ void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
+ copy_size = MIN(size, dst_size);
+
+ memcpy(dst, buf, copy_size);
+ return hw->pcm_ops->put_buffer_out(hw, buf, copy_size);
+}
+
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
+{
+ size_t dst_size, copy_size;
+ void *dst = hw->pcm_ops->get_buffer_in(hw, &dst_size);
+ copy_size = MIN(size, dst_size);
+
+ memcpy(dst, buf, copy_size);
+ hw->pcm_ops->put_buffer_in(hw, buf, copy_size);
+ return copy_size;
+}
+
+
static int audio_driver_init(AudioState *s, struct audio_driver *drv,
bool msg, Audiodev *dev)
{
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
+ if (!drv->pcm_ops->get_buffer_in) {
+ drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
+ drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
+ }
+ if (!drv->pcm_ops->get_buffer_out) {
+ drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
+ drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
+ }
+
audio_init_nb_voices_out(s, drv);
audio_init_nb_voices_in(s, drv);
s->drv = drv;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index a674c5374a..8fb1ca8a8d 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -65,6 +65,8 @@ typedef struct HWVoiceOut {
uint64_t ts_helper;
struct st_sample *mix_buf;
+ void *buf_emul;
+ size_t pos_emul, pending_emul, size_emul;
size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
@@ -87,6 +89,8 @@ typedef struct HWVoiceIn {
uint64_t ts_helper;
struct st_sample *conv_buf;
+ void *buf_emul;
+ size_t pos_emul, pending_emul, size_emul;
size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
@@ -147,17 +151,42 @@ struct audio_driver {
};
struct audio_pcm_ops {
- int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
- void (*fini_out)(HWVoiceOut *hw);
+ int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
+ void (*fini_out)(HWVoiceOut *hw);
size_t (*run_out)(HWVoiceOut *hw, size_t live);
- int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
-
- int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
- void (*fini_in) (HWVoiceIn *hw);
+ size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
+ /*
+ * get a buffer that after later can be passed to put_buffer_out; optional
+ * returns the buffer, and writes it's size to size (in bytes)
+ * this is unrelated to the above buffer_size_out function
+ */
+ void *(*get_buffer_out)(HWVoiceOut *hw, size_t *size);
+ /*
+ * put back the buffer returned by get_buffer_out; optional
+ * buf must be equal the pointer returned by get_buffer_out,
+ * size may be smaller
+ */
+ size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
+ int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+
+ int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
+ void (*fini_in) (HWVoiceIn *hw);
size_t (*run_in)(HWVoiceIn *hw);
- int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
+ size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
+ void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
+ void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
+ int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+ size_t size);
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size);
+
struct capture_callback {
struct audio_capture_ops ops;
void *opaque;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 2562bf5f00..ff4a173f18 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -71,6 +71,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
{
+ g_free(hw->buf_emul);
g_free (HWBUF);
HWBUF = NULL;
}
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 02/24] alsaaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 01/24] audio: api for mixeng code free backends Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 03/24] coreaudio: " Kővágó, Zoltán
` (18 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 302 ++++++++++++----------------------------------
1 file changed, 80 insertions(+), 222 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 591344dccd..19124d09d8 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -44,9 +44,6 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
- int wpos;
- int pending;
- void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
Audiodev *dev;
@@ -55,7 +52,6 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
- void *pcm_buf;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -602,102 +598,64 @@ static int alsa_open(bool in, struct alsa_params_req *req,
return -1;
}
-static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
+static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
- snd_pcm_sframes_t avail;
-
- avail = snd_pcm_avail_update (handle);
- if (avail < 0) {
- if (avail == -EPIPE) {
- if (!alsa_recover (handle)) {
- avail = snd_pcm_avail_update (handle);
- }
- }
-
- if (avail < 0) {
- alsa_logerr (avail,
- "Could not obtain number of available frames\n");
- return -1;
- }
- }
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ size_t pos = 0;
+ size_t len_frames = len >> hw->info.shift;
+
+ while (len_frames) {
+ char *src = advance(buf, pos);
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei(alsa->handle, src, len_frames);
+
+ if (written <= 0) {
+ switch (written) {
+ case 0:
+ trace_alsa_wrote_zero(len_frames);
+ return pos;
+
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(written, "Failed to write %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ trace_alsa_xrun_out();
+ continue;
+
+ case -ESTRPIPE:
+ /*
+ * stream is suspended and waiting for an application
+ * recovery
+ */
+ if (alsa_resume(alsa->handle)) {
+ alsa_logerr(written, "Failed to write %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ trace_alsa_resume_out();
+ continue;
- return avail;
-}
+ case -EAGAIN:
+ return pos;
-static void alsa_write_pending (ALSAVoiceOut *alsa)
-{
- HWVoiceOut *hw = &alsa->hw;
-
- while (alsa->pending) {
- int left_till_end_samples = hw->samples - alsa->wpos;
- int len = MIN (alsa->pending, left_till_end_samples);
- char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
-
- while (len) {
- snd_pcm_sframes_t written;
-
- written = snd_pcm_writei (alsa->handle, src, len);
-
- if (written <= 0) {
- switch (written) {
- case 0:
- trace_alsa_wrote_zero(len);
- return;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (written, "Failed to write %d frames\n",
- len);
- return;
- }
- trace_alsa_xrun_out();
- continue;
-
- case -ESTRPIPE:
- /* stream is suspended and waiting for an
- application recovery */
- if (alsa_resume (alsa->handle)) {
- alsa_logerr (written, "Failed to write %d frames\n",
- len);
- return;
- }
- trace_alsa_resume_out();
- continue;
-
- case -EAGAIN:
- return;
-
- default:
- alsa_logerr (written, "Failed to write %d frames from %p\n",
- len, src);
- return;
- }
+ default:
+ alsa_logerr(written, "Failed to write %zu frames from %p\n",
+ len, src);
+ return pos;
}
-
- alsa->wpos = (alsa->wpos + written) % hw->samples;
- alsa->pending -= written;
- len -= written;
}
- }
-}
-
-static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
-{
- ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- size_t decr;
- snd_pcm_sframes_t avail;
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of available playback frames\n");
- return 0;
+ pos += written << hw->info.shift;
+ if (written < len_frames) {
+ break;
+ }
+ len_frames -= written;
}
- decr = MIN (live, avail);
- decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
- alsa->pending += decr;
- alsa_write_pending (alsa);
- return decr;
+ return pos;
}
static void alsa_fini_out (HWVoiceOut *hw)
@@ -706,9 +664,6 @@ static void alsa_fini_out (HWVoiceOut *hw)
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
@@ -737,14 +692,6 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
- if (!alsa->pcm_buf) {
- dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- alsa_anal_close1 (&handle);
- return -1;
- }
-
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
@@ -839,14 +786,6 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!alsa->pcm_buf) {
- dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- alsa_anal_close1 (&handle);
- return -1;
- }
-
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
@@ -858,129 +797,48 @@ static void alsa_fini_in (HWVoiceIn *hw)
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
-static size_t alsa_run_in(HWVoiceIn *hw)
+static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- int hwshift = hw->info.shift;
- int i;
- size_t live = audio_pcm_hw_get_live_in (hw);
- size_t dead = hw->samples - live;
- size_t decr;
- struct {
- size_t add;
- size_t len;
- } bufs[2] = {
- { .add = hw->wpos, .len = 0 },
- { .add = 0, .len = 0 }
- };
- snd_pcm_sframes_t avail;
- snd_pcm_uframes_t read_samples = 0;
-
- if (!dead) {
- return 0;
- }
+ size_t pos = 0;
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of captured frames\n");
- return 0;
- }
-
- if (!avail) {
- snd_pcm_state_t state;
-
- state = snd_pcm_state (alsa->handle);
- switch (state) {
- case SND_PCM_STATE_PREPARED:
- avail = hw->samples;
- break;
- case SND_PCM_STATE_SUSPENDED:
- /* stream is suspended and waiting for an application recovery */
- if (alsa_resume (alsa->handle)) {
- dolog ("Failed to resume suspended input stream\n");
- return 0;
- }
- trace_alsa_resume_in();
- break;
- default:
- trace_alsa_no_frames(state);
- return 0;
- }
- }
+ while (len) {
+ void *dst = advance(buf, pos);
+ snd_pcm_sframes_t nread;
- decr = MIN(dead, avail);
- if (!decr) {
- return 0;
- }
+ nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
- if (hw->wpos + decr > hw->samples) {
- bufs[0].len = (hw->samples - hw->wpos);
- bufs[1].len = (decr - (hw->samples - hw->wpos));
- }
- else {
- bufs[0].len = decr;
- }
+ if (nread <= 0) {
+ switch (nread) {
+ case 0:
+ trace_alsa_read_zero(len);
+ return pos;;
- for (i = 0; i < 2; ++i) {
- void *src;
- struct st_sample *dst;
- snd_pcm_sframes_t nread;
- snd_pcm_uframes_t len;
-
- len = bufs[i].len;
-
- src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
- dst = hw->conv_buf + bufs[i].add;
-
- while (len) {
- nread = snd_pcm_readi (alsa->handle, src, len);
-
- if (nread <= 0) {
- switch (nread) {
- case 0:
- trace_alsa_read_zero(len);
- goto exit;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (nread, "Failed to read %ld frames\n", len);
- goto exit;
- }
- trace_alsa_xrun_in();
- continue;
-
- case -EAGAIN:
- goto exit;
-
- default:
- alsa_logerr (
- nread,
- "Failed to read %ld frames from %p\n",
- len,
- src
- );
- goto exit;
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(nread, "Failed to read %zu frames\n", len);
+ return pos;
}
- }
+ trace_alsa_xrun_in();
+ continue;
- hw->conv (dst, src, nread);
+ case -EAGAIN:
+ return pos;
- src = advance (src, nread << hwshift);
- dst += nread;
-
- read_samples += nread;
- len -= nread;
+ default:
+ alsa_logerr(nread, "Failed to read %zu frames to %p\n",
+ len, dst);
+ return pos;;
+ }
}
+
+ pos += nread << hw->info.shift;
+ len -= nread << hw->info.shift;
}
- exit:
- hw->wpos = (hw->wpos + read_samples) % hw->samples;
- return read_samples;
+ return pos;
}
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -1065,12 +923,12 @@ static void alsa_audio_fini (void *opaque)
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
- .run_out = alsa_run_out,
+ .write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
- .run_in = alsa_run_in,
+ .read = alsa_read,
.ctl_in = alsa_ctl_in,
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 03/24] coreaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 01/24] audio: api for mixeng code free backends Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 02/24] alsaaudio: port to the new audio backend api Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 04/24] dsoundaudio: " Kővágó, Zoltán
` (17 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/coreaudio.c | 130 ++++++++++++++++++++++++----------------------
1 file changed, 69 insertions(+), 61 deletions(-)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index d1be58b40a..5cde42f982 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,9 +43,6 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
- size_t live;
- size_t decr;
- size_t rpos;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -397,31 +394,29 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
return 0;
}
-static size_t coreaudio_run_out(HWVoiceOut *hw, size_t live)
-{
- size_t decr;
- coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
-
- if (coreaudio_lock (core, "coreaudio_run_out")) {
- return 0;
+#define COREAUDIO_WRAPPER_FUNC(name, ret_type, args_decl, args) \
+ static ret_type glue(coreaudio_, name)args_decl \
+ { \
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw; \
+ ret_type ret; \
+ \
+ if (coreaudio_lock(core, "coreaudio_" #name)) { \
+ return 0; \
+ } \
+ \
+ ret = glue(audio_generic_, name)args; \
+ \
+ coreaudio_unlock(core, "coreaudio_" #name); \
+ return ret; \
}
-
- if (core->decr > live) {
- ldebug ("core->decr %d live %d core->live %d\n",
- core->decr,
- live,
- core->live);
- }
-
- decr = MIN (core->decr, live);
- core->decr -= decr;
-
- core->live = live - decr;
- hw->rpos = core->rpos;
-
- coreaudio_unlock (core, "coreaudio_run_out");
- return decr;
-}
+COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
+ (hw, size))
+COREAUDIO_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size),
+ (hw, buf, size))
+COREAUDIO_WRAPPER_FUNC(write, size_t, (HWVoiceOut *hw, void *buf, size_t size),
+ (hw, buf, size))
+#undef COREAUDIO_WRAPPER_FUNC
/* callback to feed audiooutput buffer */
static OSStatus audioDeviceIOProc(
@@ -433,19 +428,11 @@ static OSStatus audioDeviceIOProc(
const AudioTimeStamp* inOutputTime,
void* hwptr)
{
- UInt32 frame, frameCount;
- float *out = outOutputData->mBuffers[0].mData;
+ UInt32 frameCount, pending_frames;
+ void *out = outOutputData->mBuffers[0].mData;
HWVoiceOut *hw = hwptr;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
- int rpos, live;
- struct st_sample *src;
-#ifndef FLOAT_MIXENG
-#ifdef RECIPROCAL
- const float scale = 1.f / UINT_MAX;
-#else
- const float scale = UINT_MAX;
-#endif
-#endif
+ size_t len;
if (coreaudio_lock (core, "audioDeviceIOProc")) {
inInputTime = 0;
@@ -453,42 +440,51 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
- live = core->live;
+ pending_frames = hw->pending_emul >> hw->info.shift;
/* if there are not enough samples, set signal and return */
- if (live < frameCount) {
+ if (pending_frames < frameCount) {
inInputTime = 0;
coreaudio_unlock (core, "audioDeviceIOProc(empty)");
return 0;
}
- rpos = core->rpos;
- src = hw->mix_buf + rpos;
+ len = frameCount << hw->info.shift;
+ while (len) {
+ size_t write_len;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
- /* fill buffer */
- for (frame = 0; frame < frameCount; frame++) {
-#ifdef FLOAT_MIXENG
- *out++ = src[frame].l; /* left channel */
- *out++ = src[frame].r; /* right channel */
-#else
-#ifdef RECIPROCAL
- *out++ = src[frame].l * scale; /* left channel */
- *out++ = src[frame].r * scale; /* right channel */
-#else
- *out++ = src[frame].l / scale; /* left channel */
- *out++ = src[frame].r / scale; /* right channel */
-#endif
-#endif
- }
+ write_len = MIN(MIN(hw->pending_emul, len),
+ hw->size_emul - start);
- rpos = (rpos + frameCount) % hw->samples;
- core->decr += frameCount;
- core->rpos = rpos;
+ memcpy(out, hw->buf_emul + start, write_len);
+ hw->pending_emul -= write_len;
+ len -= write_len;
+ out += write_len;
+ }
coreaudio_unlock (core, "audioDeviceIOProc");
return 0;
}
+static UInt32 coreaudio_get_flags(struct audio_pcm_info *info,
+ struct audsettings *as)
+{
+ UInt32 flags = info->sign ? kAudioFormatFlagIsSignedInteger : 0;
+ if (as->endianness) { /* 0 = little, 1 = big */
+ flags |= kAudioFormatFlagIsBigEndian;
+ }
+
+ if (flags == 0) { /* must not be 0 */
+ flags = kAudioFormatFlagsAreAllClear;
+ }
+ return flags;
+}
+
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
@@ -576,6 +572,16 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
/* set Samplerate */
core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
+ core->outputStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
+ core->outputStreamBasicDescription.mFormatFlags =
+ coreaudio_get_flags(&hw->info, as);
+ core->outputStreamBasicDescription.mBytesPerPacket =
+ core->outputStreamBasicDescription.mBytesPerFrame =
+ hw->info.nchannels * hw->info.bits / 8;
+ core->outputStreamBasicDescription.mFramesPerPacket = 1;
+ core->outputStreamBasicDescription.mChannelsPerFrame = hw->info.nchannels;
+ core->outputStreamBasicDescription.mBitsPerChannel = hw->info.bits;
+
status = coreaudio_set_streamformat(core->outputDeviceID,
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
@@ -686,7 +692,9 @@ static void coreaudio_audio_fini (void *opaque)
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
- .run_out = coreaudio_run_out,
+ .write = coreaudio_write,
+ .get_buffer_out = coreaudio_get_buffer_out,
+ .put_buffer_out = coreaudio_put_buffer_out_nowrite,
.ctl_out = coreaudio_ctl_out
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 04/24] dsoundaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (2 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 03/24] coreaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 05/24] noaudio: " Kővágó, Zoltán
` (16 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/dsound_template.h | 47 ++++---
audio/dsoundaudio.c | 305 +++++++++-------------------------------
2 files changed, 91 insertions(+), 261 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 8ece870c9e..9f10b688df 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -29,6 +29,8 @@
#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
#define FIELD dsound_capture_buffer
#define FIELD2 dsound_capture
+#define HWVOICE HWVoiceIn
+#define DSOUNDVOICE DSoundVoiceIn
#else
#define NAME "playback buffer"
#define NAME2 "DirectSound"
@@ -37,6 +39,8 @@
#define BUFPTR LPDIRECTSOUNDBUFFER
#define FIELD dsound_buffer
#define FIELD2 dsound
+#define HWVOICE HWVoiceOut
+#define DSOUNDVOICE DSoundVoiceOut
#endif
static int glue (dsound_unlock_, TYPE) (
@@ -72,8 +76,6 @@ static int glue (dsound_lock_, TYPE) (
)
{
HRESULT hr;
- LPVOID p1 = NULL, p2 = NULL;
- DWORD blen1 = 0, blen2 = 0;
DWORD flag;
#ifdef DSBTYPE_IN
@@ -81,7 +83,7 @@ static int glue (dsound_lock_, TYPE) (
#else
flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
#endif
- hr = glue(IFACE, _Lock)(buf, pos, len, &p1, &blen1, &p2, &blen2, flag);
+ hr = glue(IFACE, _Lock)(buf, pos, len, p1p, blen1p, p2p, blen2p, flag);
if (FAILED (hr)) {
#ifndef DSBTYPE_IN
@@ -96,34 +98,34 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
- if ((p1 && (blen1 & info->align)) || (p2 && (blen2 & info->align))) {
- dolog ("DirectSound returned misaligned buffer %ld %ld\n",
- blen1, blen2);
- glue (dsound_unlock_, TYPE) (buf, p1, p2, blen1, blen2);
+ if ((p1p && *p1p && (*blen1p & info->align)) ||
+ (p2p && *p2p && (*blen2p & info->align))) {
+ dolog("DirectSound returned misaligned buffer %ld %ld\n",
+ *blen1p, *blen2p);
+ glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
+ blen2p ? *blen2p : 0);
goto fail;
}
- if (!p1 && blen1) {
- dolog ("warning: !p1 && blen1=%ld\n", blen1);
- blen1 = 0;
+ if (p1p && !*p1p && *blen1p) {
+ dolog("warning: !p1 && blen1=%ld\n", *blen1p);
+ *blen1p = 0;
}
- if (!p2 && blen2) {
- dolog ("warning: !p2 && blen2=%ld\n", blen2);
- blen2 = 0;
+ if (p2p && !*p2p && *blen2p) {
+ dolog("warning: !p2 && blen2=%ld\n", *blen2p);
+ *blen2p = 0;
}
- *p1p = p1;
- *p2p = p2;
- *blen1p = blen1;
- *blen2p = blen2;
return 0;
fail:
*p1p = NULL - 1;
- *p2p = NULL - 1;
*blen1p = -1;
- *blen2p = -1;
+ if (p2p) {
+ *p2p = NULL - 1;
+ *blen2p = -1;
+ }
return -1;
}
@@ -242,7 +244,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
goto fail0;
}
- ds->first_time = 1;
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
@@ -252,15 +253,13 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
bc.dwBufferBytes, hw->info.align + 1
);
}
+ hw->size_emul = bc.dwBufferBytes;
hw->samples = bc.dwBufferBytes >> hw->info.shift;
ds->s = s;
#ifdef DEBUG_DSOUND
dolog ("caps %ld, desc %ld\n",
bc.dwBufferBytes, bd.dwBufferBytes);
-
- dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
- hw->bufsize, settings.freq, settings.nchannels, settings.fmt);
#endif
return 0;
@@ -276,3 +275,5 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
#undef BUFPTR
#undef FIELD
#undef FIELD2
+#undef HWVOICE
+#undef DSOUNDVOICE
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 2fc118b795..9960247814 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -53,19 +53,11 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
- DWORD old_pos;
- int first_time;
dsound *s;
-#ifdef DEBUG_DSOUND
- DWORD old_ppos;
- DWORD played;
- DWORD mixed;
-#endif
} DSoundVoiceOut;
typedef struct {
HWVoiceIn hw;
- int first_time;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
dsound *s;
} DSoundVoiceIn;
@@ -243,11 +235,6 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
-static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
-{
- return muldiv64(usecs, info->bytes_per_second, 1000000);
-}
-
#ifdef DEBUG_DSOUND
static void print_wave_format (WAVEFORMATEX *wfx)
{
@@ -312,33 +299,6 @@ static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
return 0;
}
-static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
-{
- int src_len1 = dst_len;
- int src_len2 = 0;
- int pos = hw->rpos + dst_len;
- struct st_sample *src1 = hw->mix_buf + hw->rpos;
- struct st_sample *src2 = NULL;
-
- if (pos > hw->samples) {
- src_len1 = hw->samples - hw->rpos;
- src2 = hw->mix_buf;
- src_len2 = dst_len - src_len1;
- pos = src_len2;
- }
-
- if (src_len1) {
- hw->clip (dst, src1, src_len1);
- }
-
- if (src_len2) {
- dst = advance (dst, src_len1 << hw->info.shift);
- hw->clip (dst, src2, src_len2);
- }
-
- hw->rpos = pos % hw->samples;
-}
-
static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
dsound *s)
{
@@ -350,7 +310,7 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
dsb,
&hw->info,
0,
- hw->samples << hw->info.shift,
+ hw->size_emul,
&p1, &p2,
&blen1, &blen2,
1,
@@ -454,139 +414,51 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static size_t dsound_run_out(HWVoiceOut *hw, size_t live)
+static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
- int err;
- HRESULT hr;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
- size_t len;
- int hwshift;
- DWORD blen1, blen2;
- DWORD len1, len2;
- DWORD decr;
- DWORD wpos, ppos, old_pos;
- LPVOID p1, p2;
- size_t bufsize;
- dsound *s = ds->s;
- AudiodevDsoundOptions *dso = &s->dev->u.dsound;
-
- if (!dsb) {
- dolog ("Attempt to run empty with playback buffer\n");
- return 0;
- }
-
- hwshift = hw->info.shift;
- bufsize = hw->samples << hwshift;
-
- hr = IDirectSoundBuffer_GetCurrentPosition (
- dsb,
- &ppos,
- ds->first_time ? &wpos : NULL
- );
- if (FAILED (hr)) {
- dsound_logerr (hr, "Could not get playback buffer position\n");
- return 0;
- }
-
- len = live << hwshift;
-
- if (ds->first_time) {
- if (dso->latency) {
- DWORD cur_blat;
+ HRESULT hr;
+ DWORD ppos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
- cur_blat = audio_ring_dist (wpos, ppos, bufsize);
- ds->first_time = 0;
- old_pos = wpos;
- old_pos +=
- usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
- old_pos %= bufsize;
- old_pos &= ~hw->info.align;
- }
- else {
- old_pos = wpos;
- }
-#ifdef DEBUG_DSOUND
- ds->played = 0;
- ds->mixed = 0;
-#endif
+ hr = IDirectSoundBuffer_GetCurrentPosition(dsb, &ppos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get playback buffer position\n");
+ *size = 0;
+ return NULL;
}
- else {
- if (ds->old_pos == ppos) {
-#ifdef DEBUG_DSOUND
- dolog ("old_pos == ppos\n");
-#endif
- return 0;
- }
-#ifdef DEBUG_DSOUND
- ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
-#endif
- old_pos = ds->old_pos;
- }
+ req_size = audio_ring_dist(ppos, hw->pos_emul, hw->size_emul);
+ req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
- if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
- len = ppos - old_pos;
- }
- else {
- if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
- len = bufsize - old_pos + ppos;
- }
+ err = dsound_lock_out(dsb, &hw->info, hw->pos_emul, req_size, &ret, NULL,
+ &act_size, NULL, false, ds->s);
+ if (err) {
+ dolog("Failed to lock buffer\n");
+ *size = 0;
+ return NULL;
}
- if (audio_bug(__func__, len > bufsize)) {
- dolog("len=%zu bufsize=%zu old_pos=%ld ppos=%ld\n",
- len, bufsize, old_pos, ppos);
- return 0;
- }
+ *size = act_size;
+ return ret;
+}
- len &= ~hw->info.align;
- if (!len) {
- return 0;
- }
+static size_t dsound_put_buffer_out(HWVoiceOut *hw, void *buf, size_t len)
+{
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+ int err = dsound_unlock_out(dsb, buf, NULL, len, 0);
-#ifdef DEBUG_DSOUND
- ds->old_ppos = ppos;
-#endif
- err = dsound_lock_out (
- dsb,
- &hw->info,
- old_pos,
- len,
- &p1, &p2,
- &blen1, &blen2,
- 0,
- s
- );
if (err) {
+ dolog("Failed to unlock buffer!!\n");
return 0;
}
+ hw->pos_emul = (hw->pos_emul + len) % hw->size_emul;
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
-
- if (p1 && len1) {
- dsound_write_sample (hw, p1, len1);
- }
-
- if (p2 && len2) {
- dsound_write_sample (hw, p2, len2);
- }
-
- dsound_unlock_out (dsb, p1, p2, blen1, blen2);
- ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
-
-#ifdef DEBUG_DSOUND
- ds->mixed += decr << hwshift;
-
- dolog ("played %lu mixed %lu diff %ld sec %f\n",
- ds->played,
- ds->mixed,
- ds->mixed - ds->played,
- abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
-#endif
- return decr;
+ return len;
}
static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -641,96 +513,49 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static size_t dsound_run_in(HWVoiceIn *hw)
+static void *dsound_get_buffer_in(HWVoiceIn *hw, size_t *size)
{
- int err;
- HRESULT hr;
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
- size_t live, len, dead;
- DWORD blen1, blen2;
- DWORD len1, len2;
- DWORD decr;
- DWORD cpos, rpos;
- LPVOID p1, p2;
- int hwshift;
- dsound *s = ds->s;
-
- if (!dscb) {
- dolog ("Attempt to run without capture buffer\n");
- return 0;
- }
-
- hwshift = hw->info.shift;
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- if (!dead) {
- return 0;
- }
-
- hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
- dscb,
- &cpos,
- ds->first_time ? &rpos : NULL
- );
- if (FAILED (hr)) {
- dsound_logerr (hr, "Could not get capture buffer position\n");
- return 0;
- }
-
- if (ds->first_time) {
- ds->first_time = 0;
- if (rpos & hw->info.align) {
- ldebug ("warning: Misaligned capture read position %ld(%d)\n",
- rpos, hw->info.align);
- }
- hw->wpos = rpos >> hwshift;
- }
+ HRESULT hr;
+ DWORD cpos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
- if (cpos & hw->info.align) {
- ldebug ("warning: Misaligned capture position %ld(%d)\n",
- cpos, hw->info.align);
+ hr = IDirectSoundCaptureBuffer_GetCurrentPosition(dscb, &cpos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get capture buffer position\n");
+ *size = 0;
+ return NULL;
}
- cpos >>= hwshift;
- len = audio_ring_dist (cpos, hw->wpos, hw->samples);
- if (!len) {
- return 0;
- }
- len = MIN (len, dead);
+ req_size = audio_ring_dist(cpos, hw->pos_emul, hw->size_emul);
+ req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
- err = dsound_lock_in (
- dscb,
- &hw->info,
- hw->wpos << hwshift,
- len << hwshift,
- &p1,
- &p2,
- &blen1,
- &blen2,
- 0,
- s
- );
+ err = dsound_lock_in(dscb, &hw->info, hw->pos_emul, req_size, &ret, NULL,
+ &act_size, NULL, false, ds->s);
if (err) {
- return 0;
+ dolog("Failed to lock buffer\n");
+ *size = 0;
+ return NULL;
}
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
+ *size = act_size;
+ return ret;
+}
- if (p1 && len1) {
- hw->conv (hw->conv_buf + hw->wpos, p1, len1);
- }
+static void dsound_put_buffer_in(HWVoiceIn *hw, void *buf, size_t len)
+{
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
+ int err = dsound_unlock_in(dscb, buf, NULL, len, 0);
- if (p2 && len2) {
- hw->conv (hw->conv_buf, p2, len2);
+ if (err) {
+ dolog("Failed to unlock buffer!!\n");
+ return;
}
-
- dsound_unlock_in (dscb, p1, p2, blen1, blen2);
- hw->wpos = (hw->wpos + decr) % hw->samples;
- return decr;
+ hw->pos_emul = (hw->pos_emul + len) % hw->size_emul;
}
static void dsound_audio_fini (void *opaque)
@@ -846,12 +671,16 @@ static void *dsound_audio_init(Audiodev *dev)
static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
- .run_out = dsound_run_out,
+ .write = audio_generic_write,
+ .get_buffer_out = dsound_get_buffer_out,
+ .put_buffer_out = dsound_put_buffer_out,
.ctl_out = dsound_ctl_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
- .run_in = dsound_run_in,
+ .read = audio_generic_read,
+ .get_buffer_in = dsound_get_buffer_in,
+ .put_buffer_in = dsound_put_buffer_in,
.ctl_in = dsound_ctl_in
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 05/24] noaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (3 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 04/24] dsoundaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 06/24] ossaudio: " Kővágó, Zoltán
` (15 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/noaudio.c | 43 +++++++++++++++++--------------------------
1 file changed, 17 insertions(+), 26 deletions(-)
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 0fb2629cf2..b054fd225b 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -41,10 +41,9 @@ typedef struct NoVoiceIn {
int64_t old_ticks;
} NoVoiceIn;
-static size_t no_run_out(HWVoiceOut *hw, size_t live)
+static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- size_t decr, samples;
int64_t now;
int64_t ticks;
int64_t bytes;
@@ -52,13 +51,9 @@ static size_t no_run_out(HWVoiceOut *hw, size_t live)
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - no->old_ticks;
bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- bytes = MIN(bytes, SIZE_MAX);
- samples = bytes >> hw->info.shift;
no->old_ticks = now;
- decr = MIN (live, samples);
- hw->rpos = (hw->rpos + decr) % hw->samples;
- return decr;
+ return MIN(len, bytes);
}
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
@@ -92,25 +87,21 @@ static void no_fini_in (HWVoiceIn *hw)
(void) hw;
}
-static size_t no_run_in(HWVoiceIn *hw)
+static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
{
+ size_t to_clear;
NoVoiceIn *no = (NoVoiceIn *) hw;
- size_t live = audio_pcm_hw_get_live_in(hw);
- size_t dead = hw->samples - live;
- size_t samples = 0;
-
- if (dead) {
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes =
- muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- no->old_ticks = now;
- bytes = MIN (bytes, SIZE_MAX);
- samples = bytes >> hw->info.shift;
- samples = MIN (samples, dead);
- }
- return samples;
+
+ int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ int64_t ticks = now - no->old_ticks;
+ int64_t bytes =
+ muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
+
+ no->old_ticks = now;
+ to_clear = MIN(bytes, size);
+
+ audio_pcm_info_clear_buf(&hw->info, buf, to_clear >> hw->info.shift);
+ return to_clear;
}
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -133,12 +124,12 @@ static void no_audio_fini (void *opaque)
static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
- .run_out = no_run_out,
+ .write = no_write,
.ctl_out = no_ctl_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
- .run_in = no_run_in,
+ .read = no_read,
.ctl_in = no_ctl_in
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 06/24] ossaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (4 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 05/24] noaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 07/24] paaudio: " Kővágó, Zoltán
` (14 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/ossaudio.c | 272 +++++++++++++++++------------------------------
1 file changed, 96 insertions(+), 176 deletions(-)
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 1696933688..2782512706 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -40,19 +40,15 @@
typedef struct OSSVoiceOut {
HWVoiceOut hw;
- void *pcm_buf;
int fd;
- int wpos;
int nfrags;
int fragsize;
int mmapped;
- int pending;
Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
HWVoiceIn hw;
- void *pcm_buf;
int fd;
int nfrags;
int fragsize;
@@ -371,98 +367,87 @@ static int oss_open(int in, struct oss_params *req, audsettings *as,
return -1;
}
-static void oss_write_pending (OSSVoiceOut *oss)
+static size_t oss_get_available_bytes(OSSVoiceOut *oss)
{
- HWVoiceOut *hw = &oss->hw;
+ int err;
+ struct count_info cntinfo;
+ assert(oss->mmapped);
- if (oss->mmapped) {
- return;
+ err = ioctl(oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
+ if (err < 0) {
+ oss_logerr(errno, "SNDCTL_DSP_GETOPTR failed\n");
+ return 0;
}
- while (oss->pending) {
- int samples_written;
- ssize_t bytes_written;
- int samples_till_end = hw->samples - oss->wpos;
- int samples_to_write = MIN (oss->pending, samples_till_end);
- int bytes_to_write = samples_to_write << hw->info.shift;
- void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift);
+ return audio_ring_dist(cntinfo.ptr, oss->hw.pos_emul, oss->hw.size_emul);
+}
- bytes_written = write (oss->fd, pcm, bytes_to_write);
- if (bytes_written < 0) {
- if (errno != EAGAIN) {
- oss_logerr (errno, "failed to write %d bytes\n",
- bytes_to_write);
- }
- break;
- }
+static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ if (oss->mmapped) {
+ *size = MIN(oss_get_available_bytes(oss), hw->size_emul - hw->pos_emul);
+ return hw->buf_emul + hw->pos_emul;
+ } else {
+ return audio_generic_get_buffer_out(hw, size);
+ }
+}
- if (bytes_written & hw->info.align) {
- dolog ("misaligned write asked for %d, but got %zd\n",
- bytes_to_write, bytes_written);
- return;
- }
+static size_t oss_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ if (oss->mmapped) {
+ assert(buf == hw->buf_emul + hw->pos_emul && size < hw->size_emul);
- samples_written = bytes_written >> hw->info.shift;
- oss->pending -= samples_written;
- oss->wpos = (oss->wpos + samples_written) % hw->samples;
- if (bytes_written - bytes_to_write) {
- break;
- }
+ hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+ return size;
+ } else {
+ return audio_generic_put_buffer_out(hw, buf, size);
}
}
-static size_t oss_run_out(HWVoiceOut *hw, size_t live)
+static size_t oss_write(HWVoiceOut *hw, void *buf, size_t len)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- int err;
- size_t decr;
- struct audio_buf_info abinfo;
- struct count_info cntinfo;
- size_t bufsize;
-
- bufsize = hw->samples << hw->info.shift;
+ size_t pos;
if (oss->mmapped) {
- int bytes, pos;
+ size_t total_len;
+ len = MIN(len, oss_get_available_bytes(oss));
- err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
- if (err < 0) {
- oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
- return 0;
- }
+ total_len = len;
+ while (len) {
+ size_t to_copy = MIN(len, hw->size_emul - hw->pos_emul);
+ memcpy(hw->buf_emul + hw->pos_emul, buf, to_copy);
- pos = hw->rpos << hw->info.shift;
- bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize);
- decr = MIN (bytes >> hw->info.shift, live);
- }
- else {
- err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
- if (err < 0) {
- oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
- return 0;
+ hw->pos_emul = (hw->pos_emul + to_copy) % hw->pos_emul;
+ buf += to_copy;
+ len -= to_copy;
}
+ return total_len;
+ }
- if (abinfo.bytes > bufsize) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- abinfo.bytes = bufsize;
- }
+ pos = 0;
+ while (len) {
+ ssize_t bytes_written;
+ void *pcm = advance(buf, pos);
- if (abinfo.bytes < 0) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- return 0;
+ bytes_written = write(oss->fd, pcm, len);
+ if (bytes_written < 0) {
+ if (errno != EAGAIN) {
+ oss_logerr(errno, "failed to write %zu bytes\n",
+ len);
+ }
+ return pos;
}
- decr = MIN (abinfo.bytes >> hw->info.shift, live);
- if (!decr) {
- return 0;
+ pos += bytes_written;
+ if (bytes_written < len) {
+ break;
}
+ len -= bytes_written;
}
-
- decr = audio_pcm_hw_clip_out (hw, oss->pcm_buf, decr, oss->pending);
- oss->pending += decr;
- oss_write_pending (oss);
-
- return decr;
+ return pos;
}
static void oss_fini_out (HWVoiceOut *hw)
@@ -473,18 +458,13 @@ static void oss_fini_out (HWVoiceOut *hw)
ldebug ("oss_fini\n");
oss_anal_close (&oss->fd);
- if (oss->pcm_buf) {
- if (oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
- if (err) {
- oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
- }
- }
- else {
- g_free (oss->pcm_buf);
+ if (oss->mmapped && hw->buf_emul) {
+ err = munmap(hw->buf_emul, hw->size_emul);
+ if (err) {
+ oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
+ hw->buf_emul, hw->size_emul);
}
- oss->pcm_buf = NULL;
+ hw->buf_emul = NULL;
}
}
@@ -535,19 +515,20 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- oss->pcm_buf = mmap (
+ hw->size_emul = hw->samples << hw->info.shift;
+ hw->buf_emul = mmap(
NULL,
- hw->samples << hw->info.shift,
+ hw->size_emul,
PROT_READ | PROT_WRITE,
MAP_SHARED,
fd,
0
);
- if (oss->pcm_buf == MAP_FAILED) {
+ if (hw->buf_emul == MAP_FAILED) {
oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
- hw->samples << hw->info.shift);
- }
- else {
+ hw->size_emul);
+ hw->buf_emul = NULL;
+ } else {
int err;
int trig = 0;
if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
@@ -567,30 +548,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
if (!oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
+ err = munmap(hw->buf_emul, hw->size_emul);
if (err) {
oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ hw->buf_emul, hw->size_emul);
}
+ hw->buf_emul = NULL;
}
}
}
- if (!oss->mmapped) {
- oss->pcm_buf = audio_calloc(__func__,
- hw->samples,
- 1 << hw->info.shift);
- if (!oss->pcm_buf) {
- dolog (
- "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
- hw->samples,
- 1 << hw->info.shift
- );
- oss_anal_close (&fd);
- return -1;
- }
- }
-
oss->fd = fd;
oss->dev = dev;
return 0;
@@ -618,7 +585,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- audio_pcm_info_clear_buf (&hw->info, oss->pcm_buf, hw->samples);
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
@@ -692,13 +659,6 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
- oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!oss->pcm_buf) {
- dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- oss_anal_close (&fd);
- return -1;
- }
oss->fd = fd;
oss->dev = dev;
@@ -710,78 +670,36 @@ static void oss_fini_in (HWVoiceIn *hw)
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
oss_anal_close (&oss->fd);
-
- g_free(oss->pcm_buf);
- oss->pcm_buf = NULL;
}
-static size_t oss_run_in(HWVoiceIn *hw)
+static size_t oss_read(HWVoiceIn *hw, void *buf, size_t len)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
- int hwshift = hw->info.shift;
- int i;
- size_t live = audio_pcm_hw_get_live_in (hw);
- size_t dead = hw->samples - live;
- size_t read_samples = 0;
- struct {
- size_t add;
- size_t len;
- } bufs[2] = {
- { .add = hw->wpos, .len = 0 },
- { .add = 0, .len = 0 }
- };
-
- if (!dead) {
- return 0;
- }
-
- if (hw->wpos + dead > hw->samples) {
- bufs[0].len = (hw->samples - hw->wpos) << hwshift;
- bufs[1].len = (dead - (hw->samples - hw->wpos)) << hwshift;
- }
- else {
- bufs[0].len = dead << hwshift;
- }
+ size_t pos = 0;
- for (i = 0; i < 2; ++i) {
+ while (len) {
ssize_t nread;
- if (bufs[i].len) {
- void *p = advance (oss->pcm_buf, bufs[i].add << hwshift);
- nread = read (oss->fd, p, bufs[i].len);
-
- if (nread > 0) {
- if (nread & hw->info.align) {
- dolog("warning: Misaligned read %zd (requested %zu), "
- "alignment %d\n", nread, bufs[i].add << hwshift,
- hw->info.align + 1);
- }
- read_samples += nread >> hwshift;
- hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift);
- }
+ void *dst = advance(buf, pos);
+ nread = read(oss->fd, dst, len);
- if (bufs[i].len - nread) {
- if (nread == -1) {
- switch (errno) {
- case EINTR:
- case EAGAIN:
- break;
- default:
- oss_logerr(
- errno,
- "Failed to read %zu bytes of audio (to %p)\n",
- bufs[i].len, p
- );
- break;
- }
- }
+ if (nread == -1) {
+ switch (errno) {
+ case EINTR:
+ case EAGAIN:
+ break;
+ default:
+ oss_logerr(errno, "Failed to read %zu bytes of audio (to %p)\n",
+ len, dst);
break;
}
}
+
+ pos += nread;
+ len -= nread;
}
- hw->wpos = (hw->wpos + read_samples) % hw->samples;
- return read_samples;
+ return pos;
}
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -845,12 +763,14 @@ static void oss_audio_fini (void *opaque)
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
- .run_out = oss_run_out,
+ .write = oss_write,
+ .get_buffer_out = oss_get_buffer_out,
+ .put_buffer_out = oss_put_buffer_out,
.ctl_out = oss_ctl_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
- .run_in = oss_run_in,
+ .read = oss_read,
.ctl_in = oss_ctl_in
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 07/24] paaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (5 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 06/24] ossaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 08/24] sdlaudio: " Kővágó, Zoltán
` (13 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 1 -
audio/audio_pt_int.c | 173 --------------------
audio/audio_pt_int.h | 22 ---
audio/paaudio.c | 372 ++++++-------------------------------------
configure | 5 -
5 files changed, 45 insertions(+), 528 deletions(-)
delete mode 100644 audio/audio_pt_int.c
delete mode 100644 audio/audio_pt_int.h
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index dca87f6347..d7490a379f 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -2,7 +2,6 @@ common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o
-common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
common-obj-y += wavcapture.o
diff --git a/audio/audio_pt_int.c b/audio/audio_pt_int.c
deleted file mode 100644
index 9f9d44ad4a..0000000000
--- a/audio/audio_pt_int.c
+++ /dev/null
@@ -1,173 +0,0 @@
-#include "qemu/osdep.h"
-#include "audio.h"
-
-#define AUDIO_CAP "audio-pt"
-
-#include "audio_int.h"
-#include "audio_pt_int.h"
-
-static void GCC_FMT_ATTR(3, 4) logerr (struct audio_pt *pt, int err,
- const char *fmt, ...)
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (pt->drv, fmt, ap);
- va_end (ap);
-
- AUD_log (NULL, "\n");
- AUD_log (pt->drv, "Reason: %s\n", strerror (err));
-}
-
-int audio_pt_init (struct audio_pt *p, void *(*func) (void *),
- void *opaque, const char *drv, const char *cap)
-{
- int err, err2;
- const char *efunc;
- sigset_t set, old_set;
-
- p->drv = drv;
-
- err = sigfillset (&set);
- if (err) {
- logerr(p, errno, "%s(%s): sigfillset failed", cap, __func__);
- return -1;
- }
-
- err = pthread_mutex_init (&p->mutex, NULL);
- if (err) {
- efunc = "pthread_mutex_init";
- goto err0;
- }
-
- err = pthread_cond_init (&p->cond, NULL);
- if (err) {
- efunc = "pthread_cond_init";
- goto err1;
- }
-
- err = pthread_sigmask (SIG_BLOCK, &set, &old_set);
- if (err) {
- efunc = "pthread_sigmask";
- goto err2;
- }
-
- err = pthread_create (&p->thread, NULL, func, opaque);
-
- err2 = pthread_sigmask (SIG_SETMASK, &old_set, NULL);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_sigmask (restore) failed",
- cap, __func__);
- /* We have failed to restore original signal mask, all bets are off,
- so terminate the process */
- exit (EXIT_FAILURE);
- }
-
- if (err) {
- efunc = "pthread_create";
- goto err2;
- }
-
- return 0;
-
- err2:
- err2 = pthread_cond_destroy (&p->cond);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_cond_destroy failed", cap, __func__);
- }
-
- err1:
- err2 = pthread_mutex_destroy (&p->mutex);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_mutex_destroy failed", cap, __func__);
- }
-
- err0:
- logerr(p, err, "%s(%s): %s failed", cap, __func__, efunc);
- return -1;
-}
-
-int audio_pt_fini (struct audio_pt *p, const char *cap)
-{
- int err, ret = 0;
-
- err = pthread_cond_destroy (&p->cond);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_destroy failed", cap, __func__);
- ret = -1;
- }
-
- err = pthread_mutex_destroy (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_destroy failed", cap, __func__);
- ret = -1;
- }
- return ret;
-}
-
-int audio_pt_lock (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_lock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_lock failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_unlock (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_unlock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_unlock failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_wait (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_cond_wait (&p->cond, &p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_wait failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_unlock_and_signal (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_unlock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_unlock failed", cap, __func__);
- return -1;
- }
- err = pthread_cond_signal (&p->cond);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_signal failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_join (struct audio_pt *p, void **arg, const char *cap)
-{
- int err;
- void *ret;
-
- err = pthread_join (p->thread, &ret);
- if (err) {
- logerr(p, err, "%s(%s): pthread_join failed", cap, __func__);
- return -1;
- }
- *arg = ret;
- return 0;
-}
diff --git a/audio/audio_pt_int.h b/audio/audio_pt_int.h
deleted file mode 100644
index 4c0c15b9af..0000000000
--- a/audio/audio_pt_int.h
+++ /dev/null
@@ -1,22 +0,0 @@
-#ifndef QEMU_AUDIO_PT_INT_H
-#define QEMU_AUDIO_PT_INT_H
-
-#include <pthread.h>
-
-struct audio_pt {
- const char *drv;
- pthread_t thread;
- pthread_cond_t cond;
- pthread_mutex_t mutex;
-};
-
-int audio_pt_init (struct audio_pt *, void *(*) (void *), void *,
- const char *, const char *);
-int audio_pt_fini (struct audio_pt *, const char *);
-int audio_pt_lock (struct audio_pt *, const char *);
-int audio_pt_unlock (struct audio_pt *, const char *);
-int audio_pt_wait (struct audio_pt *, const char *);
-int audio_pt_unlock_and_signal (struct audio_pt *, const char *);
-int audio_pt_join (struct audio_pt *, void **, const char *);
-
-#endif /* QEMU_AUDIO_PT_INT_H */
diff --git a/audio/paaudio.c b/audio/paaudio.c
index bfef9acaad..75fce53202 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -9,7 +9,6 @@
#define AUDIO_CAP "pulseaudio"
#include "audio_int.h"
-#include "audio_pt_int.h"
typedef struct PAConnection {
char *server;
@@ -30,28 +29,16 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
- size_t done;
- size_t live;
- size_t decr;
- size_t rpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
paaudio *g;
size_t samples;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
- size_t done;
- size_t dead;
- size_t incr;
- size_t wpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
const void *read_data;
- size_t read_index, read_length;
+ size_t read_length;
paaudio *g;
size_t samples;
} PAVoiceIn;
@@ -89,298 +76,96 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
}
#endif
-#define CHECK_SUCCESS_GOTO(c, rerror, expression, label) \
+#define CHECK_SUCCESS_GOTO(c, expression, label, msg) \
do { \
if (!(expression)) { \
- if (rerror) { \
- *(rerror) = pa_context_errno ((c)->context); \
- } \
+ qpa_logerr(pa_context_errno((c)->context), msg); \
goto label; \
} \
} while (0)
-#define CHECK_DEAD_GOTO(c, stream, rerror, label) \
+#define CHECK_DEAD_GOTO(c, stream, label, msg) \
do { \
if (!(c)->context || !PA_CONTEXT_IS_GOOD (pa_context_get_state((c)->context)) || \
!(stream) || !PA_STREAM_IS_GOOD (pa_stream_get_state ((stream)))) { \
if (((c)->context && pa_context_get_state ((c)->context) == PA_CONTEXT_FAILED) || \
((stream) && pa_stream_get_state ((stream)) == PA_STREAM_FAILED)) { \
- if (rerror) { \
- *(rerror) = pa_context_errno ((c)->context); \
- } \
+ qpa_logerr(pa_context_errno((c)->context), msg); \
} else { \
- if (rerror) { \
- *(rerror) = PA_ERR_BADSTATE; \
- } \
+ qpa_logerr(PA_ERR_BADSTATE, msg); \
} \
goto label; \
} \
} while (0)
-static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
+static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length)
{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
PAConnection *c = p->g->conn;
+ size_t l;
+ int r;
pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO(c, 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(c, rerror, r == 0, unlock_and_fail);
-
- if (!p->read_data) {
- pa_threaded_mainloop_wait(c->mainloop);
- CHECK_DEAD_GOTO(c, 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;
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
- p->read_index += l;
- p->read_length -= l;
+ if (!p->read_length) {
+ r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail,
+ "pa_stream_peek failed\n");
+ }
- if (!p->read_length) {
- int r;
+ l = MIN(p->read_length, length);
+ memcpy(data, p->read_data, l);
- r = pa_stream_drop (p->stream);
- p->read_data = NULL;
- p->read_length = 0;
- p->read_index = 0;
+ p->read_data += l;
+ p->read_length -= l;
- CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
- }
+ if (!p->read_length) {
+ r = pa_stream_drop(p->stream);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail,
+ "pa_stream_drop failed\n");
}
pa_threaded_mainloop_unlock(c->mainloop);
- return 0;
+ return l;
unlock_and_fail:
pa_threaded_mainloop_unlock(c->mainloop);
- return -1;
+ return 0;
}
-static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
+static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
{
+ PAVoiceOut *p = (PAVoiceOut *) hw;
PAConnection *c = p->g->conn;
+ size_t l;
+ int r;
pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
- while (length > 0) {
- size_t l;
- int r;
+ l = pa_stream_writable_size(p->stream);
- while (!(l = pa_stream_writable_size (p->stream))) {
- pa_threaded_mainloop_wait(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
- }
-
- CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
-
- if (l > length) {
- l = length;
- }
+ CHECK_SUCCESS_GOTO(c, l != (size_t) -1, unlock_and_fail,
+ "pa_stream_writable_size failed\n");
- r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
-
- data = (const uint8_t *) data + l;
- length -= l;
+ if (l > length) {
+ l = length;
}
+ r = pa_stream_write(p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
+ CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail, "pa_stream_write failed\n");
+
pa_threaded_mainloop_unlock(c->mainloop);
- return 0;
+ return l;
unlock_and_fail:
pa_threaded_mainloop_unlock(c->mainloop);
- return -1;
-}
-
-static void *qpa_thread_out (void *arg)
-{
- PAVoiceOut *pa = arg;
- HWVoiceOut *hw = &pa->hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- for (;;) {
- size_t decr, to_mix, rpos;
-
- for (;;) {
- if (pa->done) {
- goto exit;
- }
-
- if (pa->live > 0) {
- break;
- }
-
- if (audio_pt_wait(&pa->pt, __func__)) {
- goto exit;
- }
- }
-
- decr = to_mix = MIN(pa->live, pa->samples >> 5);
- rpos = pa->rpos;
-
- if (audio_pt_unlock(&pa->pt, __func__)) {
- return NULL;
- }
-
- while (to_mix) {
- int error;
- size_t chunk = MIN (to_mix, hw->samples - rpos);
- struct st_sample *src = hw->mix_buf + rpos;
-
- 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;
- }
-
- rpos = (rpos + chunk) % hw->samples;
- to_mix -= chunk;
- }
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- pa->rpos = rpos;
- pa->live -= decr;
- pa->decr += decr;
- }
-
- exit:
- audio_pt_unlock(&pa->pt, __func__);
- return NULL;
-}
-
-static size_t qpa_run_out(HWVoiceOut *hw, size_t live)
-{
- size_t decr;
- PAVoiceOut *pa = (PAVoiceOut *) hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return 0;
- }
-
- decr = 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, __func__);
- }
- else {
- audio_pt_unlock(&pa->pt, __func__);
- }
- return decr;
-}
-
-/* capture */
-static void *qpa_thread_in (void *arg)
-{
- PAVoiceIn *pa = arg;
- HWVoiceIn *hw = &pa->hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- for (;;) {
- size_t incr, to_grab, wpos;
-
- for (;;) {
- if (pa->done) {
- goto exit;
- }
-
- if (pa->dead > 0) {
- break;
- }
-
- if (audio_pt_wait(&pa->pt, __func__)) {
- goto exit;
- }
- }
-
- incr = to_grab = MIN(pa->dead, pa->samples >> 5);
- wpos = pa->wpos;
-
- if (audio_pt_unlock(&pa->pt, __func__)) {
- return NULL;
- }
-
- while (to_grab) {
- int error;
- size_t chunk = MIN (to_grab, hw->samples - wpos);
- void *buf = advance (pa->pcm_buf, wpos);
-
- 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);
- wpos = (wpos + chunk) % hw->samples;
- to_grab -= chunk;
- }
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- pa->wpos = wpos;
- pa->dead -= incr;
- pa->incr += incr;
- }
-
- exit:
- audio_pt_unlock(&pa->pt, __func__);
- return NULL;
-}
-
-static size_t qpa_run_in(HWVoiceIn *hw)
-{
- size_t live, incr, dead;
- PAVoiceIn *pa = (PAVoiceIn *) hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return 0;
- }
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- incr = 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, __func__);
- }
- else {
- audio_pt_unlock(&pa->pt, __func__);
- }
- return incr;
+ return 0;
}
static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
@@ -468,13 +253,6 @@ static void stream_state_cb (pa_stream *s, void * userdata)
}
}
-static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
-{
- PAConnection *c = userdata;
-
- pa_threaded_mainloop_signal(c->mainloop, 0);
-}
-
static pa_stream *qpa_simple_new (
PAConnection *c,
const char *name,
@@ -497,8 +275,6 @@ static pa_stream *qpa_simple_new (
}
pa_stream_set_state_callback(stream, stream_state_cb, c);
- pa_stream_set_read_callback(stream, stream_request_cb, c);
- pa_stream_set_write_callback(stream, stream_request_cb, c);
flags =
PA_STREAM_INTERPOLATE_TIMING
@@ -579,28 +355,9 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo),
&obt_as, ppdo->buffer_length);
- pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- pa->rpos = hw->rpos;
- if (!pa->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init(&pa->pt, qpa_thread_out, hw, AUDIO_CAP, __func__)) {
- goto fail3;
- }
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:
return -1;
}
@@ -647,28 +404,9 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo),
&obt_as, ppdo->buffer_length);
- pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- pa->wpos = hw->wpos;
- if (!pa->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init(&pa->pt, qpa_thread_in, hw, AUDIO_CAP, __func__)) {
- goto fail3;
- }
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:
return -1;
}
@@ -696,42 +434,22 @@ static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
static void qpa_fini_out (HWVoiceOut *hw)
{
- void *ret;
PAVoiceOut *pa = (PAVoiceOut *) hw;
- audio_pt_lock(&pa->pt, __func__);
- pa->done = 1;
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- audio_pt_join(&pa->pt, &ret, __func__);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini(&pa->pt, __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, __func__);
- pa->done = 1;
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- audio_pt_join(&pa->pt, &ret, __func__);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini(&pa->pt, __func__);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -1005,12 +723,12 @@ static void qpa_audio_fini (void *opaque)
static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
- .run_out = qpa_run_out,
+ .write = qpa_write,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
- .run_in = qpa_run_in,
+ .read = qpa_read,
.ctl_in = qpa_ctl_in
};
diff --git a/configure b/configure
index 30aad233d1..d72ea39bb7 100755
--- a/configure
+++ b/configure
@@ -297,7 +297,6 @@ host_cc="cc"
libs_cpu=""
libs_softmmu=""
libs_tools=""
-audio_pt_int=""
audio_win_int=""
libs_qga=""
debug_info="yes"
@@ -3383,7 +3382,6 @@ for drv in $audio_drv_list; do
pa | try-pa)
if $pkg_config libpulse --exists; then
pulse_libs=$($pkg_config libpulse --libs)
- audio_pt_int="yes"
if test "$drv" = "try-pa"; then
audio_drv_list=$(echo "$audio_drv_list" | sed -e 's/try-pa/pa/')
fi
@@ -6606,9 +6604,6 @@ echo "PULSE_LIBS=$pulse_libs" >> $config_host_mak
echo "COREAUDIO_LIBS=$coreaudio_libs" >> $config_host_mak
echo "DSOUND_LIBS=$dsound_libs" >> $config_host_mak
echo "OSS_LIBS=$oss_libs" >> $config_host_mak
-if test "$audio_pt_int" = "yes" ; then
- echo "CONFIG_AUDIO_PT_INT=y" >> $config_host_mak
-fi
if test "$audio_win_int" = "yes" ; then
echo "CONFIG_AUDIO_WIN_INT=y" >> $config_host_mak
fi
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 08/24] sdlaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (6 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 07/24] paaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 10/24] wavaudio: " Kővágó, Zoltán
` (12 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/sdlaudio.c | 87 +++++++++++++++++++++++-------------------------
1 file changed, 42 insertions(+), 45 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 14b11f0335..f7ac8cd101 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -41,8 +41,6 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
- size_t live;
- size_t decr;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -184,62 +182,59 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
- size_t samples = len >> hw->info.shift;
- size_t to_mix, decr;
- if (s->exit || !sdl->live) {
+ if (s->exit) {
return;
}
/* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
- to_mix = MIN(samples, sdl->live);
- decr = to_mix;
- while (to_mix) {
- size_t chunk = MIN(to_mix, hw->samples - hw->rpos);
- struct st_sample *src = hw->mix_buf + hw->rpos;
-
- /* dolog ("in callback to_mix %zu, chunk %zu\n", to_mix, chunk); */
- hw->clip(buf, src, chunk);
- hw->rpos = (hw->rpos + chunk) % hw->samples;
- to_mix -= chunk;
- buf += chunk << hw->info.shift;
+ while (hw->pending_emul && len) {
+ size_t write_len;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
+
+ write_len = MIN(MIN(hw->pending_emul, len),
+ hw->size_emul - start);
+
+ memcpy(buf, hw->buf_emul + start, write_len);
+ hw->pending_emul -= write_len;
+ len -= write_len;
+ buf += write_len;
}
- samples -= decr;
- sdl->live -= decr;
- sdl->decr += decr;
- /* dolog ("done len=%zu\n", len); */
-
- /* SDL2 does not clear the remaining buffer for us, so do it on our own */
- if (samples) {
- memset(buf, 0, samples << hw->info.shift);
+ /* clear remaining buffer that we couldn't fill with data */
+ if (len) {
+ memset(buf, 0, len);
}
}
-static size_t sdl_run_out(HWVoiceOut *hw, size_t live)
-{
- size_t decr;
- SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
-
- SDL_LockAudio();
-
- if (sdl->decr > live) {
- ldebug ("sdl->decr %d live %d sdl->live %d\n",
- sdl->decr,
- live,
- sdl->live);
+#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \
+ static ret_type glue(sdl_, name)args_decl \
+ { \
+ ret_type ret; \
+ \
+ SDL_LockAudio(); \
+ \
+ ret = glue(audio_generic_, name)args; \
+ \
+ SDL_UnlockAudio(); \
+ return ret; \
}
- decr = MIN (sdl->decr, live);
- sdl->decr -= decr;
-
- sdl->live = live;
+SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
+ (hw, size), *size = 0, sdl_unlock)
+SDL_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
+ /*nothing*/, sdl_unlock_and_post)
+SDL_WRAPPER_FUNC(write, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
+ /*nothing*/, sdl_unlock_and_post)
- SDL_UnlockAudio();
-
- return decr;
-}
+#undef SDL_WRAPPER_FUNC
static void sdl_fini_out (HWVoiceOut *hw)
{
@@ -336,7 +331,9 @@ static void sdl_audio_fini (void *opaque)
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
- .run_out = sdl_run_out,
+ .write = sdl_write,
+ .get_buffer_out = sdl_get_buffer_out,
+ .put_buffer_out = sdl_put_buffer_out_nowrite,
.ctl_out = sdl_ctl_out,
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 10/24] wavaudio: port to the new audio backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (7 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 08/24] sdlaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 11/24] audio: remove remains of the old " Kővágó, Zoltán
` (11 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
Changes from v3:
* fix compilation on 32-bit targets
audio/wavaudio.c | 54 ++++++++----------------------------------------
1 file changed, 9 insertions(+), 45 deletions(-)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index b6eeeb4e26..78af2f1338 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -36,52 +36,28 @@ typedef struct WAVVoiceOut {
HWVoiceOut hw;
FILE *f;
int64_t old_ticks;
- void *pcm_buf;
int total_samples;
} WAVVoiceOut;
-static size_t wav_run_out(HWVoiceOut *hw, size_t live)
+static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- size_t rpos, decr, samples;
- uint8_t *dst;
- struct st_sample *src;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
int64_t ticks = now - wav->old_ticks;
int64_t bytes =
muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- if (bytes > INT_MAX) {
- samples = INT_MAX >> hw->info.shift;
- }
- else {
- samples = bytes >> hw->info.shift;
- }
-
+ bytes = MIN(bytes, len);
+ bytes = bytes >> hw->info.shift << hw->info.shift;
wav->old_ticks = now;
- decr = MIN (live, samples);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int convert_samples = MIN (samples, left_till_end_samples);
-
- src = hw->mix_buf + rpos;
- dst = advance (wav->pcm_buf, rpos << hw->info.shift);
-
- hw->clip (dst, src, convert_samples);
- if (fwrite (dst, convert_samples << hw->info.shift, 1, wav->f) != 1) {
- dolog ("wav_run_out: fwrite of %d bytes failed\nReaons: %s\n",
- convert_samples << hw->info.shift, strerror (errno));
- }
- rpos = (rpos + convert_samples) % hw->samples;
- samples -= convert_samples;
- wav->total_samples += convert_samples;
+ if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
+ dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
+ bytes, strerror(errno));
}
- hw->rpos = rpos;
- return decr;
+ wav->total_samples += bytes >> hw->info.shift;
+ return bytes;
}
/* VICE code: Store number as little endian. */
@@ -137,13 +113,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
- wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!wav->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- return -1;
- }
-
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
@@ -153,8 +122,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
wav_path, strerror(errno));
- g_free (wav->pcm_buf);
- wav->pcm_buf = NULL;
return -1;
}
@@ -208,9 +175,6 @@ static void wav_fini_out (HWVoiceOut *hw)
wav->f, strerror (errno));
}
wav->f = NULL;
-
- g_free (wav->pcm_buf);
- wav->pcm_buf = NULL;
}
static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -234,7 +198,7 @@ static void wav_audio_fini (void *opaque)
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
- .run_out = wav_run_out,
+ .write = wav_write_out,
.ctl_out = wav_ctl_out,
};
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 11/24] audio: remove remains of the old backend api
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (8 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 10/24] wavaudio: " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 12/24] audio: unify input and output mixeng buffer management Kővágó, Zoltán
` (10 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 42 ++++++------------------------------------
audio/audio_int.h | 7 -------
2 files changed, 6 insertions(+), 43 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e29a1e15eb..435bcf20c1 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -541,7 +541,7 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
return m;
}
-size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
+static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug(__func__, live > hw->samples)) {
@@ -551,29 +551,7 @@ size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
return live;
}
-size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
- size_t live, size_t pending)
-{
- size_t left = hw->samples - pending;
- size_t len = MIN (left, live);
- size_t clipped = 0;
-
- while (len) {
- struct st_sample *src = hw->mix_buf + hw->rpos;
- uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
- size_t samples_till_end_of_buf = hw->samples - hw->rpos;
- size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
-
- hw->clip (dst, src, samples_to_clip);
-
- hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
- len -= samples_to_clip;
- clipped += samples_to_clip;
- }
- return clipped;
-}
-
-static void audio_pcm_hw_clip_out2(HWVoiceOut *hw, void *pcm_buf, size_t len)
+static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
size_t pos = hw->rpos;
@@ -1083,7 +1061,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
}
decr = MIN(size >> hw->info.shift, live);
- audio_pcm_hw_clip_out2(hw, buf, decr);
+ audio_pcm_hw_clip_out(hw, buf, decr);
proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
hw->info.shift;
@@ -1146,11 +1124,7 @@ static void audio_run_out (AudioState *s)
}
prev_rpos = hw->rpos;
- if (hw->pcm_ops->run_out) {
- played = hw->pcm_ops->run_out(hw, live);
- } else {
- played = audio_pcm_hw_run_out(hw, live);
- }
+ played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
@@ -1247,12 +1221,8 @@ static void audio_run_in (AudioState *s)
size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
- if (hw->pcm_ops->run_in) {
- captured = hw->pcm_ops->run_in(hw);
- } else {
- captured = audio_pcm_hw_run_in(
- hw, hw->samples - audio_pcm_hw_get_live_in(hw));
- }
+ captured = audio_pcm_hw_run_in(
+ hw, hw->samples - audio_pcm_hw_get_live_in(hw));
}
replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 8fb1ca8a8d..c76d7c39e8 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -153,7 +153,6 @@ struct audio_driver {
struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
- size_t (*run_out)(HWVoiceOut *hw, size_t live);
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
/*
* get a buffer that after later can be passed to put_buffer_out; optional
@@ -171,7 +170,6 @@ struct audio_pcm_ops {
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
- size_t (*run_in)(HWVoiceIn *hw);
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
@@ -237,11 +235,6 @@ audio_driver *audio_driver_lookup(const char *name);
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);
-size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
-
-size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
- size_t live, size_t pending);
-
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 12/24] audio: unify input and output mixeng buffer management
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (9 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 11/24] audio: remove remains of the old " Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 13/24] audio: common rate control code for timer based outputs Kővágó, Zoltán
` (9 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before
initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the
initial size of the STSampleBuffer. The next commit tries to fix this
inconsistency.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 122 +++++++++++++++++++++--------------------
audio/audio_int.h | 12 ++--
audio/audio_template.h | 19 +++----
audio/ossaudio.c | 3 +-
4 files changed, 80 insertions(+), 76 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 435bcf20c1..ba07fb77dd 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -544,8 +544,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
return live;
@@ -554,17 +554,17 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
- size_t pos = hw->rpos;
+ size_t pos = hw->mix_buf->pos;
while (len) {
- st_sample *src = hw->mix_buf + pos;
+ st_sample *src = hw->mix_buf->samples + pos;
uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
- size_t samples_till_end_of_buf = hw->samples - pos;
+ size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
hw->clip(dst, src, samples_to_clip);
- pos = (pos + samples_to_clip) % hw->samples;
+ pos = (pos + samples_to_clip) % hw->mix_buf->size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
@@ -579,17 +579,17 @@ static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
ssize_t rpos;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
- rpos = hw->wpos - live;
+ rpos = hw->conv_buf->pos - live;
if (rpos >= 0) {
return rpos;
}
else {
- return hw->samples + rpos;
+ return hw->conv_buf->size + rpos;
}
}
@@ -599,11 +599,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
- rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
+ rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
@@ -616,11 +616,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
swlim = MIN (swlim, samples);
while (swlim) {
- src = hw->conv_buf + rpos;
- if (hw->wpos > rpos) {
- isamp = hw->wpos - rpos;
+ src = hw->conv_buf->samples + rpos;
+ if (hw->conv_buf->pos > rpos) {
+ isamp = hw->conv_buf->pos - rpos;
} else {
- isamp = hw->samples - rpos;
+ isamp = hw->conv_buf->size - rpos;
}
if (!isamp) {
@@ -630,7 +630,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
- rpos = (rpos + isamp) % hw->samples;
+ rpos = (rpos + isamp) % hw->conv_buf->size;
dst += osamp;
ret += osamp;
total += isamp;
@@ -678,8 +678,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
if (nb_live1) {
size_t live = smin;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
return 0;
}
return live;
@@ -699,11 +699,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return size;
}
- hwsamples = sw->hw->samples;
+ hwsamples = sw->hw->mix_buf->size;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
return 0;
}
@@ -714,7 +714,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return 0;
}
- wpos = (sw->hw->rpos + live) % hwsamples;
+ wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
samples = size >> sw->info.shift;
dead = hwsamples - live;
@@ -740,7 +740,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
- sw->hw->mix_buf + wpos,
+ sw->hw->mix_buf->samples + wpos,
&isamp,
&osamp
);
@@ -868,7 +868,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->samples << sw->hw->info.shift;
+ return sw->hw->mix_buf->size << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -969,8 +969,9 @@ static size_t audio_get_avail (SWVoiceIn *sw)
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
+ dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
+ sw->hw->conv_buf->size);
return 0;
}
@@ -993,12 +994,13 @@ static size_t audio_get_free(SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
+ dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
+ sw->hw->mix_buf->size);
return 0;
}
- dead = sw->hw->samples - live;
+ dead = sw->hw->mix_buf->size - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
@@ -1023,12 +1025,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
n = samples;
while (n) {
- size_t till_end_of_hw = hw->samples - rpos2;
+ size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
size_t bytes = to_write << hw->info.shift;
size_t written;
- sw->buf = hw->mix_buf + rpos2;
+ sw->buf = hw->mix_buf->samples + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog("Could not mix %zu bytes into a capture "
@@ -1037,14 +1039,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
break;
}
n -= to_write;
- rpos2 = (rpos2 + to_write) % hw->samples;
+ rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
}
}
}
- n = MIN(samples, hw->samples - rpos);
- mixeng_clear(hw->mix_buf + rpos, n);
- mixeng_clear(hw->mix_buf, samples - n);
+ n = MIN(samples, hw->mix_buf->size - rpos);
+ mixeng_clear(hw->mix_buf->samples + rpos, n);
+ mixeng_clear(hw->mix_buf->samples, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@@ -1056,7 +1058,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
if (!buf) {
/* retrying will likely won't help, drop everything. */
- hw->rpos = (hw->rpos + live) % hw->samples;
+ hw->mix_buf->pos = (hw->mix_buf->pos + live) % hw->mix_buf->size;
return clipped + live;
}
@@ -1067,7 +1069,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
live -= proc;
clipped += proc;
- hw->rpos = (hw->rpos + proc) % hw->samples;
+ hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
if (proc == 0 || proc < decr) {
break;
@@ -1091,8 +1093,8 @@ static void audio_run_out (AudioState *s)
live = 0;
}
- if (audio_bug(__func__, live > hw->samples)) {
- dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
continue;
}
@@ -1123,13 +1125,13 @@ static void audio_run_out (AudioState *s)
continue;
}
- prev_rpos = hw->rpos;
+ prev_rpos = hw->mix_buf->pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
- if (audio_bug(__func__, hw->rpos >= hw->samples)) {
- dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
- hw->rpos, hw->samples, played);
- hw->rpos = 0;
+ if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
+ dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
+ hw->mix_buf->pos, hw->mix_buf->size, played);
+ hw->mix_buf->pos = 0;
}
#ifdef DEBUG_OUT
@@ -1186,6 +1188,7 @@ static void audio_run_out (AudioState *s)
static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
{
size_t conv = 0;
+ STSampleBuffer *conv_buf = hw->conv_buf;
while (samples) {
size_t proc;
@@ -1199,10 +1202,10 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
}
proc = MIN(size >> hw->info.shift,
- hw->samples - hw->wpos);
+ conv_buf->size - conv_buf->pos);
- hw->conv(hw->conv_buf + hw->wpos, buf, proc);
- hw->wpos = (hw->wpos + proc) % hw->samples;
+ hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
+ conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
@@ -1222,9 +1225,10 @@ static void audio_run_in (AudioState *s)
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
- hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+ hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
}
- replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
+ replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
+ hw->conv_buf->size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1255,14 +1259,14 @@ static void audio_run_capture (AudioState *s)
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
- rpos = hw->rpos;
+ rpos = hw->mix_buf->pos;
while (live) {
- size_t left = hw->samples - rpos;
+ size_t left = hw->mix_buf->size - rpos;
size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
- src = hw->mix_buf + rpos;
+ src = hw->mix_buf->samples + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
@@ -1270,10 +1274,10 @@ static void audio_run_capture (AudioState *s)
cb->ops.capture (cb->opaque, cap->buf,
to_capture << hw->info.shift);
}
- rpos = (rpos + to_capture) % hw->samples;
+ rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
}
- hw->rpos = rpos;
+ hw->mix_buf->pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
@@ -1321,7 +1325,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1357,7 +1361,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1764,11 +1768,11 @@ CaptureVoiceOut *AUD_add_capture(
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
- hw->mix_buf = g_new0(struct st_sample, hw->samples);
+ audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
- cap->buf = g_malloc0_n(hw->samples, 1 << hw->info.shift);
+ cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
diff --git a/audio/audio_int.h b/audio/audio_int.h
index c76d7c39e8..20021df9e8 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -52,6 +52,11 @@ struct audio_pcm_info {
typedef struct AudioState AudioState;
typedef struct SWVoiceCap SWVoiceCap;
+typedef struct STSampleBuffer {
+ size_t pos, size;
+ st_sample samples[];
+} STSampleBuffer;
+
typedef struct HWVoiceOut {
AudioState *s;
int enabled;
@@ -60,11 +65,9 @@ typedef struct HWVoiceOut {
struct audio_pcm_info info;
f_sample *clip;
-
- size_t rpos;
uint64_t ts_helper;
- struct st_sample *mix_buf;
+ STSampleBuffer *mix_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@@ -84,11 +87,10 @@ typedef struct HWVoiceIn {
t_sample *conv;
- size_t wpos;
size_t total_samples_captured;
uint64_t ts_helper;
- struct st_sample *conv_buf;
+ STSampleBuffer *conv_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index ff4a173f18..87c6d2d271 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -76,16 +76,15 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
HWBUF = NULL;
}
-static bool glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
+static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
- HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
- if (!HWBUF) {
- dolog("Could not allocate " NAME " buffer (%zu samples)\n",
- hw->samples);
- return false;
+ size_t samples = hw->samples;
+ if (audio_bug(__func__, samples == 0)) {
+ dolog("Attempted to allocate empty buffer\n");
}
- return true;
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+ HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -104,7 +103,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
- samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+ samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
if (!sw->buf) {
@@ -280,9 +279,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
- if (!glue(audio_pcm_hw_alloc_resources_, TYPE)(hw)) {
- goto err1;
- }
+ glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
QLIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
glue (s->nb_hw_voices_, TYPE) -= 1;
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 2782512706..76c082d5e2 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -585,7 +585,8 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
+ audio_pcm_info_clear_buf(
+ &hw->info, hw->buf_emul, hw->mix_buf->size);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 13/24] audio: common rate control code for timer based outputs
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (10 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 12/24] audio: unify input and output mixeng buffer management Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 14/24] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
` (8 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This commit removes the ad-hoc rate-limiting code from noaudio and
wavaudio, and replaces them with a (slightly modified) code from
spiceaudio. This way multiple write calls (for example when the
circular buffer wraps around) do not cause problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 30 +++++++++++++++++++++++++++
audio/audio_int.h | 9 ++++++++
audio/noaudio.c | 49 ++++++++++++++++++++------------------------
audio/spiceaudio.c | 51 ++++++++--------------------------------------
audio/wavaudio.c | 21 +++++++++----------
5 files changed, 79 insertions(+), 81 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index ba07fb77dd..fab1e35718 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2051,3 +2051,33 @@ const char *audio_get_id(QEMUSoundCard *card)
return "";
}
}
+
+void audio_rate_start(RateCtl *rate)
+{
+ memset(rate, 0, sizeof(RateCtl));
+ rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+}
+
+size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+ size_t bytes_avail)
+{
+ int64_t now;
+ int64_t ticks;
+ int64_t bytes;
+ int64_t samples;
+ size_t ret;
+
+ now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ ticks = now - rate->start_ticks;
+ bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
+ samples = (bytes - rate->bytes_sent) >> info->shift;
+ if (samples < 0 || samples > 65536) {
+ AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
+ audio_rate_start(rate);
+ samples = 0;
+ }
+
+ ret = MIN(samples << info->shift, bytes_avail);
+ rate->bytes_sent += ret;
+ return ret;
+}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 20021df9e8..778615ccaf 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -242,6 +242,15 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size);
void audio_run(AudioState *s, const char *msg);
+typedef struct RateCtl {
+ int64_t start_ticks;
+ int64_t bytes_sent;
+} RateCtl;
+
+void audio_rate_start(RateCtl *rate);
+size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+ size_t bytes_avail);
+
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
#define VOICE_VOLUME 3
diff --git a/audio/noaudio.c b/audio/noaudio.c
index b054fd225b..9f1cc67df9 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -33,33 +33,27 @@
typedef struct NoVoiceOut {
HWVoiceOut hw;
- int64_t old_ticks;
+ RateCtl rate;
} NoVoiceOut;
typedef struct NoVoiceIn {
HWVoiceIn hw;
- int64_t old_ticks;
+ RateCtl rate;
} NoVoiceIn;
static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- int64_t now;
- int64_t ticks;
- int64_t bytes;
-
- now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- ticks = now - no->old_ticks;
- bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- no->old_ticks = now;
- return MIN(len, bytes);
+ return audio_rate_get_bytes(&hw->info, &no->rate, len);
}
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
+ NoVoiceOut *no = (NoVoiceOut *) hw;
+
audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
+ audio_rate_start(&no->rate);
return 0;
}
@@ -70,15 +64,21 @@ static void no_fini_out (HWVoiceOut *hw)
static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ NoVoiceOut *no = (NoVoiceOut *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&no->rate);
+ }
return 0;
}
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
+ NoVoiceIn *no = (NoVoiceIn *) hw;
+
audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
+ audio_rate_start(&no->rate);
return 0;
}
@@ -89,25 +89,20 @@ static void no_fini_in (HWVoiceIn *hw)
static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
{
- size_t to_clear;
NoVoiceIn *no = (NoVoiceIn *) hw;
+ int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes =
- muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- no->old_ticks = now;
- to_clear = MIN(bytes, size);
-
- audio_pcm_info_clear_buf(&hw->info, buf, to_clear >> hw->info.shift);
- return to_clear;
+ audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+ return bytes;
}
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ NoVoiceIn *no = (NoVoiceIn *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&no->rate);
+ }
return 0;
}
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index ff4e4dcbb0..4ce4f94c6d 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -40,15 +40,10 @@
#define LINE_IN_SAMPLES (256 * 4)
#endif
-typedef struct SpiceRateCtl {
- int64_t start_ticks;
- int64_t bytes_sent;
-} SpiceRateCtl;
-
typedef struct SpiceVoiceOut {
HWVoiceOut hw;
SpicePlaybackInstance sin;
- SpiceRateCtl rate;
+ RateCtl rate;
int active;
uint32_t *frame;
uint32_t fpos;
@@ -58,7 +53,7 @@ typedef struct SpiceVoiceOut {
typedef struct SpiceVoiceIn {
HWVoiceIn hw;
SpiceRecordInstance sin;
- SpiceRateCtl rate;
+ RateCtl rate;
int active;
} SpiceVoiceIn;
@@ -89,32 +84,6 @@ static void spice_audio_fini (void *opaque)
/* nothing */
}
-static void rate_start (SpiceRateCtl *rate)
-{
- memset (rate, 0, sizeof (*rate));
- rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
-}
-
-static int rate_get_samples (struct audio_pcm_info *info, SpiceRateCtl *rate)
-{
- int64_t now;
- int64_t ticks;
- int64_t bytes;
- int64_t samples;
-
- now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- ticks = now - rate->start_ticks;
- bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
- samples = (bytes - rate->bytes_sent) >> info->shift;
- if (samples < 0 || samples > 65536) {
- error_report("Resetting rate control (%" PRId64 " samples)", samples);
- rate_start(rate);
- samples = 0;
- }
- rate->bytes_sent += samples << info->shift;
- return samples;
-}
-
/* playback */
static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
@@ -154,7 +123,6 @@ static void line_out_fini (HWVoiceOut *hw)
static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
- size_t decr;
if (!out->frame) {
spice_server_playback_get_buffer(&out->sin, &out->frame, &out->fsize);
@@ -162,12 +130,10 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
}
if (out->frame) {
- decr = rate_get_samples(&hw->info, &out->rate);
- decr = MIN(out->fsize - out->fpos, decr);
-
- *size = decr << hw->info.shift;
+ *size = audio_rate_get_bytes(
+ &hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift);
} else {
- rate_start(&out->rate);
+ audio_rate_start(&out->rate);
}
return out->frame + out->fpos;
}
@@ -197,7 +163,7 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
break;
}
out->active = 1;
- rate_start (&out->rate);
+ audio_rate_start(&out->rate);
spice_server_playback_start (&out->sin);
break;
case VOICE_DISABLE:
@@ -273,8 +239,7 @@ static void line_in_fini (HWVoiceIn *hw)
static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- uint64_t delta_samp = rate_get_samples(&hw->info, &in->rate);
- uint64_t to_read = MIN(len >> 2, delta_samp);
+ uint64_t to_read = audio_rate_get_bytes(&hw->info, &in->rate, len) >> 2;
size_t ready = spice_server_record_get_samples(&in->sin, buf, to_read);
/* XXX: do we need this? */
@@ -296,7 +261,7 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
break;
}
in->active = 1;
- rate_start (&in->rate);
+ audio_rate_start(&in->rate);
spice_server_record_start (&in->sin);
break;
case VOICE_DISABLE:
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 78af2f1338..85f5ff9c28 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -35,21 +35,15 @@
typedef struct WAVVoiceOut {
HWVoiceOut hw;
FILE *f;
- int64_t old_ticks;
+ RateCtl rate;
int total_samples;
} WAVVoiceOut;
static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - wav->old_ticks;
- int64_t bytes =
- muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- bytes = MIN(bytes, len);
- bytes = bytes >> hw->info.shift << hw->info.shift;
- wav->old_ticks = now;
+ int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
+ assert(bytes >> hw->info.shift << hw->info.shift == bytes);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
@@ -130,6 +124,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
strerror(errno));
return -1;
}
+
+ audio_rate_start(&wav->rate);
return 0;
}
@@ -179,8 +175,11 @@ static void wav_fini_out (HWVoiceOut *hw)
static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&wav->rate);
+ }
return 0;
}
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 14/24] audio: split ctl_* functions into enable_* and volume_*
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (11 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 13/24] audio: common rate control code for timer based outputs Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 15/24] audio: add mixing-engine option (documentation) Kővágó, Zoltán
` (7 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This way we no longer need vararg functions, improving compile time
error detection. Also now it's possible to check actually what commands
are supported, without needing to manually update ctl_caps.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 62 ++++++++----------
audio/audio.c | 45 ++++++++-----
audio/audio_int.h | 15 ++---
audio/audio_template.h | 1 -
audio/coreaudio.c | 13 ++--
audio/dsoundaudio.c | 50 ++++++---------
audio/noaudio.c | 14 ++---
audio/ossaudio.c | 79 +++++++++--------------
audio/paaudio.c | 139 +++++++++++++++++------------------------
audio/sdlaudio.c | 17 +----
audio/spiceaudio.c | 102 +++++++++++++-----------------
audio/wavaudio.c | 7 +--
12 files changed, 223 insertions(+), 321 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 19124d09d8..cfe42284a6 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -731,34 +731,28 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
return 0;
}
-static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void alsa_enable_out(HWVoiceOut *hw, bool enable)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = apdo->try_poll;
+ if (enable) {
+ bool poll_mode = apdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_out (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
- return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
+ ldebug("enabling voice\n");
+ if (poll_mode && alsa_poll_out(hw)) {
+ poll_mode = 0;
}
-
- case VOICE_DISABLE:
- ldebug ("disabling voice\n");
+ hw->poll_mode = poll_mode;
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
+ } else {
+ ldebug("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
- alsa_fini_poll (&alsa->pollhlp);
+ alsa_fini_poll(&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
}
-
- return -1;
}
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -841,35 +835,29 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
return pos;
}
-static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void alsa_enable_in(HWVoiceIn *hw, bool enable)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = apdo->try_poll;
+ if (enable) {
+ bool poll_mode = apdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_in (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
-
- return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+ ldebug("enabling voice\n");
+ if (poll_mode && alsa_poll_in(hw)) {
+ poll_mode = 0;
}
+ hw->poll_mode = poll_mode;
- case VOICE_DISABLE:
+ alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
+ } else {
ldebug ("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
- alsa_fini_poll (&alsa->pollhlp);
+ alsa_fini_poll(&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
+ alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
}
-
- return -1;
}
static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
@@ -924,12 +912,12 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.write = alsa_write,
- .ctl_out = alsa_ctl_out,
+ .enable_out = alsa_enable_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.read = alsa_read,
- .ctl_in = alsa_ctl_in,
+ .enable_in = alsa_enable_in,
};
static struct audio_driver alsa_audio_driver = {
diff --git a/audio/audio.c b/audio/audio.c
index fab1e35718..7128ee98dc 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -636,7 +636,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
total += isamp;
}
- if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ if (!hw->pcm_ops->volume_in) {
mixeng_volume (sw->buf, ret, &sw->vol);
}
@@ -723,7 +723,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
if (swlim) {
sw->conv (sw->buf, buf, swlim);
- if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ if (!sw->hw->pcm_ops->volume_out) {
mixeng_volume (sw->buf, swlim, &sw->vol);
}
}
@@ -890,7 +890,9 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_out(hw, VOICE_ENABLE);
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, true);
+ }
audio_reset_timer (s);
}
}
@@ -935,7 +937,9 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_in(hw, VOICE_ENABLE);
+ if (hw->pcm_ops->enable_in) {
+ hw->pcm_ops->enable_in(hw, true);
+ }
audio_reset_timer (s);
}
}
@@ -952,7 +956,9 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (nb_active == 1) {
hw->enabled = 0;
- hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
+ if (hw->pcm_ops->enable_in) {
+ hw->pcm_ops->enable_in(hw, false);
+ }
}
}
}
@@ -1105,7 +1111,9 @@ static void audio_run_out (AudioState *s)
#endif
hw->enabled = 0;
hw->pending_disable = 0;
- hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, false);
+ }
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = 0;
audio_recalc_and_notify_capture (sc->cap);
@@ -1470,15 +1478,18 @@ static void audio_vm_change_state_handler (void *opaque, int running,
AudioState *s = opaque;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
- int op = running ? VOICE_ENABLE : VOICE_DISABLE;
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
- hwo->pcm_ops->ctl_out(hwo, op);
+ if (hwo->pcm_ops->enable_out) {
+ hwo->pcm_ops->enable_out(hwo, running);
+ }
}
while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
- hwi->pcm_ops->ctl_in(hwi, op);
+ if (hwi->pcm_ops->enable_in) {
+ hwi->pcm_ops->enable_in(hwi, running);
+ }
}
audio_reset_timer (s);
}
@@ -1498,8 +1509,8 @@ static void free_audio_state(AudioState *s)
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
- if (hwo->enabled) {
- hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+ if (hwo->enabled && hwo->pcm_ops->enable_out) {
+ hwo->pcm_ops->enable_out(hwo, false);
}
hwo->pcm_ops->fini_out (hwo);
@@ -1515,8 +1526,8 @@ static void free_audio_state(AudioState *s)
}
QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
- if (hwi->enabled) {
- hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+ if (hwi->enabled && hwi->pcm_ops->enable_in) {
+ hwi->pcm_ops->enable_in(hwi, false);
}
hwi->pcm_ops->fini_in (hwi);
QLIST_REMOVE(hwi, entries);
@@ -1838,8 +1849,8 @@ void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
- if (hw->pcm_ops->ctl_out) {
- hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
+ if (hw->pcm_ops->volume_out) {
+ hw->pcm_ops->volume_out(hw, &sw->vol);
}
}
}
@@ -1853,8 +1864,8 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
- if (hw->pcm_ops->ctl_in) {
- hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
+ if (hw->pcm_ops->volume_in) {
+ hw->pcm_ops->volume_in(hw, &sw->vol);
}
}
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 778615ccaf..22a703c13e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -74,7 +74,6 @@ typedef struct HWVoiceOut {
size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
- int ctl_caps;
struct audio_pcm_ops *pcm_ops;
QLIST_ENTRY (HWVoiceOut) entries;
} HWVoiceOut;
@@ -96,7 +95,6 @@ typedef struct HWVoiceIn {
size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
- int ctl_caps;
struct audio_pcm_ops *pcm_ops;
QLIST_ENTRY (HWVoiceIn) entries;
} HWVoiceIn;
@@ -148,7 +146,6 @@ struct audio_driver {
int max_voices_in;
int voice_size_out;
int voice_size_in;
- int ctl_caps;
QLIST_ENTRY(audio_driver) next;
};
@@ -168,14 +165,16 @@ struct audio_pcm_ops {
* size may be smaller
*/
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
- int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+ void (*enable_out)(HWVoiceOut *hw, bool enable);
+ void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
- int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
+ void (*enable_in)(HWVoiceIn *hw, bool enable);
+ void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
};
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
@@ -251,12 +250,6 @@ void audio_rate_start(RateCtl *rate);
size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
size_t bytes_avail);
-#define VOICE_ENABLE 1
-#define VOICE_DISABLE 2
-#define VOICE_VOLUME 3
-
-#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
-
static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
{
return (dst >= src) ? (dst - src) : (len - src + dst);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 87c6d2d271..235d1acbbe 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -254,7 +254,6 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
hw->s = s;
hw->pcm_ops = drv->pcm_ops;
- hw->ctl_caps = drv->ctl_caps;
QLIST_INIT (&hw->sw_head);
#ifdef DAC
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 5cde42f982..1427c9f622 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -648,13 +648,12 @@ static void coreaudio_fini_out (HWVoiceOut *hw)
}
}
-static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void coreaudio_enable_out(HWVoiceOut *hw, bool enable)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
/* start playback */
if (!isPlaying(core->outputDeviceID)) {
status = AudioDeviceStart(core->outputDeviceID, core->ioprocid);
@@ -662,9 +661,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
coreaudio_logerr (status, "Could not resume playback\n");
}
}
- break;
-
- case VOICE_DISABLE:
+ } else {
/* stop playback */
if (!audio_is_cleaning_up()) {
if (isPlaying(core->outputDeviceID)) {
@@ -675,9 +672,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
}
}
}
- break;
}
- return 0;
}
static void *coreaudio_audio_init(Audiodev *dev)
@@ -695,7 +690,7 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.write = coreaudio_write,
.get_buffer_out = coreaudio_get_buffer_out,
.put_buffer_out = coreaudio_put_buffer_out_nowrite,
- .ctl_out = coreaudio_ctl_out
+ .enable_out = coreaudio_enable_out
};
static struct audio_driver coreaudio_audio_driver = {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 9960247814..d4a4757445 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -361,7 +361,7 @@ static int dsound_open (dsound *s)
return 0;
}
-static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void dsound_enable_out(HWVoiceOut *hw, bool enable)
{
HRESULT hr;
DWORD status;
@@ -371,18 +371,17 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
if (!dsb) {
dolog ("Attempt to control voice without a buffer\n");
- return 0;
+ return;
}
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (dsound_get_status_out (dsb, &status, s)) {
- return -1;
+ return;
}
if (status & DSBSTATUS_PLAYING) {
dolog ("warning: Voice is already playing\n");
- return 0;
+ return;
}
dsound_clear_sample (hw, dsb, s);
@@ -390,28 +389,24 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start playing buffer\n");
- return -1;
+ return;
}
- break;
-
- case VOICE_DISABLE:
+ } else {
if (dsound_get_status_out (dsb, &status, s)) {
- return -1;
+ return;
}
if (status & DSBSTATUS_PLAYING) {
hr = IDirectSoundBuffer_Stop (dsb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop playing buffer\n");
- return -1;
+ return;
}
}
else {
dolog ("warning: Voice is not playing\n");
}
- break;
}
- return 0;
}
static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
@@ -461,7 +456,7 @@ static size_t dsound_put_buffer_out(HWVoiceOut *hw, void *buf, size_t len)
return len;
}
-static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void dsound_enable_in(HWVoiceIn *hw, bool enable)
{
HRESULT hr;
DWORD status;
@@ -470,18 +465,17 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
if (!dscb) {
dolog ("Attempt to control capture voice without a buffer\n");
- return -1;
+ return;
}
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (dsound_get_status_in (dscb, &status)) {
- return -1;
+ return;
}
if (status & DSCBSTATUS_CAPTURING) {
dolog ("warning: Voice is already capturing\n");
- return 0;
+ return;
}
/* clear ?? */
@@ -489,28 +483,24 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start capturing\n");
- return -1;
+ return;
}
- break;
-
- case VOICE_DISABLE:
+ } else {
if (dsound_get_status_in (dscb, &status)) {
- return -1;
+ return;
}
if (status & DSCBSTATUS_CAPTURING) {
hr = IDirectSoundCaptureBuffer_Stop (dscb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop capturing\n");
- return -1;
+ return;
}
}
else {
dolog ("warning: Voice is not capturing\n");
}
- break;
}
- return 0;
}
static void *dsound_get_buffer_in(HWVoiceIn *hw, size_t *size)
@@ -674,14 +664,14 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.write = audio_generic_write,
.get_buffer_out = dsound_get_buffer_out,
.put_buffer_out = dsound_put_buffer_out,
- .ctl_out = dsound_ctl_out,
+ .enable_out = dsound_enable_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
.read = audio_generic_read,
.get_buffer_in = dsound_get_buffer_in,
.put_buffer_in = dsound_put_buffer_in,
- .ctl_in = dsound_ctl_in
+ .enable_in = dsound_enable_in,
};
static struct audio_driver dsound_audio_driver = {
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 9f1cc67df9..ec8a287f36 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -62,14 +62,13 @@ static void no_fini_out (HWVoiceOut *hw)
(void) hw;
}
-static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void no_enable_out(HWVoiceOut *hw, bool enable)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&no->rate);
}
- return 0;
}
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -96,14 +95,13 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
return bytes;
}
-static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void no_enable_in(HWVoiceIn *hw, bool enable)
{
NoVoiceIn *no = (NoVoiceIn *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&no->rate);
}
- return 0;
}
static void *no_audio_init(Audiodev *dev)
@@ -120,12 +118,12 @@ static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
.write = no_write,
- .ctl_out = no_ctl_out,
+ .enable_out = no_enable_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
.read = no_read,
- .ctl_in = no_ctl_in
+ .enable_in = no_enable_in
};
static struct audio_driver no_audio_driver = {
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 76c082d5e2..0c4451e972 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -563,60 +563,50 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
-static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void oss_enable_out(HWVoiceOut *hw, bool enable)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = opdo->try_poll;
+ if (enable) {
+ bool poll_mode = opdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode) {
- oss_poll_out (hw);
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
-
- if (!oss->mmapped) {
- return 0;
- }
+ ldebug("enabling voice\n");
+ if (poll_mode) {
+ oss_poll_out(hw);
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
- audio_pcm_info_clear_buf(
- &hw->info, hw->buf_emul, hw->mix_buf->size);
- trig = PCM_ENABLE_OUTPUT;
- if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- oss_logerr (
- errno,
- "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
- );
- return -1;
- }
+ if (!oss->mmapped) {
+ return;
}
- break;
- case VOICE_DISABLE:
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
+ trig = PCM_ENABLE_OUTPUT;
+ if (ioctl(oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+ oss_logerr(errno,
+ "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n");
+ return;
+ }
+ } else {
if (hw->poll_mode) {
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
hw->poll_mode = 0;
}
if (!oss->mmapped) {
- return 0;
+ return;
}
ldebug ("disabling voice\n");
trig = 0;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
- return -1;
+ return;
}
- break;
}
- return 0;
}
static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -703,32 +693,25 @@ static size_t oss_read(HWVoiceIn *hw, void *buf, size_t len)
return pos;
}
-static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void oss_enable_in(HWVoiceIn *hw, bool enable)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = opdo->try_poll;
+ if (enable) {
+ bool poll_mode = opdo->try_poll;
- if (poll_mode) {
- oss_poll_in (hw);
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
+ if (poll_mode) {
+ oss_poll_in(hw);
+ poll_mode = 0;
}
- break;
-
- case VOICE_DISABLE:
+ hw->poll_mode = poll_mode;
+ } else {
if (hw->poll_mode) {
hw->poll_mode = 0;
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
}
- break;
}
- return 0;
}
static void oss_init_per_direction(AudiodevOssPerDirectionOptions *opdo)
@@ -767,12 +750,12 @@ static struct audio_pcm_ops oss_pcm_ops = {
.write = oss_write,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
- .ctl_out = oss_ctl_out,
+ .enable_out = oss_enable_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.read = oss_read,
- .ctl_in = oss_ctl_in
+ .enable_in = oss_enable_in
};
static struct audio_driver oss_audio_driver = {
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 75fce53202..ed31f863f7 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -452,7 +452,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
}
}
-static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
{
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
@@ -463,49 +463,36 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
#endif
- switch (cmd) {
- case VOICE_VOLUME:
- {
- SWVoiceOut *sw;
- va_list ap;
-
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceOut *);
- va_end (ap);
-
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
-
- pa_threaded_mainloop_lock(c->mainloop);
-
- op = pa_context_set_sink_input_volume(c->context,
- pa_stream_get_index (pa->stream),
- &v, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_sink_input_volume() failed\n");
- } else {
- pa_operation_unref(op);
- }
-
- op = pa_context_set_sink_input_mute(c->context,
- pa_stream_get_index (pa->stream),
- sw->vol.mute, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_sink_input_mute() failed\n");
- } else {
- pa_operation_unref(op);
- }
-
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ v.channels = 2;
+ v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
+ v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ op = pa_context_set_sink_input_volume(c->context,
+ pa_stream_get_index(pa->stream),
+ &v, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_volume() failed\n");
+ } else {
+ pa_operation_unref(op);
}
- return 0;
+
+ op = pa_context_set_sink_input_mute(c->context,
+ pa_stream_get_index(pa->stream),
+ vol->mute, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_mute() failed\n");
+ } else {
+ pa_operation_unref(op);
+ }
+
+ pa_threaded_mainloop_unlock(c->mainloop);
}
-static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
{
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
@@ -516,46 +503,33 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
pa_cvolume_init (&v);
#endif
- switch (cmd) {
- case VOICE_VOLUME:
- {
- SWVoiceIn *sw;
- va_list ap;
-
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceIn *);
- va_end (ap);
-
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
-
- pa_threaded_mainloop_lock(c->mainloop);
-
- op = pa_context_set_source_output_volume(c->context,
- pa_stream_get_index(pa->stream),
- &v, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_source_output_volume() failed\n");
- } else {
- pa_operation_unref(op);
- }
-
- op = pa_context_set_source_output_mute(c->context,
- pa_stream_get_index (pa->stream),
- sw->vol.mute, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_source_output_mute() failed\n");
- } else {
- pa_operation_unref (op);
- }
-
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ v.channels = 2;
+ v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
+ v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ op = pa_context_set_source_output_volume(c->context,
+ pa_stream_get_index(pa->stream),
+ &v, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_volume() failed\n");
+ } else {
+ pa_operation_unref(op);
}
- return 0;
+
+ op = pa_context_set_source_output_mute(c->context,
+ pa_stream_get_index(pa->stream),
+ vol->mute, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_mute() failed\n");
+ } else {
+ pa_operation_unref(op);
+ }
+
+ pa_threaded_mainloop_unlock(c->mainloop);
}
static int qpa_validate_per_direction_opts(Audiodev *dev,
@@ -724,12 +698,12 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.write = qpa_write,
- .ctl_out = qpa_ctl_out,
+ .volume_out = qpa_volume_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.read = qpa_read,
- .ctl_in = qpa_ctl_in
+ .volume_in = qpa_volume_in
};
static struct audio_driver pa_audio_driver = {
@@ -743,7 +717,6 @@ static struct audio_driver pa_audio_driver = {
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (PAVoiceOut),
.voice_size_in = sizeof (PAVoiceIn),
- .ctl_caps = VOICE_VOLUME_CAP
};
static void register_audio_pa(void)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index f7ac8cd101..5c6bcfcb3e 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -285,20 +285,9 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
-static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void sdl_enable_out(HWVoiceOut *hw, bool enable)
{
- (void) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- SDL_PauseAudio (0);
- break;
-
- case VOICE_DISABLE:
- SDL_PauseAudio (1);
- break;
- }
- return 0;
+ SDL_PauseAudio(!enable);
}
static void *sdl_audio_init(Audiodev *dev)
@@ -334,7 +323,7 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.write = sdl_write,
.get_buffer_out = sdl_get_buffer_out,
.put_buffer_out = sdl_put_buffer_out_nowrite,
- .ctl_out = sdl_ctl_out,
+ .enable_out = sdl_enable_out,
};
static struct audio_driver sdl_audio_driver = {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 4ce4f94c6d..9860f9c5e1 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -153,22 +153,20 @@ static size_t line_out_put_buffer(HWVoiceOut *hw, void *buf, size_t size)
return size;
}
-static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
+static void line_out_enable(HWVoiceOut *hw, bool enable)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (out->active) {
- break;
+ return;
}
out->active = 1;
audio_rate_start(&out->rate);
spice_server_playback_start (&out->sin);
- break;
- case VOICE_DISABLE:
+ } else {
if (!out->active) {
- break;
+ return;
}
out->active = 0;
if (out->frame) {
@@ -177,29 +175,21 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
out->frame = NULL;
}
spice_server_playback_stop (&out->sin);
- break;
- case VOICE_VOLUME:
- {
-#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
- SWVoiceOut *sw;
- va_list ap;
- uint16_t vol[2];
-
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceOut *);
- va_end (ap);
-
- vol[0] = sw->vol.l / ((1ULL << 16) + 1);
- vol[1] = sw->vol.r / ((1ULL << 16) + 1);
- spice_server_playback_set_volume (&out->sin, 2, vol);
- spice_server_playback_set_mute (&out->sin, sw->vol.mute);
-#endif
- break;
- }
}
+}
- return 0;
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+ uint16_t svol[2];
+
+ svol[0] = vol->l / ((1ULL << 16) + 1);
+ svol[1] = vol->r / ((1ULL << 16) + 1);
+ spice_server_playback_set_volume(&out->sin, 2, svol);
+ spice_server_playback_set_mute(&out->sin, vol->mute);
}
+#endif
/* record */
@@ -251,48 +241,38 @@ static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
return ready << 2;
}
-static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
+static void line_in_enable(HWVoiceIn *hw, bool enable)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (in->active) {
- break;
+ return;
}
in->active = 1;
audio_rate_start(&in->rate);
spice_server_record_start (&in->sin);
- break;
- case VOICE_DISABLE:
+ } else {
if (!in->active) {
- break;
+ return;
}
in->active = 0;
spice_server_record_stop (&in->sin);
- break;
- case VOICE_VOLUME:
- {
-#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
- SWVoiceIn *sw;
- va_list ap;
- uint16_t vol[2];
-
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceIn *);
- va_end (ap);
-
- vol[0] = sw->vol.l / ((1ULL << 16) + 1);
- vol[1] = sw->vol.r / ((1ULL << 16) + 1);
- spice_server_record_set_volume (&in->sin, 2, vol);
- spice_server_record_set_mute (&in->sin, sw->vol.mute);
-#endif
- break;
- }
}
+}
- return 0;
+#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
+static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw);
+ uint16_t svol[2];
+
+ svol[0] = vol->l / ((1ULL << 16) + 1);
+ svol[1] = vol->r / ((1ULL << 16) + 1);
+ spice_server_record_set_volume(&in->sin, 2, svol);
+ spice_server_record_set_mute(&in->sin, vol->mute);
}
+#endif
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
@@ -300,12 +280,19 @@ static struct audio_pcm_ops audio_callbacks = {
.write = audio_generic_write,
.get_buffer_out = line_out_get_buffer,
.put_buffer_out = line_out_put_buffer,
- .ctl_out = line_out_ctl,
+ .enable_out = line_out_enable,
+#if (SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && \
+ (SPICE_INTERFACE_PLAYBACK_MINOR >= 2)
+ .volume_out = line_out_volume,
+#endif
.init_in = line_in_init,
.fini_in = line_in_fini,
.read = line_in_read,
- .ctl_in = line_in_ctl,
+ .enable_in = line_in_enable,
+#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
+ .volume_in = line_in_volume,
+#endif
};
static struct audio_driver spice_audio_driver = {
@@ -318,9 +305,6 @@ static struct audio_driver spice_audio_driver = {
.max_voices_in = 1,
.voice_size_out = sizeof (SpiceVoiceOut),
.voice_size_in = sizeof (SpiceVoiceIn),
-#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
- .ctl_caps = VOICE_VOLUME_CAP
-#endif
};
void qemu_spice_audio_init (void)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 85f5ff9c28..47efdc1b1e 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -173,14 +173,13 @@ static void wav_fini_out (HWVoiceOut *hw)
wav->f = NULL;
}
-static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void wav_enable_out(HWVoiceOut *hw, bool enable)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&wav->rate);
}
- return 0;
}
static void *wav_audio_init(Audiodev *dev)
@@ -198,7 +197,7 @@ static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.write = wav_write_out,
- .ctl_out = wav_ctl_out,
+ .enable_out = wav_enable_out,
};
static struct audio_driver wav_audio_driver = {
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (12 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 14/24] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-23 13:08 ` Markus Armbruster
2019-09-19 21:24 ` [PATCH v4 16/24] audio: make mixeng optional Kővágó, Zoltán
` (6 subsequent siblings)
20 siblings, 1 reply; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
This will allow us to disable mixeng when we use a decent backend.
Disabling mixeng have a few advantages:
* we no longer convert the audio output from one format to another, when
the underlying audio system would just convert it to a third format.
We no longer convert, only the underlying system, when needed.
* the underlying system probably has better resampling and sample format
converting methods anyway...
* we may support formats that the mixeng currently does not support (S24
or float samples, more than two channels)
* when using an audio server (like pulseaudio) different sound card
outputs will show up as separate streams, even if we use only one
backend
Disadvantages:
* audio capturing no longer works (wavcapture, and vnc audio extension)
* some backends only support a single playback stream or very picky
about the audio format. In this case we can't disable mixeng.
However mixeng is not removed, only made optional, so this shouldn't be
a big concern.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
Changes from v1:
* renamed mixeng to mixing-engine
qapi/audio.json | 5 +++++
qemu-options.hx | 6 ++++++
2 files changed, 11 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index 9fefdf5186..0535eff794 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -11,6 +11,10 @@
# General audio backend options that are used for both playback and
# recording.
#
+# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
+# set to off, fixed-settings must be also off. Not every backend
+# compatible with the off setting (default on, since 4.2)
+#
# @fixed-settings: use fixed settings for host input/output. When off,
# frequency, channels and format must not be
# specified (default true)
@@ -31,6 +35,7 @@
##
{ 'struct': 'AudiodevPerDirectionOptions',
'data': {
+ '*mixing-engine': 'bool',
'*fixed-settings': 'bool',
'*frequency': 'uint32',
'*channels': 'uint32',
diff --git a/qemu-options.hx b/qemu-options.hx
index bbfd936d29..395427422a 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
" specifies the audio backend to use\n"
" id= identifier of the backend\n"
" timer-period= timer period in microseconds\n"
+ " in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n"
" in|out.fixed-settings= use fixed settings for host audio\n"
" in|out.frequency= frequency to use with fixed settings\n"
" in|out.channels= number of channels to use with fixed settings\n"
@@ -503,6 +504,11 @@ Identifies the audio backend.
Sets the timer @var{period} used by the audio subsystem in microseconds.
Default is 10000 (10 ms).
+@item in|out.mixing-engine=on|off
+Use QEMU's mixing engine to mix all streams inside QEMU. When off,
+@var{fixed-settings} must be off too. Not every backend is fully
+compatible with the off setting. Default is on.
+
@item in|out.fixed-settings=on|off
Use fixed settings for host audio. When off, it will change based on
how the guest opens the sound card. In this case you must not specify
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-19 21:24 ` [PATCH v4 15/24] audio: add mixing-engine option (documentation) Kővágó, Zoltán
@ 2019-09-23 13:08 ` Markus Armbruster
2019-09-24 0:21 ` Zoltán Kővágó
0 siblings, 1 reply; 35+ messages in thread
From: Markus Armbruster @ 2019-09-23 13:08 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> This will allow us to disable mixeng when we use a decent backend.
>
> Disabling mixeng have a few advantages:
> * we no longer convert the audio output from one format to another, when
> the underlying audio system would just convert it to a third format.
> We no longer convert, only the underlying system, when needed.
> * the underlying system probably has better resampling and sample format
> converting methods anyway...
> * we may support formats that the mixeng currently does not support (S24
> or float samples, more than two channels)
> * when using an audio server (like pulseaudio) different sound card
> outputs will show up as separate streams, even if we use only one
> backend
>
> Disadvantages:
> * audio capturing no longer works (wavcapture, and vnc audio extension)
> * some backends only support a single playback stream or very picky
> about the audio format. In this case we can't disable mixeng.
>
> However mixeng is not removed, only made optional, so this shouldn't be
> a big concern.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
>
> Notes:
> Changes from v1:
>
> * renamed mixeng to mixing-engine
>
> qapi/audio.json | 5 +++++
> qemu-options.hx | 6 ++++++
> 2 files changed, 11 insertions(+)
>
> diff --git a/qapi/audio.json b/qapi/audio.json
> index 9fefdf5186..0535eff794 100644
> --- a/qapi/audio.json
> +++ b/qapi/audio.json
> @@ -11,6 +11,10 @@
> # General audio backend options that are used for both playback and
> # recording.
> #
> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
> +# set to off, fixed-settings must be also off. Not every backend
> +# compatible with the off setting (default on, since 4.2)
> +#
Last sentence no verb.
Which backends are compatible?
What happens when you try the off setting with incompatible backends?
> # @fixed-settings: use fixed settings for host input/output. When off,
> # frequency, channels and format must not be
> # specified (default true)
> @@ -31,6 +35,7 @@
> ##
> { 'struct': 'AudiodevPerDirectionOptions',
> 'data': {
> + '*mixing-engine': 'bool',
> '*fixed-settings': 'bool',
> '*frequency': 'uint32',
> '*channels': 'uint32',
> diff --git a/qemu-options.hx b/qemu-options.hx
> index bbfd936d29..395427422a 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
> " specifies the audio backend to use\n"
> " id= identifier of the backend\n"
> " timer-period= timer period in microseconds\n"
> + " in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n"
> " in|out.fixed-settings= use fixed settings for host audio\n"
> " in|out.frequency= frequency to use with fixed settings\n"
> " in|out.channels= number of channels to use with fixed settings\n"
> @@ -503,6 +504,11 @@ Identifies the audio backend.
> Sets the timer @var{period} used by the audio subsystem in microseconds.
> Default is 10000 (10 ms).
>
> +@item in|out.mixing-engine=on|off
> +Use QEMU's mixing engine to mix all streams inside QEMU. When off,
> +@var{fixed-settings} must be off too. Not every backend is fully
> +compatible with the off setting. Default is on.
> +
> @item in|out.fixed-settings=on|off
> Use fixed settings for host audio. When off, it will change based on
> how the guest opens the sound card. In this case you must not specify
^ permalink raw reply [flat|nested] 35+ messages in thread* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-23 13:08 ` Markus Armbruster
@ 2019-09-24 0:21 ` Zoltán Kővágó
2019-09-25 9:49 ` Markus Armbruster
0 siblings, 1 reply; 35+ messages in thread
From: Zoltán Kővágó @ 2019-09-24 0:21 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Gerd Hoffmann
On 2019-09-23 15:08, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> This will allow us to disable mixeng when we use a decent backend.
>>
>> Disabling mixeng have a few advantages:
>> * we no longer convert the audio output from one format to another, when
>> the underlying audio system would just convert it to a third format.
>> We no longer convert, only the underlying system, when needed.
>> * the underlying system probably has better resampling and sample format
>> converting methods anyway...
>> * we may support formats that the mixeng currently does not support (S24
>> or float samples, more than two channels)
>> * when using an audio server (like pulseaudio) different sound card
>> outputs will show up as separate streams, even if we use only one
>> backend
>>
>> Disadvantages:
>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>> * some backends only support a single playback stream or very picky
>> about the audio format. In this case we can't disable mixeng.
>>
>> However mixeng is not removed, only made optional, so this shouldn't be
>> a big concern.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>>
>> Notes:
>> Changes from v1:
>>
>> * renamed mixeng to mixing-engine
>>
>> qapi/audio.json | 5 +++++
>> qemu-options.hx | 6 ++++++
>> 2 files changed, 11 insertions(+)
>>
>> diff --git a/qapi/audio.json b/qapi/audio.json
>> index 9fefdf5186..0535eff794 100644
>> --- a/qapi/audio.json
>> +++ b/qapi/audio.json
>> @@ -11,6 +11,10 @@
>> # General audio backend options that are used for both playback and
>> # recording.
>> #
>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>> +# set to off, fixed-settings must be also off. Not every backend
>> +# compatible with the off setting (default on, since 4.2)
>> +#
>
> Last sentence no verb.
>
> Which backends are compatible?
Actually that's a simplification, it depends on a few things. When
mixeng is off, qemu will try to use the same format as the emulated
sound card, and if the backend doesn't support that format, it won't
work (no audio). Also attaching multiple sound cards to the same
audiodev might not work, if the backend doesn't support multiple
playback streams. If you use pulseaudio, it'll work without problems,
if you use alsa, it depends on your device. If you use a hw: device
directly, you'll likely only be able to use one emulated sound card with
a few selected audio formats. If you use dmix: (and plug), alsa will
handle the conversion and mixing, so it will work no matter what format
the emulated sound card uses. With OSS the situation is probably
similar, it depends on the kernel/hw what works and what not. wav and
spice certainly doesn't support multiple streams. I'm not completely
sure about the other backends right now, but I think dsound and
coreaudio can handle the necessary sample format conversions and mixing.
> What happens when you try the off setting with incompatible backends?
See above.
>
>> # @fixed-settings: use fixed settings for host input/output. When off,
>> # frequency, channels and format must not be
>> # specified (default true)
>> @@ -31,6 +35,7 @@
>> ##
>> { 'struct': 'AudiodevPerDirectionOptions',
>> 'data': {
>> + '*mixing-engine': 'bool',
>> '*fixed-settings': 'bool',
>> '*frequency': 'uint32',
>> '*channels': 'uint32',
>> diff --git a/qemu-options.hx b/qemu-options.hx
>> index bbfd936d29..395427422a 100644
>> --- a/qemu-options.hx
>> +++ b/qemu-options.hx
>> @@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
>> " specifies the audio backend to use\n"
>> " id= identifier of the backend\n"
>> " timer-period= timer period in microseconds\n"
>> + " in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n"
>> " in|out.fixed-settings= use fixed settings for host audio\n"
>> " in|out.frequency= frequency to use with fixed settings\n"
>> " in|out.channels= number of channels to use with fixed settings\n"
>> @@ -503,6 +504,11 @@ Identifies the audio backend.
>> Sets the timer @var{period} used by the audio subsystem in microseconds.
>> Default is 10000 (10 ms).
>>
>> +@item in|out.mixing-engine=on|off
>> +Use QEMU's mixing engine to mix all streams inside QEMU. When off,
>> +@var{fixed-settings} must be off too. Not every backend is fully
>> +compatible with the off setting. Default is on.
>> +
>> @item in|out.fixed-settings=on|off
>> Use fixed settings for host audio. When off, it will change based on
>> how the guest opens the sound card. In this case you must not specify
^ permalink raw reply [flat|nested] 35+ messages in thread* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-24 0:21 ` Zoltán Kővágó
@ 2019-09-25 9:49 ` Markus Armbruster
2019-09-29 19:07 ` Zoltán Kővágó
0 siblings, 1 reply; 35+ messages in thread
From: Markus Armbruster @ 2019-09-25 9:49 UTC (permalink / raw)
To: Zoltán Kővágó; +Cc: qemu-devel, Gerd Hoffmann
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-09-23 15:08, Markus Armbruster wrote:
>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>
>>> This will allow us to disable mixeng when we use a decent backend.
>>>
>>> Disabling mixeng have a few advantages:
>>> * we no longer convert the audio output from one format to another, when
>>> the underlying audio system would just convert it to a third format.
>>> We no longer convert, only the underlying system, when needed.
>>> * the underlying system probably has better resampling and sample format
>>> converting methods anyway...
>>> * we may support formats that the mixeng currently does not support (S24
>>> or float samples, more than two channels)
>>> * when using an audio server (like pulseaudio) different sound card
>>> outputs will show up as separate streams, even if we use only one
>>> backend
>>>
>>> Disadvantages:
>>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>>> * some backends only support a single playback stream or very picky
>>> about the audio format. In this case we can't disable mixeng.
>>>
>>> However mixeng is not removed, only made optional, so this shouldn't be
>>> a big concern.
>>>
>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>> ---
>>>
>>> Notes:
>>> Changes from v1:
>>> * renamed mixeng to mixing-engine
>>>
>>> qapi/audio.json | 5 +++++
>>> qemu-options.hx | 6 ++++++
>>> 2 files changed, 11 insertions(+)
>>>
>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>> index 9fefdf5186..0535eff794 100644
>>> --- a/qapi/audio.json
>>> +++ b/qapi/audio.json
>>> @@ -11,6 +11,10 @@
>>> # General audio backend options that are used for both playback and
>>> # recording.
>>> #
>>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>>> +# set to off, fixed-settings must be also off. Not every backend
>>> +# compatible with the off setting (default on, since 4.2)
>>> +#
>>
>> Last sentence no verb.
>>
>> Which backends are compatible?
>
> Actually that's a simplification, it depends on a few things. When
> mixeng is off, qemu will try to use the same format as the emulated
> sound card, and if the backend doesn't support that format, it won't
> work (no audio). Also attaching multiple sound cards to the same
> audiodev might not work, if the backend doesn't support multiple
> playback streams. If you use pulseaudio, it'll work without problems,
> if you use alsa, it depends on your device. If you use a hw: device
> directly, you'll likely only be able to use one emulated sound card
> with a few selected audio formats. If you use dmix: (and plug), alsa
> will handle the conversion and mixing, so it will work no matter what
> format the emulated sound card uses. With OSS the situation is
> probably similar, it depends on the kernel/hw what works and what not.
> wav and spice certainly doesn't support multiple streams. I'm not
> completely sure about the other backends right now, but I think dsound
> and coreaudio can handle the necessary sample format conversions and
> mixing.
>
>> What happens when you try the off setting with incompatible backends?
> See above.
What happens *exactly*?
I'm asking because I'm concerned about the user experience. When a user
asks for a combination of things QEMU can't provide, such as mixeng off
with an incompatible backend, QEMU should fail with a suitable error
message. Does it?
Sometimes rejecting non-working configurations is impractical. Is it
here?
If yes, we should call out the problematic configurations in
documentation.
[...]
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-25 9:49 ` Markus Armbruster
@ 2019-09-29 19:07 ` Zoltán Kővágó
2019-10-01 6:23 ` Markus Armbruster
0 siblings, 1 reply; 35+ messages in thread
From: Zoltán Kővágó @ 2019-09-29 19:07 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Gerd Hoffmann
On 2019-09-25 11:49, Markus Armbruster wrote:
> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>
>> On 2019-09-23 15:08, Markus Armbruster wrote:
>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>
>>>> This will allow us to disable mixeng when we use a decent backend.
>>>>
>>>> Disabling mixeng have a few advantages:
>>>> * we no longer convert the audio output from one format to another, when
>>>> the underlying audio system would just convert it to a third format.
>>>> We no longer convert, only the underlying system, when needed.
>>>> * the underlying system probably has better resampling and sample format
>>>> converting methods anyway...
>>>> * we may support formats that the mixeng currently does not support (S24
>>>> or float samples, more than two channels)
>>>> * when using an audio server (like pulseaudio) different sound card
>>>> outputs will show up as separate streams, even if we use only one
>>>> backend
>>>>
>>>> Disadvantages:
>>>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>>>> * some backends only support a single playback stream or very picky
>>>> about the audio format. In this case we can't disable mixeng.
>>>>
>>>> However mixeng is not removed, only made optional, so this shouldn't be
>>>> a big concern.
>>>>
>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>> ---
>>>>
>>>> Notes:
>>>> Changes from v1:
>>>> * renamed mixeng to mixing-engine
>>>>
>>>> qapi/audio.json | 5 +++++
>>>> qemu-options.hx | 6 ++++++
>>>> 2 files changed, 11 insertions(+)
>>>>
>>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>>> index 9fefdf5186..0535eff794 100644
>>>> --- a/qapi/audio.json
>>>> +++ b/qapi/audio.json
>>>> @@ -11,6 +11,10 @@
>>>> # General audio backend options that are used for both playback and
>>>> # recording.
>>>> #
>>>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>>>> +# set to off, fixed-settings must be also off. Not every backend
>>>> +# compatible with the off setting (default on, since 4.2)
>>>> +#
>>>
>>> Last sentence no verb.
>>>
>>> Which backends are compatible?
>>
>> Actually that's a simplification, it depends on a few things. When
>> mixeng is off, qemu will try to use the same format as the emulated
>> sound card, and if the backend doesn't support that format, it won't
>> work (no audio). Also attaching multiple sound cards to the same
>> audiodev might not work, if the backend doesn't support multiple
>> playback streams. If you use pulseaudio, it'll work without problems,
>> if you use alsa, it depends on your device. If you use a hw: device
>> directly, you'll likely only be able to use one emulated sound card
>> with a few selected audio formats. If you use dmix: (and plug), alsa
>> will handle the conversion and mixing, so it will work no matter what
>> format the emulated sound card uses. With OSS the situation is
>> probably similar, it depends on the kernel/hw what works and what not.
>> wav and spice certainly doesn't support multiple streams. I'm not
>> completely sure about the other backends right now, but I think dsound
>> and coreaudio can handle the necessary sample format conversions and
>> mixing.
>>
>>> What happens when you try the off setting with incompatible backends?
>> See above.
>
> What happens *exactly*?
>
> I'm asking because I'm concerned about the user experience. When a user
> asks for a combination of things QEMU can't provide, such as mixeng off
> with an incompatible backend, QEMU should fail with a suitable error
> message. Does it?
Error handling is not the best in the audio subsystem, if something
fails it generally just prints a warning to the console and continues,
and something will happen... For example, this is what happens when I
try to open one hw device twice. I ran qemu with:
-audiodev
alsa,id=foo,in.dev=hw:1,,0,out.mixing-engine=off,out.dev=hw:1,,0 -device
piix4-usb-uhci -device usb-audio,audiodev=foo -device AC97,audiodev=foo
When the guest tried to initialize the AC97 card, I got an error:
alsa: Could not initialize DAC
alsa: Failed to open `hw:1,0':
alsa: Reason: Device or resource busy
And it just continued. And the sound worked, but with wrong sample rate
(AC97 wants 44100 Hz, but USB audio previously opened the alsa device
with 48000 Hz). I'll fix this bug in the next revision,
audio_pcm_hw_add_* shouldn't fall back to other HWs without mixeng. But
even with that, the result will be that one emulated sound card will
work and the other won't.
It's not ideal, but fixing it would require a lot of effort. Right now,
if you specify an invalid audiodev for alsa (even with mixeng), it'll
just print an error to the console and continue without audio.
> Sometimes rejecting non-working configurations is impractical. Is it
> here?
I think it is. It depends on the backend, its settings, the frontend
(emulated sound card), and how the guest uses it. We currently don't
know what formats does a backend support, what formats can a frontend
produce, and even if we would know that, just because a frontend can
produce a format that the backend doesn't understand doesn't mean that
it will actually do it. For example, right now with this patch series
applied, usb-audio can produce 7.1 audio. If we want to be strict, it
means we can only use it with backends that support at least 8 channels,
even if the user only wants to use stereo audio.
> If yes, we should call out the problematic configurations in
> documentation.
I think we should rather list known working configurations, and leave
the others as "try at your own risk" because there's too many things
that can go wrong. (pulseaudio will work, alsa with dmix too. Need to
check coreaudio, dsound and oss. spice and wavcapture won't work.)
Regards,
Zoltan
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-09-29 19:07 ` Zoltán Kővágó
@ 2019-10-01 6:23 ` Markus Armbruster
2019-10-03 23:27 ` Zoltán Kővágó
0 siblings, 1 reply; 35+ messages in thread
From: Markus Armbruster @ 2019-10-01 6:23 UTC (permalink / raw)
To: Zoltán Kővágó; +Cc: qemu-devel, Gerd Hoffmann
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-09-25 11:49, Markus Armbruster wrote:
>> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>>
>>> On 2019-09-23 15:08, Markus Armbruster wrote:
>>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>>
>>>>> This will allow us to disable mixeng when we use a decent backend.
>>>>>
>>>>> Disabling mixeng have a few advantages:
>>>>> * we no longer convert the audio output from one format to another, when
>>>>> the underlying audio system would just convert it to a third format.
>>>>> We no longer convert, only the underlying system, when needed.
>>>>> * the underlying system probably has better resampling and sample format
>>>>> converting methods anyway...
>>>>> * we may support formats that the mixeng currently does not support (S24
>>>>> or float samples, more than two channels)
>>>>> * when using an audio server (like pulseaudio) different sound card
>>>>> outputs will show up as separate streams, even if we use only one
>>>>> backend
>>>>>
>>>>> Disadvantages:
>>>>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>>>>> * some backends only support a single playback stream or very picky
>>>>> about the audio format. In this case we can't disable mixeng.
>>>>>
>>>>> However mixeng is not removed, only made optional, so this shouldn't be
>>>>> a big concern.
>>>>>
>>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>> ---
>>>>>
>>>>> Notes:
>>>>> Changes from v1:
>>>>> * renamed mixeng to mixing-engine
>>>>>
>>>>> qapi/audio.json | 5 +++++
>>>>> qemu-options.hx | 6 ++++++
>>>>> 2 files changed, 11 insertions(+)
>>>>>
>>>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>>>> index 9fefdf5186..0535eff794 100644
>>>>> --- a/qapi/audio.json
>>>>> +++ b/qapi/audio.json
>>>>> @@ -11,6 +11,10 @@
>>>>> # General audio backend options that are used for both playback and
>>>>> # recording.
>>>>> #
>>>>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>>>>> +# set to off, fixed-settings must be also off. Not every backend
>>>>> +# compatible with the off setting (default on, since 4.2)
>>>>> +#
>>>>
>>>> Last sentence no verb.
>>>>
>>>> Which backends are compatible?
>>>
>>> Actually that's a simplification, it depends on a few things. When
>>> mixeng is off, qemu will try to use the same format as the emulated
>>> sound card, and if the backend doesn't support that format, it won't
>>> work (no audio). Also attaching multiple sound cards to the same
>>> audiodev might not work, if the backend doesn't support multiple
>>> playback streams. If you use pulseaudio, it'll work without problems,
>>> if you use alsa, it depends on your device. If you use a hw: device
>>> directly, you'll likely only be able to use one emulated sound card
>>> with a few selected audio formats. If you use dmix: (and plug), alsa
>>> will handle the conversion and mixing, so it will work no matter what
>>> format the emulated sound card uses. With OSS the situation is
>>> probably similar, it depends on the kernel/hw what works and what not.
>>> wav and spice certainly doesn't support multiple streams. I'm not
>>> completely sure about the other backends right now, but I think dsound
>>> and coreaudio can handle the necessary sample format conversions and
>>> mixing.
>>>
>>>> What happens when you try the off setting with incompatible backends?
>>> See above.
>>
>> What happens *exactly*?
>>
>> I'm asking because I'm concerned about the user experience. When a user
>> asks for a combination of things QEMU can't provide, such as mixeng off
>> with an incompatible backend, QEMU should fail with a suitable error
>> message. Does it?
>
> Error handling is not the best in the audio subsystem, if something
> fails it generally just prints a warning to the console and continues,
> and something will happen... For example, this is what happens when I
> try to open one hw device twice. I ran qemu with:
>
> -audiodev
> alsa,id=foo,in.dev=hw:1,,0,out.mixing-engine=off,out.dev=hw:1,,0
> -device piix4-usb-uhci -device usb-audio,audiodev=foo -device
> AC97,audiodev=foo
>
> When the guest tried to initialize the AC97 card, I got an error:
>
> alsa: Could not initialize DAC
> alsa: Failed to open `hw:1,0':
> alsa: Reason: Device or resource busy
>
> And it just continued. And the sound worked, but with wrong sample
> rate (AC97 wants 44100 Hz, but USB audio previously opened the alsa
> device with 48000 Hz). I'll fix this bug in the next revision,
> audio_pcm_hw_add_* shouldn't fall back to other HWs without mixeng.
> But even with that, the result will be that one emulated sound card
> will work and the other won't.
>
> It's not ideal, but fixing it would require a lot of effort. Right
> now, if you specify an invalid audiodev for alsa (even with mixeng),
> it'll just print an error to the console and continue without audio.
Should we document this general error handling deficiency somehow?
>> Sometimes rejecting non-working configurations is impractical. Is it
>> here?
>
> I think it is. It depends on the backend, its settings, the frontend
> (emulated sound card), and how the guest uses it. We currently don't
> know what formats does a backend support, what formats can a frontend
> produce, and even if we would know that, just because a frontend can
> produce a format that the backend doesn't understand doesn't mean that
> it will actually do it. For example, right now with this patch series
> applied, usb-audio can produce 7.1 audio. If we want to be strict, it
> means we can only use it with backends that support at least 8
> channels, even if the user only wants to use stereo audio.
>
>> If yes, we should call out the problematic configurations in
>> documentation.
>
> I think we should rather list known working configurations, and leave
> the others as "try at your own risk" because there's too many things
> that can go wrong. (pulseaudio will work, alsa with dmix too. Need
> to check coreaudio, dsound and oss. spice and wavcapture won't work.)
Far from ideal, but better than nothing.
Possibly naive idea: what about automatically falling back to mixeng on
when mixeng off doesn't work? Requires detecting "doesn't work", which
I understand just isn't there. Any other reasons why this couldn't be
done? Way out of scope for this series, of course.
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-10-01 6:23 ` Markus Armbruster
@ 2019-10-03 23:27 ` Zoltán Kővágó
2019-10-07 8:21 ` Markus Armbruster
0 siblings, 1 reply; 35+ messages in thread
From: Zoltán Kővágó @ 2019-10-03 23:27 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Gerd Hoffmann
On 2019-10-01 08:23, Markus Armbruster wrote:
> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>
>> On 2019-09-25 11:49, Markus Armbruster wrote:
>>> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>>>
>>>> On 2019-09-23 15:08, Markus Armbruster wrote:
>>>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>>>
>>>>>> This will allow us to disable mixeng when we use a decent backend.
>>>>>>
>>>>>> Disabling mixeng have a few advantages:
>>>>>> * we no longer convert the audio output from one format to another, when
>>>>>> the underlying audio system would just convert it to a third format.
>>>>>> We no longer convert, only the underlying system, when needed.
>>>>>> * the underlying system probably has better resampling and sample format
>>>>>> converting methods anyway...
>>>>>> * we may support formats that the mixeng currently does not support (S24
>>>>>> or float samples, more than two channels)
>>>>>> * when using an audio server (like pulseaudio) different sound card
>>>>>> outputs will show up as separate streams, even if we use only one
>>>>>> backend
>>>>>>
>>>>>> Disadvantages:
>>>>>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>>>>>> * some backends only support a single playback stream or very picky
>>>>>> about the audio format. In this case we can't disable mixeng.
>>>>>>
>>>>>> However mixeng is not removed, only made optional, so this shouldn't be
>>>>>> a big concern.
>>>>>>
>>>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>>> ---
>>>>>>
>>>>>> Notes:
>>>>>> Changes from v1:
>>>>>> * renamed mixeng to mixing-engine
>>>>>>
>>>>>> qapi/audio.json | 5 +++++
>>>>>> qemu-options.hx | 6 ++++++
>>>>>> 2 files changed, 11 insertions(+)
>>>>>>
>>>>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>>>>> index 9fefdf5186..0535eff794 100644
>>>>>> --- a/qapi/audio.json
>>>>>> +++ b/qapi/audio.json
>>>>>> @@ -11,6 +11,10 @@
>>>>>> # General audio backend options that are used for both playback and
>>>>>> # recording.
>>>>>> #
>>>>>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>>>>>> +# set to off, fixed-settings must be also off. Not every backend
>>>>>> +# compatible with the off setting (default on, since 4.2)
>>>>>> +#
>>>>>
>>>>> Last sentence no verb.
>>>>>
>>>>> Which backends are compatible?
>>>>
>>>> Actually that's a simplification, it depends on a few things. When
>>>> mixeng is off, qemu will try to use the same format as the emulated
>>>> sound card, and if the backend doesn't support that format, it won't
>>>> work (no audio). Also attaching multiple sound cards to the same
>>>> audiodev might not work, if the backend doesn't support multiple
>>>> playback streams. If you use pulseaudio, it'll work without problems,
>>>> if you use alsa, it depends on your device. If you use a hw: device
>>>> directly, you'll likely only be able to use one emulated sound card
>>>> with a few selected audio formats. If you use dmix: (and plug), alsa
>>>> will handle the conversion and mixing, so it will work no matter what
>>>> format the emulated sound card uses. With OSS the situation is
>>>> probably similar, it depends on the kernel/hw what works and what not.
>>>> wav and spice certainly doesn't support multiple streams. I'm not
>>>> completely sure about the other backends right now, but I think dsound
>>>> and coreaudio can handle the necessary sample format conversions and
>>>> mixing.
>>>>
>>>>> What happens when you try the off setting with incompatible backends?
>>>> See above.
>>>
>>> What happens *exactly*?
>>>
>>> I'm asking because I'm concerned about the user experience. When a user
>>> asks for a combination of things QEMU can't provide, such as mixeng off
>>> with an incompatible backend, QEMU should fail with a suitable error
>>> message. Does it?
>>
>> Error handling is not the best in the audio subsystem, if something
>> fails it generally just prints a warning to the console and continues,
>> and something will happen... For example, this is what happens when I
>> try to open one hw device twice. I ran qemu with:
>>
>> -audiodev
>> alsa,id=foo,in.dev=hw:1,,0,out.mixing-engine=off,out.dev=hw:1,,0
>> -device piix4-usb-uhci -device usb-audio,audiodev=foo -device
>> AC97,audiodev=foo
>>
>> When the guest tried to initialize the AC97 card, I got an error:
>>
>> alsa: Could not initialize DAC
>> alsa: Failed to open `hw:1,0':
>> alsa: Reason: Device or resource busy
>>
>> And it just continued. And the sound worked, but with wrong sample
>> rate (AC97 wants 44100 Hz, but USB audio previously opened the alsa
>> device with 48000 Hz). I'll fix this bug in the next revision,
>> audio_pcm_hw_add_* shouldn't fall back to other HWs without mixeng.
>> But even with that, the result will be that one emulated sound card
>> will work and the other won't.
>>
>> It's not ideal, but fixing it would require a lot of effort. Right
>> now, if you specify an invalid audiodev for alsa (even with mixeng),
>> it'll just print an error to the console and continue without audio.
>
> Should we document this general error handling deficiency somehow?
It could be useful to mention it, especially for people who write script
and expect qemu to fail if they specify an invalid parameter. I'm not
sure how in-depth this explanation should be however.
>>> Sometimes rejecting non-working configurations is impractical. Is it
>>> here?
>>
>> I think it is. It depends on the backend, its settings, the frontend
>> (emulated sound card), and how the guest uses it. We currently don't
>> know what formats does a backend support, what formats can a frontend
>> produce, and even if we would know that, just because a frontend can
>> produce a format that the backend doesn't understand doesn't mean that
>> it will actually do it. For example, right now with this patch series
>> applied, usb-audio can produce 7.1 audio. If we want to be strict, it
>> means we can only use it with backends that support at least 8
>> channels, even if the user only wants to use stereo audio.
>>
>>> If yes, we should call out the problematic configurations in
>>> documentation.
>>
>> I think we should rather list known working configurations, and leave
>> the others as "try at your own risk" because there's too many things
>> that can go wrong. (pulseaudio will work, alsa with dmix too. Need
>> to check coreaudio, dsound and oss. spice and wavcapture won't work.)
>
> Far from ideal, but better than nothing.
>
> Possibly naive idea: what about automatically falling back to mixeng on
> when mixeng off doesn't work? Requires detecting "doesn't work", which
> I understand just isn't there. Any other reasons why this couldn't be
> done? Way out of scope for this series, of course.
There are two cases when mixeng can help: a) when the frontend requires
a format not supported by the backend, and b) when multiple frontends
wants to use the same backend, but the backend doesn't support multiple
channels. This fallback logic would help with a) but it would be much
harder to solve b) (I'm not saying it's impossible, but it would require
us to convert the already running stream back to mixeng). However,
since the goal of this setting is to get rid of limitations of mixeng,
currently you only want to turn it off if when you want to use 5.1 or
7.1 audio, where falling back to mixeng is not an option. (The other
use case was when using pulseaudio, you wanted different pa streams per
emulated sound card, but since you can have multiple -audiodevs now,
it's no longer necessary to disable mixeng.)
The ideal solution is probably porting as much as possible to gstreamer,
but this is even more out of scope:
https://wiki.qemu.org/Internships/ProjectIdeas/AudioGStreamer
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 15/24] audio: add mixing-engine option (documentation)
2019-10-03 23:27 ` Zoltán Kővágó
@ 2019-10-07 8:21 ` Markus Armbruster
0 siblings, 0 replies; 35+ messages in thread
From: Markus Armbruster @ 2019-10-07 8:21 UTC (permalink / raw)
To: Zoltán Kővágó; +Cc: qemu-devel, Gerd Hoffmann
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-10-01 08:23, Markus Armbruster wrote:
>> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>>
>>> On 2019-09-25 11:49, Markus Armbruster wrote:
>>>> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>>>>
>>>>> On 2019-09-23 15:08, Markus Armbruster wrote:
>>>>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>>>>
>>>>>>> This will allow us to disable mixeng when we use a decent backend.
>>>>>>>
>>>>>>> Disabling mixeng have a few advantages:
>>>>>>> * we no longer convert the audio output from one format to another, when
>>>>>>> the underlying audio system would just convert it to a third format.
>>>>>>> We no longer convert, only the underlying system, when needed.
>>>>>>> * the underlying system probably has better resampling and sample format
>>>>>>> converting methods anyway...
>>>>>>> * we may support formats that the mixeng currently does not support (S24
>>>>>>> or float samples, more than two channels)
>>>>>>> * when using an audio server (like pulseaudio) different sound card
>>>>>>> outputs will show up as separate streams, even if we use only one
>>>>>>> backend
>>>>>>>
>>>>>>> Disadvantages:
>>>>>>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>>>>>>> * some backends only support a single playback stream or very picky
>>>>>>> about the audio format. In this case we can't disable mixeng.
>>>>>>>
>>>>>>> However mixeng is not removed, only made optional, so this shouldn't be
>>>>>>> a big concern.
>>>>>>>
>>>>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>>>> ---
>>>>>>>
>>>>>>> Notes:
>>>>>>> Changes from v1:
>>>>>>> * renamed mixeng to mixing-engine
>>>>>>>
>>>>>>> qapi/audio.json | 5 +++++
>>>>>>> qemu-options.hx | 6 ++++++
>>>>>>> 2 files changed, 11 insertions(+)
>>>>>>>
>>>>>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>>>>>> index 9fefdf5186..0535eff794 100644
>>>>>>> --- a/qapi/audio.json
>>>>>>> +++ b/qapi/audio.json
>>>>>>> @@ -11,6 +11,10 @@
>>>>>>> # General audio backend options that are used for both playback and
>>>>>>> # recording.
>>>>>>> #
>>>>>>> +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU. When
>>>>>>> +# set to off, fixed-settings must be also off. Not every backend
>>>>>>> +# compatible with the off setting (default on, since 4.2)
>>>>>>> +#
>>>>>>
>>>>>> Last sentence no verb.
>>>>>>
>>>>>> Which backends are compatible?
>>>>>
>>>>> Actually that's a simplification, it depends on a few things. When
>>>>> mixeng is off, qemu will try to use the same format as the emulated
>>>>> sound card, and if the backend doesn't support that format, it won't
>>>>> work (no audio). Also attaching multiple sound cards to the same
>>>>> audiodev might not work, if the backend doesn't support multiple
>>>>> playback streams. If you use pulseaudio, it'll work without problems,
>>>>> if you use alsa, it depends on your device. If you use a hw: device
>>>>> directly, you'll likely only be able to use one emulated sound card
>>>>> with a few selected audio formats. If you use dmix: (and plug), alsa
>>>>> will handle the conversion and mixing, so it will work no matter what
>>>>> format the emulated sound card uses. With OSS the situation is
>>>>> probably similar, it depends on the kernel/hw what works and what not.
>>>>> wav and spice certainly doesn't support multiple streams. I'm not
>>>>> completely sure about the other backends right now, but I think dsound
>>>>> and coreaudio can handle the necessary sample format conversions and
>>>>> mixing.
>>>>>
>>>>>> What happens when you try the off setting with incompatible backends?
>>>>> See above.
>>>>
>>>> What happens *exactly*?
>>>>
>>>> I'm asking because I'm concerned about the user experience. When a user
>>>> asks for a combination of things QEMU can't provide, such as mixeng off
>>>> with an incompatible backend, QEMU should fail with a suitable error
>>>> message. Does it?
>>>
>>> Error handling is not the best in the audio subsystem, if something
>>> fails it generally just prints a warning to the console and continues,
>>> and something will happen... For example, this is what happens when I
>>> try to open one hw device twice. I ran qemu with:
>>>
>>> -audiodev
>>> alsa,id=foo,in.dev=hw:1,,0,out.mixing-engine=off,out.dev=hw:1,,0
>>> -device piix4-usb-uhci -device usb-audio,audiodev=foo -device
>>> AC97,audiodev=foo
>>>
>>> When the guest tried to initialize the AC97 card, I got an error:
>>>
>>> alsa: Could not initialize DAC
>>> alsa: Failed to open `hw:1,0':
>>> alsa: Reason: Device or resource busy
>>>
>>> And it just continued. And the sound worked, but with wrong sample
>>> rate (AC97 wants 44100 Hz, but USB audio previously opened the alsa
>>> device with 48000 Hz). I'll fix this bug in the next revision,
>>> audio_pcm_hw_add_* shouldn't fall back to other HWs without mixeng.
>>> But even with that, the result will be that one emulated sound card
>>> will work and the other won't.
>>>
>>> It's not ideal, but fixing it would require a lot of effort. Right
>>> now, if you specify an invalid audiodev for alsa (even with mixeng),
>>> it'll just print an error to the console and continue without audio.
>>
>> Should we document this general error handling deficiency somehow?
>
> It could be useful to mention it, especially for people who write
> script and expect qemu to fail if they specify an invalid parameter.
> I'm not sure how in-depth this explanation should be however.
Common issue when writing documentation.
One way to tackle it: start unspecific, then add specifics until it
feels useful.
The other direction can also work: write up everything, then cut until
it feels useful. Makes sense mostly when you need to write up
everything anyway, say for a developer audience, or even just for
yourself. Nothing promotes clear thinking like writing does[*].
>>>> Sometimes rejecting non-working configurations is impractical. Is it
>>>> here?
>>>
>>> I think it is. It depends on the backend, its settings, the frontend
>>> (emulated sound card), and how the guest uses it. We currently don't
>>> know what formats does a backend support, what formats can a frontend
>>> produce, and even if we would know that, just because a frontend can
>>> produce a format that the backend doesn't understand doesn't mean that
>>> it will actually do it. For example, right now with this patch series
>>> applied, usb-audio can produce 7.1 audio. If we want to be strict, it
>>> means we can only use it with backends that support at least 8
>>> channels, even if the user only wants to use stereo audio.
>>>
>>>> If yes, we should call out the problematic configurations in
>>>> documentation.
>>>
>>> I think we should rather list known working configurations, and leave
>>> the others as "try at your own risk" because there's too many things
>>> that can go wrong. (pulseaudio will work, alsa with dmix too. Need
>>> to check coreaudio, dsound and oss. spice and wavcapture won't work.)
>>
>> Far from ideal, but better than nothing.
>>
>> Possibly naive idea: what about automatically falling back to mixeng on
>> when mixeng off doesn't work? Requires detecting "doesn't work", which
>> I understand just isn't there. Any other reasons why this couldn't be
>> done? Way out of scope for this series, of course.
>
> There are two cases when mixeng can help: a) when the frontend
> requires a format not supported by the backend, and b) when multiple
> frontends wants to use the same backend, but the backend doesn't
> support multiple channels. This fallback logic would help with a) but
> it would be much harder to solve b) (I'm not saying it's impossible,
> but it would require us to convert the already running stream back to
> mixeng). However, since the goal of this setting is to get rid of
> limitations of mixeng, currently you only want to turn it off if when
> you want to use 5.1 or 7.1 audio, where falling back to mixeng is not
> an option. (The other use case was when using pulseaudio, you wanted
> different pa streams per emulated sound card, but since you can have
> multiple -audiodevs now, it's no longer necessary to disable mixeng.)
"Currently you only want to turn it off if when you want to use 5.1 or
7.1 audio" is useful advice. I figure it should be worked into
documentation if it's not already there.
Suggest to record the remainder of your answer for posterity by working
it into the commit message, along with this thought:
> The ideal solution is probably porting as much as possible to
> gstreamer, but this is even more out of scope:
> https://wiki.qemu.org/Internships/ProjectIdeas/AudioGStreamer
Thanks!
[*] "If you're thinking without writing, you only think you're
thinking." Leslie Lamport
^ permalink raw reply [flat|nested] 35+ messages in thread
* [PATCH v4 16/24] audio: make mixeng optional
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (13 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 15/24] audio: add mixing-engine option (documentation) Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 18/24] audio: support more than two channels in volume setting Kővágó, Zoltán
` (5 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Implementation of the previously added mixing-engine option.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 70 ++++++++++++++++++++++++++++++++++++++----
audio/audio_template.h | 20 ++++++++----
2 files changed, 78 insertions(+), 12 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 7128ee98dc..d616a4af98 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -838,32 +838,46 @@ static void audio_timer (void *opaque)
*/
size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
+ HWVoiceOut *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_write(sw, buf, size);
+ if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
+ return audio_pcm_sw_write(sw, buf, size);
+ } else {
+ return hw->pcm_ops->write(hw, buf, size);
+ }
}
size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
+ HWVoiceIn *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_read(sw, buf, size);
+ if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
+ return audio_pcm_sw_read(sw, buf, size);
+ } else {
+ return hw->pcm_ops->read(hw, buf, size);
+ }
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
@@ -1090,6 +1104,26 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
+ if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ sw = hw->sw_head.lh_first;
+
+ if (hw->pending_disable) {
+ hw->enabled = 0;
+ hw->pending_disable = 0;
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, false);
+ }
+ }
+
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
size_t played, live, prev_rpos, free;
int nb_live, cleanup_required;
@@ -1227,6 +1261,17 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
+ if (!audio_get_pdo_in(s->dev)->mixing_engine) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ SWVoiceIn *sw = hw->sw_head.lh_first;
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
size_t captured = 0, min;
@@ -1751,6 +1796,11 @@ CaptureVoiceOut *AUD_add_capture(
s = audio_init(NULL, NULL);
}
+ if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+ dolog("Can't capture with mixeng disabled\n");
+ return NULL;
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1905,9 +1955,13 @@ void audio_create_pdos(Audiodev *dev)
static void audio_validate_per_direction_opts(
AudiodevPerDirectionOptions *pdo, Error **errp)
{
+ if (!pdo->has_mixing_engine) {
+ pdo->has_mixing_engine = true;
+ pdo->mixing_engine = true;
+ }
if (!pdo->has_fixed_settings) {
pdo->has_fixed_settings = true;
- pdo->fixed_settings = true;
+ pdo->fixed_settings = pdo->mixing_engine;
}
if (!pdo->fixed_settings &&
(pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
@@ -1915,6 +1969,10 @@ static void audio_validate_per_direction_opts(
"You can't use frequency, channels or format with fixed-settings=off");
return;
}
+ if (!pdo->mixing_engine && pdo->fixed_settings) {
+ error_setg(errp, "You can't use fixed-settings without mixeng");
+ return;
+ }
if (!pdo->has_frequency) {
pdo->has_frequency = true;
@@ -1926,7 +1984,7 @@ static void audio_validate_per_direction_opts(
}
if (!pdo->has_voices) {
pdo->has_voices = true;
- pdo->voices = 1;
+ pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
}
if (!pdo->has_format) {
pdo->has_format = true;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 235d1acbbe..b6c5466cff 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,13 +78,17 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
- size_t samples = hw->samples;
- if (audio_bug(__func__, samples == 0)) {
- dolog("Attempted to allocate empty buffer\n");
- }
+ if (glue(audio_get_pdo_, TYPE)(hw->s->dev)->mixing_engine) {
+ size_t samples = hw->samples;
+ if (audio_bug(__func__, samples == 0)) {
+ dolog("Attempted to allocate empty buffer\n");
+ }
- HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
- HWBUF->size = samples;
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+ HWBUF->size = samples;
+ } else {
+ HWBUF = NULL;
+ }
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -103,6 +107,10 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
+ if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
+ return 0;
+ }
+
samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 18/24] audio: support more than two channels in volume setting
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (14 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 16/24] audio: make mixeng optional Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 19/24] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
` (4 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 30 ++++++++++++++++++++++--------
audio/audio.h | 10 ++++++++++
audio/audio_int.h | 4 ++--
audio/paaudio.c | 20 ++++++++++++--------
audio/spiceaudio.c | 14 ++++++++------
5 files changed, 54 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index d616a4af98..f1c145dfcd 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1891,31 +1891,45 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_out(sw, &vol);
+}
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
+ 255;
if (hw->pcm_ops->volume_out) {
- hw->pcm_ops->volume_out(hw, &sw->vol);
+ hw->pcm_ops->volume_out(hw, vol);
}
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_in(sw, &vol);
+}
+
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
+ 255;
if (hw->pcm_ops->volume_in) {
- hw->pcm_ops->volume_in(hw, &sw->vol);
+ hw->pcm_ops->volume_in(hw, vol);
}
}
}
diff --git a/audio/audio.h b/audio/audio.h
index c74abb8c47..0db3c7dd5e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -124,6 +124,16 @@ uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol);
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol);
+#define AUDIO_MAX_CHANNELS 16
+typedef struct Volume {
+ bool mute;
+ int channels;
+ uint8_t vol[AUDIO_MAX_CHANNELS];
+} Volume;
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol);
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol);
+
SWVoiceIn *AUD_open_in (
QEMUSoundCard *card,
SWVoiceIn *sw,
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 22a703c13e..9176db249b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -166,7 +166,7 @@ struct audio_pcm_ops {
*/
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
void (*enable_out)(HWVoiceOut *hw, bool enable);
- void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
+ void (*volume_out)(HWVoiceOut *hw, Volume *vol);
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
@@ -174,7 +174,7 @@ struct audio_pcm_ops {
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
void (*enable_in)(HWVoiceIn *hw, bool enable);
- void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
+ void (*volume_in)(HWVoiceIn *hw, Volume *vol);
};
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6ccdf31415..d195b1caa8 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -531,20 +531,22 @@ static void qpa_fini_in (HWVoiceIn *hw)
}
}
-static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void qpa_volume_out(HWVoiceOut *hw, Volume *vol)
{
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
@@ -571,20 +573,22 @@ static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
pa_threaded_mainloop_unlock(c->mainloop);
}
-static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void qpa_volume_in(HWVoiceIn *hw, Volume *vol)
{
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 9860f9c5e1..6ed7f7a79e 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -179,13 +179,14 @@ static void line_out_enable(HWVoiceOut *hw, bool enable)
}
#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
-static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void line_out_volume(HWVoiceOut *hw, Volume *vol)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_playback_set_volume(&out->sin, 2, svol);
spice_server_playback_set_mute(&out->sin, vol->mute);
}
@@ -262,13 +263,14 @@ static void line_in_enable(HWVoiceIn *hw, bool enable)
}
#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
-static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void line_in_volume(HWVoiceIn *hw, Volume *vol)
{
SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_record_set_volume(&in->sin, 2, svol);
spice_server_record_set_mute(&in->sin, vol->mute);
}
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 19/24] audio: replace shift in audio_pcm_info with bytes_per_frame
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (15 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 18/24] audio: support more than two channels in volume setting Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 20/24] audio: basic support for multichannel audio Kővágó, Zoltán
` (3 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples). But if we want to add support for surround sound,
this no longer holds true.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 11 +++---
audio/audio.c | 74 ++++++++++++++++++++---------------------
audio/audio_int.h | 3 +-
audio/coreaudio.c | 4 +--
audio/dsound_template.h | 10 +++---
audio/dsoundaudio.c | 4 +--
audio/noaudio.c | 2 +-
audio/ossaudio.c | 14 ++++----
audio/spiceaudio.c | 3 +-
audio/wavaudio.c | 6 ++--
10 files changed, 66 insertions(+), 65 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index cfe42284a6..eddf013a53 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -602,7 +602,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
size_t pos = 0;
- size_t len_frames = len >> hw->info.shift;
+ size_t len_frames = len / hw->info.bytes_per_frame;
while (len_frames) {
char *src = advance(buf, pos);
@@ -648,7 +648,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
}
}
- pos += written << hw->info.shift;
+ pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
@@ -802,7 +802,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
- nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
+ nread = snd_pcm_readi(
+ alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
@@ -828,8 +829,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
}
}
- pos += nread << hw->info.shift;
- len -= nread << hw->info.shift;
+ pos += nread * hw->info.bytes_per_frame;
+ len -= nread * hw->info.bytes_per_frame;
}
return pos;
diff --git a/audio/audio.c b/audio/audio.c
index f1c145dfcd..c00f4deddd 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
- int bits = 8, sign = 0, shift = 0;
+ int bits = 8, sign = 0, mul;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
case AUDIO_FORMAT_U8:
+ mul = 1;
break;
case AUDIO_FORMAT_S16:
@@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
- shift = 1;
+ mul = 2;
break;
case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
- shift = 2;
+ mul = 4;
break;
default:
@@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
- info->shift = (as->nchannels == 2) + shift;
- info->align = (1 << info->shift) - 1;
- info->bytes_per_second = info->freq << info->shift;
+ info->bytes_per_frame = as->nchannels * mul;
+ info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
@@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
}
if (info->sign) {
- memset (buf, 0x00, len << info->shift);
+ memset(buf, 0x00, len * info->bytes_per_frame);
}
else {
switch (info->bits) {
case 8:
- memset (buf, 0x80, len << info->shift);
+ memset(buf, 0x80, len * info->bytes_per_frame);
break;
case 16:
{
int i;
uint16_t *p = buf;
- int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
int i;
uint32_t *p = buf;
- int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
- uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+ uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
@@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
return 0;
}
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
@@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
/*
@@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
- size >> sw->info.shift,
+ size / sw->info.bytes_per_frame,
ret,
sw->total_hw_samples_mixed
);
#endif
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@@ -882,7 +880,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->mix_buf->size << sw->hw->info.shift;
+ return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+ live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
);
- return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static size_t audio_get_free(SWVoiceOut *sw)
@@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+ live, dead, (((int64_t) dead << 32) / sw->ratio) *
+ sw->info.bytes_per_frame);
#endif
- return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1047,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
while (n) {
size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
- size_t bytes = to_write << hw->info.shift;
+ size_t bytes = to_write * hw->info.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
@@ -1082,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
return clipped + live;
}
- decr = MIN(size >> hw->info.shift, live);
+ decr = MIN(size / hw->info.bytes_per_frame, live);
audio_pcm_hw_clip_out(hw, buf, decr);
- proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
- hw->info.shift;
+ proc = hw->pcm_ops->put_buffer_out(hw, buf,
+ decr * hw->info.bytes_per_frame) /
+ hw->info.bytes_per_frame;
live -= proc;
clipped += proc;
@@ -1234,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
while (samples) {
size_t proc;
- size_t size = samples << hw->info.shift;
+ size_t size = samples * hw->info.bytes_per_frame;
void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
- assert((size & hw->info.align) == 0);
+ assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
hw->pcm_ops->put_buffer_in(hw, buf, size);
break;
}
- proc = MIN(size >> hw->info.shift,
+ proc = MIN(size / hw->info.bytes_per_frame,
conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1251,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
samples -= proc;
conv += proc;
- hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+ hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
@@ -1325,7 +1325,7 @@ static void audio_run_capture (AudioState *s)
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
- to_capture << hw->info.shift);
+ to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
@@ -1378,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->conv_buf->size << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1414,7 +1414,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->mix_buf->size << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1833,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
- cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
+ cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
@@ -2153,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
- samples = (bytes - rate->bytes_sent) >> info->shift;
+ samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
if (samples < 0 || samples > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
audio_rate_start(rate);
samples = 0;
}
- ret = MIN(samples << info->shift, bytes_avail);
+ ret = MIN(samples * info->bytes_per_frame, bytes_avail);
rate->bytes_sent += ret;
return ret;
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9176db249b..5ba2078346 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -43,8 +43,7 @@ struct audio_pcm_info {
int sign;
int freq;
int nchannels;
- int align;
- int shift;
+ int bytes_per_frame;
int bytes_per_second;
int swap_endianness;
};
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 1427c9f622..66f0f459cf 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
- pending_frames = hw->pending_emul >> hw->info.shift;
+ pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
/* if there are not enough samples, set signal and return */
if (pending_frames < frameCount) {
@@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc(
return 0;
}
- len = frameCount << hw->info.shift;
+ len = frameCount * hw->info.bytes_per_frame;
while (len) {
size_t write_len;
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 9f10b688df..7a15f91ce5 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
- if ((p1p && *p1p && (*blen1p & info->align)) ||
- (p2p && *p2p && (*blen2p & info->align))) {
+ if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
+ (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
dolog("DirectSound returned misaligned buffer %ld %ld\n",
*blen1p, *blen2p);
glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
- if (bc.dwBufferBytes & hw->info.align) {
+ if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
dolog (
"GetCaps returned misaligned buffer size %ld, alignment %d\n",
- bc.dwBufferBytes, hw->info.align + 1
+ bc.dwBufferBytes, hw->info.bytes_per_frame
);
}
hw->size_emul = bc.dwBufferBytes;
- hw->samples = bc.dwBufferBytes >> hw->info.shift;
+ hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
ds->s = s;
#ifdef DEBUG_DSOUND
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index d4a4757445..c265c0094b 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
return;
}
- len1 = blen1 >> hw->info.shift;
- len2 = blen2 >> hw->info.shift;
+ len1 = blen1 / hw->info.bytes_per_frame;
+ len2 = blen2 / hw->info.bytes_per_frame;
#ifdef DEBUG_DSOUND
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
diff --git a/audio/noaudio.c b/audio/noaudio.c
index ec8a287f36..ff99b253ff 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
NoVoiceIn *no = (NoVoiceIn *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
- audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+ audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
return bytes;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 0c4451e972..c43faeeea4 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = hw->samples << hw->info.shift;
+ hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap(
NULL,
hw->size_emul,
@@ -644,12 +644,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->fd = fd;
oss->dev = dev;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ed7f7a79e..b6b5da4812 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
if (out->frame) {
*size = audio_rate_get_bytes(
- &hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift);
+ &hw->info, &out->rate,
+ (out->fsize - out->fpos) * hw->info.bytes_per_frame);
} else {
audio_rate_start(&out->rate);
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 47efdc1b1e..e46d834bd3 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
- assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+ assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
bytes, strerror(errno));
}
- wav->total_samples += bytes >> hw->info.shift;
+ wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
uint8_t rlen[4];
uint8_t dlen[4];
- uint32_t datalen = wav->total_samples << hw->info.shift;
+ uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 20/24] audio: basic support for multichannel audio
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (16 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 19/24] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 21/24] paaudio: channel-map option Kővágó, Zoltán
` (2 subsequent siblings)
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Which currently only means removing some checks. Old code won't require
more than two channels, but new code will need it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 7 -------
audio/audio.c | 2 +-
2 files changed, 1 insertion(+), 8 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index eddf013a53..f37ce1ce85 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -493,13 +493,6 @@ static int alsa_open(bool in, struct alsa_params_req *req,
goto err;
}
- if (nchannels != 1 && nchannels != 2) {
- alsa_logerr2 (err, typ,
- "Can not handle obtained number of channels %d\n",
- nchannels);
- goto err;
- }
-
if (apdo->buffer_length) {
int dir = 0;
unsigned int btime = apdo->buffer_length;
diff --git a/audio/audio.c b/audio/audio.c
index c00f4deddd..7fc3aa9d16 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -242,7 +242,7 @@ static int audio_validate_settings (struct audsettings *as)
{
int invalid;
- invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid = as->nchannels < 1;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 21/24] paaudio: channel-map option
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (17 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 20/24] audio: basic support for multichannel audio Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-23 13:12 ` Markus Armbruster
2019-09-19 21:24 ` [PATCH v4 22/24] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 24/24] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
20 siblings, 1 reply; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
Add an option to change the channel map used by pulseaudio. If not
specified, falls back to an OSS compatible channel map.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 18 ++++++++++++++----
qapi/audio.json | 7 +++++--
qemu-options.hx | 9 +++++++++
3 files changed, 28 insertions(+), 6 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index d195b1caa8..20402b0718 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -338,17 +338,27 @@ static pa_stream *qpa_simple_new (
pa_stream_direction_t dir,
const char *dev,
const pa_sample_spec *ss,
- const pa_channel_map *map,
+ const char *map,
const pa_buffer_attr *attr,
int *rerror)
{
int r;
pa_stream *stream;
pa_stream_flags_t flags;
+ pa_channel_map pa_map;
pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new(c->context, name, ss, map);
+ if (map && !pa_channel_map_parse(&pa_map, map)) {
+ dolog("Invalid channel map specified: '%s'\n", map);
+ map = NULL;
+ }
+ if (!map) {
+ pa_channel_map_init_extend(&pa_map, ss->channels,
+ PA_CHANNEL_MAP_OSS);
+ }
+
+ stream = pa_stream_new(c->context, name, ss, &pa_map);
if (!stream) {
goto fail;
}
@@ -421,7 +431,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
&ba, /* buffering attributes */
&error
);
@@ -470,7 +480,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
&ba, /* buffering attributes */
&error
);
diff --git a/qapi/audio.json b/qapi/audio.json
index 0535eff794..07003808cb 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -214,13 +214,16 @@
# @latency: latency you want PulseAudio to achieve in microseconds
# (default 15000)
#
+# @channel-map: channel map to use (default: OSS compatible map, since: 4.2)
+#
# Since: 4.0
##
{ 'struct': 'AudiodevPaPerDirectionOptions',
'base': 'AudiodevPerDirectionOptions',
'data': {
- '*name': 'str',
- '*latency': 'uint32' } }
+ '*name': 'str',
+ '*latency': 'uint32',
+ '*channel-map': 'str' } }
##
# @AudiodevPaOptions:
diff --git a/qemu-options.hx b/qemu-options.hx
index 395427422a..f3bc342f98 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -471,6 +471,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
"-audiodev pa,id=id[,prop[=value][,...]]\n"
" server= PulseAudio server address\n"
" in|out.name= source/sink device name\n"
+ " in|out.channel-map= channel map to use\n"
#endif
#ifdef CONFIG_AUDIO_SDL
"-audiodev sdl,id=id[,prop[=value][,...]]\n"
@@ -636,6 +637,14 @@ Sets the PulseAudio @var{server} to connect to.
@item in|out.name=@var{sink}
Use the specified source/sink for recording/playback.
+@item in|out.channel-map=@var{map}
+Use the specified channel map. The default is an OSS compatible
+channel map. Do not forget to escape commas inside the map:
+
+@example
+-audiodev pa,id=example,sink.channel-map=front-left,,front-right
+@end example
+
@end table
@item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-19 21:24 ` [PATCH v4 21/24] paaudio: channel-map option Kővágó, Zoltán
@ 2019-09-23 13:12 ` Markus Armbruster
2019-09-24 0:43 ` Zoltán Kővágó
0 siblings, 1 reply; 35+ messages in thread
From: Markus Armbruster @ 2019-09-23 13:12 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> Add an option to change the channel map used by pulseaudio. If not
> specified, falls back to an OSS compatible channel map.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> audio/paaudio.c | 18 ++++++++++++++----
> qapi/audio.json | 7 +++++--
> qemu-options.hx | 9 +++++++++
> 3 files changed, 28 insertions(+), 6 deletions(-)
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index d195b1caa8..20402b0718 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -338,17 +338,27 @@ static pa_stream *qpa_simple_new (
> pa_stream_direction_t dir,
> const char *dev,
> const pa_sample_spec *ss,
> - const pa_channel_map *map,
> + const char *map,
> const pa_buffer_attr *attr,
> int *rerror)
> {
> int r;
> pa_stream *stream;
> pa_stream_flags_t flags;
> + pa_channel_map pa_map;
>
> pa_threaded_mainloop_lock(c->mainloop);
>
> - stream = pa_stream_new(c->context, name, ss, map);
> + if (map && !pa_channel_map_parse(&pa_map, map)) {
> + dolog("Invalid channel map specified: '%s'\n", map);
> + map = NULL;
> + }
> + if (!map) {
> + pa_channel_map_init_extend(&pa_map, ss->channels,
> + PA_CHANNEL_MAP_OSS);
> + }
> +
> + stream = pa_stream_new(c->context, name, ss, &pa_map);
> if (!stream) {
> goto fail;
> }
> @@ -421,7 +431,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
> PA_STREAM_PLAYBACK,
> ppdo->has_name ? ppdo->name : NULL,
> &ss,
> - NULL, /* channel map */
> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
> &ba, /* buffering attributes */
> &error
> );
> @@ -470,7 +480,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> PA_STREAM_RECORD,
> ppdo->has_name ? ppdo->name : NULL,
> &ss,
> - NULL, /* channel map */
> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
> &ba, /* buffering attributes */
> &error
> );
> diff --git a/qapi/audio.json b/qapi/audio.json
> index 0535eff794..07003808cb 100644
> --- a/qapi/audio.json
> +++ b/qapi/audio.json
> @@ -214,13 +214,16 @@
> # @latency: latency you want PulseAudio to achieve in microseconds
> # (default 15000)
> #
> +# @channel-map: channel map to use (default: OSS compatible map, since: 4.2)
> +#
> # Since: 4.0
> ##
> { 'struct': 'AudiodevPaPerDirectionOptions',
> 'base': 'AudiodevPerDirectionOptions',
> 'data': {
> - '*name': 'str',
> - '*latency': 'uint32' } }
> + '*name': 'str',
> + '*latency': 'uint32',
> + '*channel-map': 'str' } }
>
> ##
> # @AudiodevPaOptions:
> diff --git a/qemu-options.hx b/qemu-options.hx
> index 395427422a..f3bc342f98 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -471,6 +471,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
> "-audiodev pa,id=id[,prop[=value][,...]]\n"
> " server= PulseAudio server address\n"
> " in|out.name= source/sink device name\n"
> + " in|out.channel-map= channel map to use\n"
> #endif
> #ifdef CONFIG_AUDIO_SDL
> "-audiodev sdl,id=id[,prop[=value][,...]]\n"
> @@ -636,6 +637,14 @@ Sets the PulseAudio @var{server} to connect to.
> @item in|out.name=@var{sink}
> Use the specified source/sink for recording/playback.
>
> +@item in|out.channel-map=@var{map}
> +Use the specified channel map. The default is an OSS compatible
> +channel map. Do not forget to escape commas inside the map:
Awkward.
> +
> +@example
> +-audiodev pa,id=example,sink.channel-map=front-left,,front-right
> +@end example
Makes me realize new AudiodevPaPerDirectionOptions member @channel-map
is a list encoded in a string. QAPI heavily frowns upon encoding stuff
in strings. Any reason why you can't (or don't want to) make it
['str']?
> +
> @end table
>
> @item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
^ permalink raw reply [flat|nested] 35+ messages in thread* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-23 13:12 ` Markus Armbruster
@ 2019-09-24 0:43 ` Zoltán Kővágó
2019-09-25 12:17 ` Markus Armbruster
0 siblings, 1 reply; 35+ messages in thread
From: Zoltán Kővágó @ 2019-09-24 0:43 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Gerd Hoffmann
On 2019-09-23 15:12, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> Add an option to change the channel map used by pulseaudio. If not
>> specified, falls back to an OSS compatible channel map.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> audio/paaudio.c | 18 ++++++++++++++----
>> qapi/audio.json | 7 +++++--
>> qemu-options.hx | 9 +++++++++
>> 3 files changed, 28 insertions(+), 6 deletions(-)
>>
>> diff --git a/audio/paaudio.c b/audio/paaudio.c
>> index d195b1caa8..20402b0718 100644
>> --- a/audio/paaudio.c
>> +++ b/audio/paaudio.c
>> @@ -338,17 +338,27 @@ static pa_stream *qpa_simple_new (
>> pa_stream_direction_t dir,
>> const char *dev,
>> const pa_sample_spec *ss,
>> - const pa_channel_map *map,
>> + const char *map,
>> const pa_buffer_attr *attr,
>> int *rerror)
>> {
>> int r;
>> pa_stream *stream;
>> pa_stream_flags_t flags;
>> + pa_channel_map pa_map;
>>
>> pa_threaded_mainloop_lock(c->mainloop);
>>
>> - stream = pa_stream_new(c->context, name, ss, map);
>> + if (map && !pa_channel_map_parse(&pa_map, map)) {
>> + dolog("Invalid channel map specified: '%s'\n", map);
>> + map = NULL;
>> + }
>> + if (!map) {
>> + pa_channel_map_init_extend(&pa_map, ss->channels,
>> + PA_CHANNEL_MAP_OSS);
>> + }
>> +
>> + stream = pa_stream_new(c->context, name, ss, &pa_map);
>> if (!stream) {
>> goto fail;
>> }
>> @@ -421,7 +431,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
>> PA_STREAM_PLAYBACK,
>> ppdo->has_name ? ppdo->name : NULL,
>> &ss,
>> - NULL, /* channel map */
>> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
>> &ba, /* buffering attributes */
>> &error
>> );
>> @@ -470,7 +480,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>> PA_STREAM_RECORD,
>> ppdo->has_name ? ppdo->name : NULL,
>> &ss,
>> - NULL, /* channel map */
>> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
>> &ba, /* buffering attributes */
>> &error
>> );
>> diff --git a/qapi/audio.json b/qapi/audio.json
>> index 0535eff794..07003808cb 100644
>> --- a/qapi/audio.json
>> +++ b/qapi/audio.json
>> @@ -214,13 +214,16 @@
>> # @latency: latency you want PulseAudio to achieve in microseconds
>> # (default 15000)
>> #
>> +# @channel-map: channel map to use (default: OSS compatible map, since: 4.2)
>> +#
>> # Since: 4.0
>> ##
>> { 'struct': 'AudiodevPaPerDirectionOptions',
>> 'base': 'AudiodevPerDirectionOptions',
>> 'data': {
>> - '*name': 'str',
>> - '*latency': 'uint32' } }
>> + '*name': 'str',
>> + '*latency': 'uint32',
>> + '*channel-map': 'str' } }
>>
>> ##
>> # @AudiodevPaOptions:
>> diff --git a/qemu-options.hx b/qemu-options.hx
>> index 395427422a..f3bc342f98 100644
>> --- a/qemu-options.hx
>> +++ b/qemu-options.hx
>> @@ -471,6 +471,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
>> "-audiodev pa,id=id[,prop[=value][,...]]\n"
>> " server= PulseAudio server address\n"
>> " in|out.name= source/sink device name\n"
>> + " in|out.channel-map= channel map to use\n"
>> #endif
>> #ifdef CONFIG_AUDIO_SDL
>> "-audiodev sdl,id=id[,prop[=value][,...]]\n"
>> @@ -636,6 +637,14 @@ Sets the PulseAudio @var{server} to connect to.
>> @item in|out.name=@var{sink}
>> Use the specified source/sink for recording/playback.
>>
>> +@item in|out.channel-map=@var{map}
>> +Use the specified channel map. The default is an OSS compatible
>> +channel map. Do not forget to escape commas inside the map:
>
> Awkward.
>
>> +
>> +@example
>> +-audiodev pa,id=example,sink.channel-map=front-left,,front-right
>> +@end example
>
> Makes me realize new AudiodevPaPerDirectionOptions member @channel-map
> is a list encoded in a string. QAPI heavily frowns upon encoding stuff
> in strings. Any reason why you can't (or don't want to) make it
> ['str']?
Hmm, I don't think it's used too frequently on structs parsed by qapi
opts visitor. What would be the command line format in that case?
Something like this?
-audiodev
pa,id=example,sink.channel-map=front-left,sink.channel-map=front-right
I think it's simply a string because while conceptually it's a string,
we don't try to interpret it, we just pass the string to
pa_channel_map_parse. Of course we could take a list and instead either
rebuild the string or reimplement half of pa_channel_map_parse by
manually calling pa_channel_position_from_string.
Oh now that I looked again at the pulseaudio docs, channel-map doesn't
have to be a list, it can be also a "well-known mapping name".
>
>> +
>> @end table
>>
>> @item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
^ permalink raw reply [flat|nested] 35+ messages in thread* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-24 0:43 ` Zoltán Kővágó
@ 2019-09-25 12:17 ` Markus Armbruster
2019-09-25 14:13 ` Gerd Hoffmann
0 siblings, 1 reply; 35+ messages in thread
From: Markus Armbruster @ 2019-09-25 12:17 UTC (permalink / raw)
To: Zoltán Kővágó; +Cc: qemu-devel, Gerd Hoffmann
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-09-23 15:12, Markus Armbruster wrote:
>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>
>>> Add an option to change the channel map used by pulseaudio. If not
>>> specified, falls back to an OSS compatible channel map.
>>>
>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>> ---
>>> audio/paaudio.c | 18 ++++++++++++++----
>>> qapi/audio.json | 7 +++++--
>>> qemu-options.hx | 9 +++++++++
>>> 3 files changed, 28 insertions(+), 6 deletions(-)
>>>
>>> diff --git a/audio/paaudio.c b/audio/paaudio.c
>>> index d195b1caa8..20402b0718 100644
>>> --- a/audio/paaudio.c
>>> +++ b/audio/paaudio.c
>>> @@ -338,17 +338,27 @@ static pa_stream *qpa_simple_new (
>>> pa_stream_direction_t dir,
>>> const char *dev,
>>> const pa_sample_spec *ss,
>>> - const pa_channel_map *map,
>>> + const char *map,
>>> const pa_buffer_attr *attr,
>>> int *rerror)
>>> {
>>> int r;
>>> pa_stream *stream;
>>> pa_stream_flags_t flags;
>>> + pa_channel_map pa_map;
>>> pa_threaded_mainloop_lock(c->mainloop);
>>> - stream = pa_stream_new(c->context, name, ss, map);
>>> + if (map && !pa_channel_map_parse(&pa_map, map)) {
>>> + dolog("Invalid channel map specified: '%s'\n", map);
>>> + map = NULL;
>>> + }
>>> + if (!map) {
>>> + pa_channel_map_init_extend(&pa_map, ss->channels,
>>> + PA_CHANNEL_MAP_OSS);
>>> + }
>>> +
>>> + stream = pa_stream_new(c->context, name, ss, &pa_map);
>>> if (!stream) {
>>> goto fail;
>>> }
>>> @@ -421,7 +431,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
>>> PA_STREAM_PLAYBACK,
>>> ppdo->has_name ? ppdo->name : NULL,
>>> &ss,
>>> - NULL, /* channel map */
>>> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
>>> &ba, /* buffering attributes */
>>> &error
>>> );
>>> @@ -470,7 +480,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>>> PA_STREAM_RECORD,
>>> ppdo->has_name ? ppdo->name : NULL,
>>> &ss,
>>> - NULL, /* channel map */
>>> + ppdo->has_channel_map ? ppdo->channel_map : NULL,
>>> &ba, /* buffering attributes */
>>> &error
>>> );
>>> diff --git a/qapi/audio.json b/qapi/audio.json
>>> index 0535eff794..07003808cb 100644
>>> --- a/qapi/audio.json
>>> +++ b/qapi/audio.json
>>> @@ -214,13 +214,16 @@
>>> # @latency: latency you want PulseAudio to achieve in microseconds
>>> # (default 15000)
>>> #
>>> +# @channel-map: channel map to use (default: OSS compatible map, since: 4.2)
>>> +#
>>> # Since: 4.0
>>> ##
>>> { 'struct': 'AudiodevPaPerDirectionOptions',
>>> 'base': 'AudiodevPerDirectionOptions',
>>> 'data': {
>>> - '*name': 'str',
>>> - '*latency': 'uint32' } }
>>> + '*name': 'str',
>>> + '*latency': 'uint32',
>>> + '*channel-map': 'str' } }
>>> ##
>>> # @AudiodevPaOptions:
>>> diff --git a/qemu-options.hx b/qemu-options.hx
>>> index 395427422a..f3bc342f98 100644
>>> --- a/qemu-options.hx
>>> +++ b/qemu-options.hx
>>> @@ -471,6 +471,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
>>> "-audiodev pa,id=id[,prop[=value][,...]]\n"
>>> " server= PulseAudio server address\n"
>>> " in|out.name= source/sink device name\n"
>>> + " in|out.channel-map= channel map to use\n"
>>> #endif
>>> #ifdef CONFIG_AUDIO_SDL
>>> "-audiodev sdl,id=id[,prop[=value][,...]]\n"
>>> @@ -636,6 +637,14 @@ Sets the PulseAudio @var{server} to connect to.
>>> @item in|out.name=@var{sink}
>>> Use the specified source/sink for recording/playback.
>>> +@item in|out.channel-map=@var{map}
>>> +Use the specified channel map. The default is an OSS compatible
>>> +channel map. Do not forget to escape commas inside the map:
>>
>> Awkward.
>>
>>> +
>>> +@example
>>> +-audiodev pa,id=example,sink.channel-map=front-left,,front-right
>>> +@end example
>>
>> Makes me realize new AudiodevPaPerDirectionOptions member @channel-map
>> is a list encoded in a string. QAPI heavily frowns upon encoding stuff
>> in strings. Any reason why you can't (or don't want to) make it
>> ['str']?
>
> Hmm, I don't think it's used too frequently on structs parsed by qapi
> opts visitor. What would be the command line format in that case?
> Something like this?
>
> -audiodev
> pa,id=example,sink.channel-map=front-left,sink.channel-map=front-right
Let's talk concepts before details. I'll get back to this below.
> I think it's simply a string because while conceptually it's a string,
> we don't try to interpret it, we just pass the string to
> pa_channel_map_parse. Of course we could take a list and instead
> either rebuild the string or reimplement half of pa_channel_map_parse
> by manually calling pa_channel_position_from_string.
Precedence: BlockdevOptionsRbd member @auth-client-required. Under the
hood, this is Ceph configuration option auth-client-required.
Semantically a list of well-known values. rados_conf_set() takes it
encoded in a string. We could have plumbed that straight through QAPI
by making @auth-client-required a 'str'. Instead, we made it
['RbdAuthMode']. rbd.c has to encode the list in a string for
rados_conf_set().
Why did we bother? Besides the dogmatic reason "do not encode stuff in
strings in QAPI", there's a pragmatic one: introspection.
@auth-client-required's type ['RbdAuthMode'] tells exactly what the
acceptable values are. If Ceph ever grows another mode, QEMU support
for it will be visible in introspection: enum RbdAuthMode will have
another value.
Baking the acceptable modes into QEMU means new modes need a QEMU update
to work. New modes that just work without a QEMU update feel unlikely.
If we're wrong, and a QEMU update is impractical, you can still work
around the limitation with a Ceph configuration file.
Which of these considerations other than "do not encode in strings"
apply to PA I can't say.
Let's turn the question around: is there any reason to break the "do not
encode in strings" rule other than "it saves a few lines of code"?
> Oh now that I looked again at the pulseaudio docs, channel-map doesn't
> have to be a list, it can be also a "well-known mapping name".
Unambiguous because the well-known mapping names are not valid channel
position list members.
Semantially, this is a disjoint union of well-known mapping names and
channel position lists.
The tools QAPI provides for disjoint unions are union and alternate
types. In this case, an alternate of 'str' and ['str'] would do.
Well-known mapping names use the 'str' branch, channel position lists
the ['str'] branch.
>>
>>> +
>>> @end table
>>> @item -audiodev
>>> sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
I promised you to get back to command line syntax.
The opts visitor uses repeated keys for lists, just like you guessed.
It supports only a subset of the QAPI types, and has weird magic for
integer lists.
-audiodev actually uses the QObject input visitor, which is more general
and somewhat less magical. Its CLI syntax for lists is even wordier,
though. I'd expect something like
-audiodev pa,id=example,sink.channel-map.0=front-left,sink.channel-map.1=front-right
Aside: CLI QAPIfication will have to find a way to accomodate the opts
visitor's syntax and quirks, or break compatibility.
^ permalink raw reply [flat|nested] 35+ messages in thread* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-25 12:17 ` Markus Armbruster
@ 2019-09-25 14:13 ` Gerd Hoffmann
2019-09-29 18:13 ` Zoltán Kővágó
0 siblings, 1 reply; 35+ messages in thread
From: Gerd Hoffmann @ 2019-09-25 14:13 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Zoltán Kővágó
Hi,
> > Oh now that I looked again at the pulseaudio docs, channel-map doesn't
> > have to be a list, it can be also a "well-known mapping name".
>
> Unambiguous because the well-known mapping names are not valid channel
> position list members.
Do we have well-known mapping names for the common use cases?
So maybe just support these?
cheers,
Gerd
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-25 14:13 ` Gerd Hoffmann
@ 2019-09-29 18:13 ` Zoltán Kővágó
2019-09-30 6:36 ` Gerd Hoffmann
0 siblings, 1 reply; 35+ messages in thread
From: Zoltán Kővágó @ 2019-09-29 18:13 UTC (permalink / raw)
To: Gerd Hoffmann, Markus Armbruster; +Cc: qemu-devel
On 2019-09-25 16:13, Gerd Hoffmann wrote:
> Hi,
>
>>> Oh now that I looked again at the pulseaudio docs, channel-map doesn't
>>> have to be a list, it can be also a "well-known mapping name".
>>
>> Unambiguous because the well-known mapping names are not valid channel
>> position list members.
>
> Do we have well-known mapping names for the common use cases?
> So maybe just support these?
>
> cheers,
> Gerd
>
It's surprisingly hard to figure out what are these "well-known names",
I could only find them in the source:
https://github.com/michaelwu/pulseaudio/blob/16e3dccfa88455ebd329de27a98a3d979a8bdc8e/src/pulse/channelmap.c#L538
It doesn't provide much over the "auto detect channel maps from the
number of channels" pulseaudio already has (the only ambiguous case is
surround-41 vs surround-50). IIRC I originally added this feature
because USB audio and pulseaudio didn't agree about the order of
channels, for example in 5.1 pa defaults to
left,right,rear-left,rear-right,front-center,lfe while USB audio expects
left,right,center,lfe,rear-left,rear-right order. You can't specify this
with the well-known names.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 35+ messages in thread
* Re: [PATCH v4 21/24] paaudio: channel-map option
2019-09-29 18:13 ` Zoltán Kővágó
@ 2019-09-30 6:36 ` Gerd Hoffmann
0 siblings, 0 replies; 35+ messages in thread
From: Gerd Hoffmann @ 2019-09-30 6:36 UTC (permalink / raw)
To: Zoltán Kővágó; +Cc: Markus Armbruster, qemu-devel
Hi,
> vs surround-50). IIRC I originally added this feature because USB audio and
> pulseaudio didn't agree about the order of channels, for example in 5.1 pa
> defaults to left,right,rear-left,rear-right,front-center,lfe while USB audio
> expects left,right,center,lfe,rear-left,rear-right order. You can't specify
> this with the well-known names.
Shouldn't backend and frontend be able to negotiate a channel map
without user invention?
Initially, with only usb-audio supporting 5.1/7.1 we can simply declare
the usb-audio channel map as the one which is used. When other devices
are added some day (hda-surround?) we might need a ctl function so the
frontend can set the channel map (similar to the way volume is handled
today).
cheers,
Gerd
^ permalink raw reply [flat|nested] 35+ messages in thread
* [PATCH v4 22/24] usb-audio: do not count on avail bytes actually available
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (18 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 21/24] paaudio: channel-map option Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
2019-09-19 21:24 ` [PATCH v4 24/24] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This assumption is no longer true when mixeng is turned off.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 28 +++++++++++++++++-----------
1 file changed, 17 insertions(+), 11 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index ae42e5a2f1..74c99b1f12 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -319,30 +319,31 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p)
{
uint32_t free = buf->size - (buf->prod - buf->cons);
- if (!free) {
+ if (free < USBAUDIO_PACKET_SIZE) {
return 0;
}
if (p->iov.size != USBAUDIO_PACKET_SIZE) {
return 0;
}
- assert(free >= USBAUDIO_PACKET_SIZE);
+
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE);
buf->prod += USBAUDIO_PACKET_SIZE;
return USBAUDIO_PACKET_SIZE;
}
-static uint8_t *streambuf_get(struct streambuf *buf)
+static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
uint32_t used = buf->prod - buf->cons;
uint8_t *data;
if (!used) {
+ *len = 0;
return NULL;
}
- assert(used >= USBAUDIO_PACKET_SIZE);
data = buf->data + (buf->cons % buf->size);
- buf->cons += USBAUDIO_PACKET_SIZE;
+ *len = MIN(buf->prod - buf->cons,
+ buf->size - (buf->cons % buf->size));
return data;
}
@@ -374,16 +375,21 @@ static void output_callback(void *opaque, int avail)
USBAudioState *s = opaque;
uint8_t *data;
- for (;;) {
- if (avail < USBAUDIO_PACKET_SIZE) {
+ while (avail) {
+ size_t written, len;
+
+ data = streambuf_get(&s->out.buf, &len);
+ if (!data) {
return;
}
- data = streambuf_get(&s->out.buf);
- if (!data) {
+
+ written = AUD_write(s->out.voice, data, len);
+ avail -= written;
+ s->out.buf.cons += written;
+
+ if (written < len) {
return;
}
- AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE);
- avail -= USBAUDIO_PACKET_SIZE;
}
}
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread* [PATCH v4 24/24] usbaudio: change playback counters to 64 bit
2019-09-19 21:24 [PATCH v4 00/24] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (19 preceding siblings ...)
2019-09-19 21:24 ` [PATCH v4 22/24] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
@ 2019-09-19 21:24 ` Kővágó, Zoltán
20 siblings, 0 replies; 35+ messages in thread
From: Kővágó, Zoltán @ 2019-09-19 21:24 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
With stereo playback, they need about 375 minutes of continuous audio
playback to overflow, which is usually not a problem (as stopping and
later resuming playback resets the counters). But with 7.1 audio, they
only need about 95 minutes to overflow.
After the overflow, the buf->prod % USBAUDIO_PACKET_SIZE(channels)
assertion no longer holds true, which will result in overflowing the
buffer. With 64 bit variables, it would take about 762000 years to
overflow.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 14 ++++++++------
1 file changed, 8 insertions(+), 6 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index e42bdfbdc1..ea604bbb8e 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -578,9 +578,9 @@ static const USBDesc desc_audio_multi = {
struct streambuf {
uint8_t *data;
- uint32_t size;
- uint32_t prod;
- uint32_t cons;
+ size_t size;
+ uint64_t prod;
+ uint64_t cons;
};
static void streambuf_init(struct streambuf *buf, uint32_t size,
@@ -601,7 +601,7 @@ static void streambuf_fini(struct streambuf *buf)
static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
{
- uint32_t free = buf->size - (buf->prod - buf->cons);
+ int64_t free = buf->size - (buf->prod - buf->cons);
if (free < USBAUDIO_PACKET_SIZE(channels)) {
return 0;
@@ -610,6 +610,8 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
return 0;
}
+ /* can happen if prod overflows */
+ assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE(channels));
buf->prod += USBAUDIO_PACKET_SIZE(channels);
@@ -618,10 +620,10 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
- uint32_t used = buf->prod - buf->cons;
+ int64_t used = buf->prod - buf->cons;
uint8_t *data;
- if (!used) {
+ if (used <= 0) {
*len = 0;
return NULL;
}
--
2.23.0
^ permalink raw reply related [flat|nested] 35+ messages in thread