* [Qemu-devel] [PULL 00/26] Audio 20190919 patches
@ 2019-09-19 8:36 Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends Gerd Hoffmann
` (26 more replies)
0 siblings, 27 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
The following changes since commit f8c3db33a5e863291182f8862ddf81618a7c6194:
target/sparc: Switch to do_transaction_failed() hook (2019-09-17 12:01:00 +0100)
are available in the Git repository at:
git://git.kraxel.org/qemu tags/audio-20190919-pull-request
for you to fetch changes up to cf0c1c2aa32db5d658c3c797ad995a6d571bad96:
audio: fix ALSA period-length typo in documentation (2019-09-19 10:32:48 +0200)
----------------------------------------------------------------
audio: make mixeng optional.
audio: add surround sound support.
audio: documentation fixes.
----------------------------------------------------------------
Kővágó, Zoltán (24):
audio: api for mixeng code free backends
alsaaudio: port to the new audio backend api
coreaudio: port to the new audio backend api
dsoundaudio: port to the new audio backend api
noaudio: port to the new audio backend api
ossaudio: port to the new audio backend api
paaudio: port to the new audio backend api
sdlaudio: port to the new audio backend api
spiceaudio: port to the new audio backend api
wavaudio: port to the new audio backend api
audio: remove remains of the old backend api
audio: unify input and output mixeng buffer management
audio: common rate control code for timer based outputs
audio: split ctl_* functions into enable_* and volume_*
audio: add mixeng option (documentation)
audio: make mixeng optional
paaudio: get/put_buffer functions
audio: support more than two channels in volume setting
audio: replace shift in audio_pcm_info with bytes_per_frame
audio: basic support for multichannel audio
paaudio: channel-map option
usb-audio: do not count on avail bytes actually available
usb-audio: support more than two channels of audio
usbaudio: change playback counters to 64 bit
Stefan Hajnoczi (2):
audio: fix buffer-length typo in documentation
audio: fix ALSA period-length typo in documentation
configure | 5 -
audio/audio.h | 10 +
audio/audio_int.h | 79 ++++--
audio/audio_pt_int.h | 22 --
audio/audio_template.h | 31 ++-
audio/dsound_template.h | 53 ++--
audio/alsaaudio.c | 378 ++++++++-------------------
audio/audio.c | 527 +++++++++++++++++++++++++++++---------
audio/audio_pt_int.c | 173 -------------
audio/coreaudio.c | 143 ++++++-----
audio/dsoundaudio.c | 383 ++++++++-------------------
audio/noaudio.c | 76 +++---
audio/ossaudio.c | 376 ++++++++++-----------------
audio/paaudio.c | 554 +++++++++++++---------------------------
audio/sdlaudio.c | 104 ++++----
audio/spiceaudio.c | 260 +++++++------------
audio/wavaudio.c | 78 ++----
hw/usb/dev-audio.c | 461 ++++++++++++++++++++++++++++-----
audio/Makefile.objs | 1 -
qapi/audio.json | 12 +-
qemu-options.hx | 23 +-
21 files changed, 1719 insertions(+), 2030 deletions(-)
delete mode 100644 audio/audio_pt_int.h
delete mode 100644 audio/audio_pt_int.c
--
2.18.1
^ permalink raw reply [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 02/26] alsaaudio: port to the new audio backend api Gerd Hoffmann
` (25 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 6ba56ad4506c283aa903e8029f5fa53520b27544.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 41 ++++++--
audio/audio_template.h | 1 +
audio/audio.c | 216 ++++++++++++++++++++++++++++++++++++++++-
3 files changed, 250 insertions(+), 8 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index a674c5374a06..8fb1ca8a8d0f 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, ...);
+ 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, struct audsettings *as, void *drv_opaque);
- void (*fini_in) (HWVoiceIn *hw);
+ 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 2562bf5f0062..ff4a173f1810 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;
}
diff --git a/audio/audio.c b/audio/audio.c
index e99fcd0694e9..e29a1e15eb30 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;
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 02/26] alsaaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 03/26] coreaudio: " Gerd Hoffmann
` (24 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3978a3642e68da4d0af61c7618fcaa4ee22b009f.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/alsaaudio.c | 308 +++++++++++++---------------------------------
1 file changed, 83 insertions(+), 225 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 591344dccd1f..19124d09d845 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)
-{
- 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;
- }
- }
-
- return avail;
-}
-
-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;
- }
- }
-
- alsa->wpos = (alsa->wpos + written) % hw->samples;
- alsa->pending -= written;
- len -= written;
- }
- }
-}
-
-static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
+static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- size_t decr;
- snd_pcm_sframes_t avail;
+ size_t pos = 0;
+ size_t len_frames = len >> hw->info.shift;
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of available playback frames\n");
- return 0;
+ 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;
+
+ case -EAGAIN:
+ return pos;
+
+ default:
+ alsa_logerr(written, "Failed to write %zu frames from %p\n",
+ len, src);
+ return pos;
+ }
+ }
+
+ 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;
+ size_t pos = 0;
- if (!dead) {
- return 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;
- }
- }
-
- decr = MIN(dead, avail);
- if (!decr) {
- return 0;
- }
-
- 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;
- }
-
- for (i = 0; i < 2; ++i) {
- void *src;
- struct st_sample *dst;
+ while (len) {
+ void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
- snd_pcm_uframes_t len;
- len = bufs[i].len;
+ nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
- src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
- dst = hw->conv_buf + bufs[i].add;
+ if (nread <= 0) {
+ switch (nread) {
+ case 0:
+ trace_alsa_read_zero(len);
+ return pos;;
- 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;
+
+ case -EAGAIN:
+ return pos;
+
+ default:
+ alsa_logerr(nread, "Failed to read %zu frames to %p\n",
+ len, dst);
+ return pos;;
}
-
- hw->conv (dst, src, nread);
-
- src = advance (src, nread << hwshift);
- dst += nread;
-
- read_samples += nread;
- len -= nread;
}
+
+ 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,
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 03/26] coreaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 02/26] alsaaudio: port to the new audio backend api Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 04/26] dsoundaudio: " Gerd Hoffmann
` (23 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 60d68c051ed180c7315f7cdd6084b58b6fc9bb6d.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/coreaudio.c | 130 ++++++++++++++++++++++++----------------------
1 file changed, 69 insertions(+), 61 deletions(-)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index d1be58b40aa8..5cde42f9826c 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);
+
+ memcpy(out, hw->buf_emul + start, write_len);
+ hw->pending_emul -= write_len;
+ len -= write_len;
+ out += write_len;
}
- rpos = (rpos + frameCount) % hw->samples;
- core->decr += frameCount;
- core->rpos = rpos;
-
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
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 04/26] dsoundaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (2 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 03/26] coreaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 05/26] noaudio: " Gerd Hoffmann
` (22 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 0019bfb5ae50b0750b839460b5dbc1b3073f02e7.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/dsound_template.h | 47 +++---
audio/dsoundaudio.c | 329 ++++++++++------------------------------
2 files changed, 103 insertions(+), 273 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 8ece870c9ef7..9f10b688df57 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 2fc118b795d0..9960247814c7 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;
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
HRESULT hr;
+ DWORD ppos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
+
+ hr = IDirectSoundBuffer_GetCurrentPosition(dsb, &ppos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get playback buffer position\n");
+ *size = 0;
+ return NULL;
+ }
+
+ req_size = audio_ring_dist(ppos, hw->pos_emul, hw->size_emul);
+ req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
+
+ 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;
+ }
+
+ *size = act_size;
+ return ret;
+}
+
+static size_t dsound_put_buffer_out(HWVoiceOut *hw, void *buf, size_t len)
+{
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;
+ int err = dsound_unlock_out(dsb, buf, NULL, len, 0);
- 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;
-
- 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
- }
- 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;
- }
-
- 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;
- }
- }
-
- if (audio_bug(__func__, len > bufsize)) {
- dolog("len=%zu bufsize=%zu old_pos=%ld ppos=%ld\n",
- len, bufsize, old_pos, ppos);
- return 0;
- }
-
- len &= ~hw->info.align;
- if (!len) {
- return 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;
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
HRESULT hr;
+ DWORD cpos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
+
+ hr = IDirectSoundCaptureBuffer_GetCurrentPosition(dscb, &cpos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get capture buffer position\n");
+ *size = 0;
+ return NULL;
+ }
+
+ 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->pos_emul, req_size, &ret, NULL,
+ &act_size, NULL, false, ds->s);
+ if (err) {
+ dolog("Failed to lock buffer\n");
+ *size = 0;
+ return NULL;
+ }
+
+ *size = act_size;
+ return ret;
+}
+
+static void dsound_put_buffer_in(HWVoiceIn *hw, void *buf, size_t len)
+{
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;
+ int err = dsound_unlock_in(dscb, buf, NULL, len, 0);
- 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;
- }
-
- if (cpos & hw->info.align) {
- ldebug ("warning: Misaligned capture position %ld(%d)\n",
- cpos, hw->info.align);
- }
- cpos >>= hwshift;
-
- len = audio_ring_dist (cpos, hw->wpos, hw->samples);
- if (!len) {
- return 0;
- }
- len = MIN (len, dead);
-
- err = dsound_lock_in (
- dscb,
- &hw->info,
- hw->wpos << hwshift,
- len << hwshift,
- &p1,
- &p2,
- &blen1,
- &blen2,
- 0,
- s
- );
if (err) {
- return 0;
+ dolog("Failed to unlock buffer!!\n");
+ return;
}
-
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
-
- if (p1 && len1) {
- hw->conv (hw->conv_buf + hw->wpos, p1, len1);
- }
-
- if (p2 && len2) {
- hw->conv (hw->conv_buf, p2, len2);
- }
-
- 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
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 05/26] noaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (3 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 04/26] dsoundaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 06/26] ossaudio: " Gerd Hoffmann
` (21 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 066dc6dd54f4382d80de4376306f585b7fb47805.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/noaudio.c | 39 +++++++++++++++------------------------
1 file changed, 15 insertions(+), 24 deletions(-)
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 0fb2629cf283..b054fd225b66 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);
+ 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;
+ 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
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 06/26] ossaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (4 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 05/26] noaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 07/26] paaudio: " Gerd Hoffmann
` (20 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 44f4e888975c1d94f5d89e945df9782c0f541582.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/ossaudio.c | 288 +++++++++++++++++------------------------------
1 file changed, 104 insertions(+), 184 deletions(-)
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 169693368886..278251270691 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);
+ err = ioctl(oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
+ if (err < 0) {
+ oss_logerr(errno, "SNDCTL_DSP_GETOPTR failed\n");
+ return 0;
+ }
+
+ return audio_ring_dist(cntinfo.ptr, oss->hw.pos_emul, oss->hw.size_emul);
+}
+
+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);
+ }
+}
+
+static size_t oss_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
if (oss->mmapped) {
- return;
+ assert(buf == hw->buf_emul + hw->pos_emul && size < hw->size_emul);
+
+ 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_write(HWVoiceOut *hw, void *buf, size_t len)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ size_t pos;
+
+ if (oss->mmapped) {
+ size_t total_len;
+ len = MIN(len, oss_get_available_bytes(oss));
+
+ 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);
+
+ hw->pos_emul = (hw->pos_emul + to_copy) % hw->pos_emul;
+ buf += to_copy;
+ len -= to_copy;
+ }
+ return total_len;
}
- while (oss->pending) {
- int samples_written;
+ pos = 0;
+ while (len) {
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);
+ void *pcm = advance(buf, pos);
- bytes_written = write (oss->fd, pcm, bytes_to_write);
+ bytes_written = write(oss->fd, pcm, len);
if (bytes_written < 0) {
if (errno != EAGAIN) {
- oss_logerr (errno, "failed to write %d bytes\n",
- bytes_to_write);
+ oss_logerr(errno, "failed to write %zu bytes\n",
+ len);
}
- break;
- }
-
- if (bytes_written & hw->info.align) {
- dolog ("misaligned write asked for %d, but got %zd\n",
- bytes_to_write, bytes_written);
- return;
+ return pos;
}
- 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) {
+ pos += bytes_written;
+ if (bytes_written < len) {
break;
}
+ len -= bytes_written;
}
-}
-
-static size_t oss_run_out(HWVoiceOut *hw, size_t live)
-{
- 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;
-
- if (oss->mmapped) {
- int bytes, pos;
-
- err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
- if (err < 0) {
- oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
- return 0;
- }
-
- 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;
- }
-
- if (abinfo.bytes > bufsize) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- abinfo.bytes = bufsize;
- }
-
- if (abinfo.bytes < 0) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- return 0;
- }
-
- decr = MIN (abinfo.bytes >> hw->info.shift, live);
- if (!decr) {
- return 0;
- }
- }
-
- 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);
- }
+ 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);
}
- else {
- g_free (oss->pcm_buf);
- }
- 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 }
- };
+ size_t pos = 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;
- }
-
- 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);
+ void *dst = advance(buf, pos);
+ nread = read(oss->fd, dst, 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);
- }
-
- 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
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 07/26] paaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (5 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 06/26] ossaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 08/26] sdlaudio: " Gerd Hoffmann
` (19 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: f8f15e33739cd18305a3719270b52d61ece68bd5.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
configure | 5 -
audio/audio_pt_int.h | 22 ---
audio/audio_pt_int.c | 173 --------------------
audio/paaudio.c | 372 ++++++-------------------------------------
audio/Makefile.objs | 1 -
5 files changed, 45 insertions(+), 528 deletions(-)
delete mode 100644 audio/audio_pt_int.h
delete mode 100644 audio/audio_pt_int.c
diff --git a/configure b/configure
index 30aad233d17c..d72ea39bb7eb 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
diff --git a/audio/audio_pt_int.h b/audio/audio_pt_int.h
deleted file mode 100644
index 4c0c15b9af94..000000000000
--- 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/audio_pt_int.c b/audio/audio_pt_int.c
deleted file mode 100644
index 9f9d44ad4a63..000000000000
--- 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/paaudio.c b/audio/paaudio.c
index bfef9acaadd0..75fce5320269 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);
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
- 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;
-
- 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, l != (size_t) -1, unlock_and_fail,
+ "pa_stream_writable_size failed\n");
- CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
-
- if (l > length) {
- l = length;
- }
-
- r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO(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/audio/Makefile.objs b/audio/Makefile.objs
index dca87f63470d..d7490a379f3e 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
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 08/26] sdlaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (6 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 07/26] paaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 09/26] spiceaudio: " Gerd Hoffmann
` (18 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3e95a57a0663f1f4464cd3515e252628791c971e.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/sdlaudio.c | 87 +++++++++++++++++++++++-------------------------
1 file changed, 42 insertions(+), 45 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 14b11f033521..f7ac8cd10188 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_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->live = live;
-
- 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,
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 09/26] spiceaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (7 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 08/26] sdlaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 10/26] wavaudio: " Gerd Hoffmann
` (17 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 71731ff2437028514284394de0f60d195d42e593.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/spiceaudio.c | 116 ++++++++++++++++-----------------------------
1 file changed, 42 insertions(+), 74 deletions(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 26873c7f22a5..ff4e4dcbb022 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -51,7 +51,7 @@ typedef struct SpiceVoiceOut {
SpiceRateCtl rate;
int active;
uint32_t *frame;
- uint32_t *fpos;
+ uint32_t fpos;
uint32_t fsize;
} SpiceVoiceOut;
@@ -60,7 +60,6 @@ typedef struct SpiceVoiceIn {
SpiceRecordInstance sin;
SpiceRateCtl rate;
int active;
- uint32_t samples[LINE_IN_SAMPLES];
} SpiceVoiceIn;
static const SpicePlaybackInterface playback_sif = {
@@ -152,44 +151,40 @@ static void line_out_fini (HWVoiceOut *hw)
spice_server_remove_interface (&out->sin.base);
}
-static size_t line_out_run (HWVoiceOut *hw, size_t live)
+static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
{
- SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- size_t rpos, decr;
- size_t samples;
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+ size_t decr;
- if (!live) {
- return 0;
+ if (!out->frame) {
+ spice_server_playback_get_buffer(&out->sin, &out->frame, &out->fsize);
+ out->fpos = 0;
}
- decr = rate_get_samples (&hw->info, &out->rate);
- decr = MIN (live, decr);
+ if (out->frame) {
+ decr = rate_get_samples(&hw->info, &out->rate);
+ decr = MIN(out->fsize - out->fpos, decr);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int len = MIN (samples, left_till_end_samples);
+ *size = decr << hw->info.shift;
+ } else {
+ rate_start(&out->rate);
+ }
+ return out->frame + out->fpos;
+}
+
+static size_t line_out_put_buffer(HWVoiceOut *hw, void *buf, size_t size)
+{
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
- if (!out->frame) {
- spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize);
- out->fpos = out->frame;
- }
- if (out->frame) {
- len = MIN (len, out->fsize);
- hw->clip (out->fpos, hw->mix_buf + rpos, len);
- out->fsize -= len;
- out->fpos += len;
- if (out->fsize == 0) {
- spice_server_playback_put_samples (&out->sin, out->frame);
- out->frame = out->fpos = NULL;
- }
- }
- rpos = (rpos + len) % hw->samples;
- samples -= len;
+ assert(buf == out->frame + out->fpos && out->fpos <= out->fsize);
+ out->fpos += size >> 2;
+
+ if (out->fpos == out->fsize) { /* buffer full */
+ spice_server_playback_put_samples(&out->sin, out->frame);
+ out->frame = NULL;
}
- hw->rpos = rpos;
- return decr;
+
+ return size;
}
static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
@@ -211,9 +206,9 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
}
out->active = 0;
if (out->frame) {
- memset (out->fpos, 0, out->fsize << 2);
+ memset(out->frame + out->fpos, 0, (out->fsize - out->fpos) << 2);
spice_server_playback_put_samples (&out->sin, out->frame);
- out->frame = out->fpos = NULL;
+ out->frame = NULL;
}
spice_server_playback_stop (&out->sin);
break;
@@ -275,49 +270,20 @@ static void line_in_fini (HWVoiceIn *hw)
spice_server_remove_interface (&in->sin.base);
}
-static size_t line_in_run(HWVoiceIn *hw)
+static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- size_t num_samples;
- int ready;
- size_t len[2];
- uint64_t delta_samp;
- const uint32_t *samples;
+ uint64_t delta_samp = rate_get_samples(&hw->info, &in->rate);
+ uint64_t to_read = MIN(len >> 2, delta_samp);
+ size_t ready = spice_server_record_get_samples(&in->sin, buf, to_read);
- if (!(num_samples = hw->samples - audio_pcm_hw_get_live_in (hw))) {
- return 0;
- }
-
- delta_samp = rate_get_samples (&hw->info, &in->rate);
- num_samples = MIN (num_samples, delta_samp);
-
- ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples);
- samples = in->samples;
+ /* XXX: do we need this? */
if (ready == 0) {
- static const uint32_t silence[LINE_IN_SAMPLES];
- samples = silence;
- ready = LINE_IN_SAMPLES;
+ memset(buf, 0, to_read << 2);
+ ready = to_read;
}
- num_samples = MIN (ready, num_samples);
-
- if (hw->wpos + num_samples > hw->samples) {
- len[0] = hw->samples - hw->wpos;
- len[1] = num_samples - len[0];
- } else {
- len[0] = num_samples;
- len[1] = 0;
- }
-
- hw->conv (hw->conv_buf + hw->wpos, samples, len[0]);
-
- if (len[1]) {
- hw->conv (hw->conv_buf, samples + len[0], len[1]);
- }
-
- hw->wpos = (hw->wpos + num_samples) % hw->samples;
-
- return num_samples;
+ return ready << 2;
}
static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
@@ -366,12 +332,14 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
- .run_out = line_out_run,
+ .write = audio_generic_write,
+ .get_buffer_out = line_out_get_buffer,
+ .put_buffer_out = line_out_put_buffer,
.ctl_out = line_out_ctl,
.init_in = line_in_init,
.fini_in = line_in_fini,
- .run_in = line_in_run,
+ .read = line_in_read,
.ctl_in = line_in_ctl,
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 10/26] wavaudio: port to the new audio backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (8 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 09/26] spiceaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 11/26] audio: remove remains of the old " Gerd Hoffmann
` (16 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: fc447ed1336d60025485bbe6f3a4da52b1359077.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/wavaudio.c | 54 ++++++++----------------------------------------
1 file changed, 9 insertions(+), 45 deletions(-)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index b6eeeb4e26ef..7816097db8f9 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 %zu bytes failed\nReaons: %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,
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 11/26] audio: remove remains of the old backend api
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (9 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 10/26] wavaudio: " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 12/26] audio: unify input and output mixeng buffer management Gerd Hoffmann
` (15 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3c160307074a29e5826a89994ab7cfdee7b8ccf8.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 7 -------
audio/audio.c | 42 ++++++------------------------------------
2 files changed, 6 insertions(+), 43 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 8fb1ca8a8d0f..c76d7c39e84c 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);
diff --git a/audio/audio.c b/audio/audio.c
index e29a1e15eb30..435bcf20c139 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);
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 12/26] audio: unify input and output mixeng buffer management
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (10 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 11/26] audio: remove remains of the old " Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 13/26] audio: common rate control code for timer based outputs Gerd Hoffmann
` (14 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: c79a75db0d6abf1d86332c6ed354e6254ed2305f.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 12 ++--
audio/audio_template.h | 19 +++----
audio/audio.c | 122 +++++++++++++++++++++--------------------
audio/ossaudio.c | 3 +-
4 files changed, 80 insertions(+), 76 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index c76d7c39e84c..20021df9e8be 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 ff4a173f1810..87c6d2d27102 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/audio.c b/audio/audio.c
index 435bcf20c139..ba07fb77dd4f 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/ossaudio.c b/audio/ossaudio.c
index 278251270691..76c082d5e2a5 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 (
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 13/26] audio: common rate control code for timer based outputs
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (11 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 12/26] audio: unify input and output mixeng buffer management Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 14/26] audio: split ctl_* functions into enable_* and volume_* Gerd Hoffmann
` (13 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 431af275a903b435f44fffa94896dbfd5bd4ecce.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 9 ++++++++
audio/audio.c | 30 +++++++++++++++++++++++++++
audio/noaudio.c | 49 ++++++++++++++++++++------------------------
audio/spiceaudio.c | 51 ++++++++--------------------------------------
audio/wavaudio.c | 21 +++++++++----------
5 files changed, 79 insertions(+), 81 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 20021df9e8be..778615ccafa4 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/audio.c b/audio/audio.c
index ba07fb77dd4f..fab1e3571838 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/noaudio.c b/audio/noaudio.c
index b054fd225b66..9f1cc67df942 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 ff4e4dcbb022..4ce4f94c6dca 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 7816097db8f9..cb2783e82ac6 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 %zu bytes failed\nReaons: %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;
}
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 14/26] audio: split ctl_* functions into enable_* and volume_*
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (12 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 13/26] audio: common rate control code for timer based outputs Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation) Gerd Hoffmann
` (12 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 7f40c7fff80879916356103cc05d14e5626cb880.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 15 ++---
audio/audio_template.h | 1 -
audio/alsaaudio.c | 62 ++++++++------------
audio/audio.c | 45 +++++++++------
audio/coreaudio.c | 13 ++---
audio/dsoundaudio.c | 50 +++++++---------
audio/noaudio.c | 14 ++---
audio/ossaudio.c | 79 ++++++++++---------------
audio/paaudio.c | 127 ++++++++++++++++-------------------------
audio/sdlaudio.c | 17 +-----
audio/spiceaudio.c | 102 ++++++++++++++-------------------
audio/wavaudio.c | 7 +--
12 files changed, 217 insertions(+), 315 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 778615ccafa4..22a703c13e1c 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 87c6d2d27102..235d1acbbebb 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/alsaaudio.c b/audio/alsaaudio.c
index 19124d09d845..cfe42284a6aa 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 fab1e3571838..7128ee98dc97 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/coreaudio.c b/audio/coreaudio.c
index 5cde42f9826c..1427c9f622d9 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 9960247814c7..d4a4757445b0 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 9f1cc67df942..ec8a287f3689 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 76c082d5e2a5..0c4451e972de 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 75fce5320269..ed31f863f7fe 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;
+ 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;
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceOut *);
- va_end (ap);
+ pa_threaded_mainloop_lock(c->mainloop);
- 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);
- }
+ 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);
+ }
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ 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);
}
- return 0;
+
+ 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;
+ 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;
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceIn *);
- va_end (ap);
+ pa_threaded_mainloop_lock(c->mainloop);
- 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);
- }
+ 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);
+ }
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ 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);
}
- return 0;
+
+ 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 f7ac8cd10188..5c6bcfcb3e9d 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 4ce4f94c6dca..9860f9c5e16c 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];
+static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+ uint16_t svol[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);
+ 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
- break;
- }
- }
-
- return 0;
-}
/* 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];
+static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw);
+ uint16_t svol[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);
+ 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
- break;
- }
- }
-
- return 0;
-}
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 cb2783e82ac6..2cc8b137bad2 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 = {
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation)
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (13 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 14/26] audio: split ctl_* functions into enable_* and volume_* Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 15:27 ` Eric Blake
2019-09-19 8:36 ` [Qemu-devel] [PULL 16/26] audio: make mixeng optional Gerd Hoffmann
` (11 subsequent siblings)
26 siblings, 1 reply; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: b1a5c621da034c8b8485c9b2097080b9760f274a.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
qapi/audio.json | 5 +++++
qemu-options.hx | 6 ++++++
2 files changed, 11 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index 9fefdf5186dd..0535eff7948f 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 bbfd936d29ec..734a3b5d6569 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-engineeng= 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
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 16/26] audio: make mixeng optional
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (14 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation) Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 17/26] paaudio: get/put_buffer functions Gerd Hoffmann
` (10 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Implementation of the previously added mixing-engine option.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 0adb9ca41b5abad2e048e9e36137446e86d5905c.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_template.h | 20 ++++++++----
audio/audio.c | 70 ++++++++++++++++++++++++++++++++++++++----
2 files changed, 78 insertions(+), 12 deletions(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 235d1acbbebb..b6c5466cff9e 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));
diff --git a/audio/audio.c b/audio/audio.c
index 7128ee98dc97..d616a4af98bd 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;
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 17/26] paaudio: get/put_buffer functions
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (15 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 16/26] audio: make mixeng optional Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 18/26] audio: support more than two channels in volume setting Gerd Hoffmann
` (9 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
This lets us avoid some buffer copying when using mixeng.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 4554ff54ad04f706ad0e9af87fe07650fd6d9ac1.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/paaudio.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 83 insertions(+)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index ed31f863f7fe..6ccdf3141598 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -98,6 +98,59 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
} \
} while (0)
+static void *qpa_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
+ PAConnection *c = p->g->conn;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
+
+ 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");
+ }
+
+ *size = MIN(p->read_length, *size);
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return (void *) p->read_data;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+ *size = 0;
+ return NULL;
+}
+
+static void qpa_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
+ PAConnection *c = p->g->conn;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock,
+ "pa_threaded_mainloop_lock failed\n");
+
+ assert(buf == p->read_data && size <= p->read_length);
+
+ p->read_data += size;
+ p->read_length -= size;
+
+ if (size && !p->read_length) {
+ r = pa_stream_drop(p->stream);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock, "pa_stream_drop failed\n");
+ }
+
+unlock:
+ pa_threaded_mainloop_unlock(c->mainloop);
+}
+
static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length)
{
PAVoiceIn *p = (PAVoiceIn *) hw;
@@ -136,6 +189,32 @@ unlock_and_fail:
return 0;
}
+static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ PAVoiceOut *p = (PAVoiceOut *) hw;
+ PAConnection *c = p->g->conn;
+ void *ret;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
+
+ *size = -1;
+ r = pa_stream_begin_write(p->stream, &ret, size);
+ CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail,
+ "pa_stream_begin_write failed\n");
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return ret;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+ *size = 0;
+ return NULL;
+}
+
static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
{
PAVoiceOut *p = (PAVoiceOut *) hw;
@@ -698,11 +777,15 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.write = qpa_write,
+ .get_buffer_out = qpa_get_buffer_out,
+ .put_buffer_out = qpa_write, /* pa handles it */
.volume_out = qpa_volume_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.read = qpa_read,
+ .get_buffer_in = qpa_get_buffer_in,
+ .put_buffer_in = qpa_put_buffer_in,
.volume_in = qpa_volume_in
};
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 18/26] audio: support more than two channels in volume setting
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (16 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 17/26] paaudio: get/put_buffer functions Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 19/26] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
` (8 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 4c938a46b760792319dbc2f61a442a41a36718d3.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio.h | 10 ++++++++++
audio/audio_int.h | 4 ++--
audio/audio.c | 30 ++++++++++++++++++++++--------
audio/paaudio.c | 20 ++++++++++++--------
audio/spiceaudio.c | 14 ++++++++------
5 files changed, 54 insertions(+), 24 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index c74abb8c4718..0db3c7dd5e06 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 22a703c13e1c..9176db249b23 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/audio.c b/audio/audio.c
index d616a4af98bd..f1c145dfcdeb 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/paaudio.c b/audio/paaudio.c
index 6ccdf3141598..d195b1caa8d8 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 9860f9c5e16c..6ed7f7a79e39 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);
}
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 19/26] audio: replace shift in audio_pcm_info with bytes_per_frame
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (17 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 18/26] audio: support more than two channels in volume setting Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 20/26] audio: basic support for multichannel audio Gerd Hoffmann
` (7 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 43893f0b3fa0631cbb9971fc2bc6924deddcc729.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_int.h | 3 +-
audio/dsound_template.h | 10 +++---
audio/alsaaudio.c | 11 +++---
audio/audio.c | 74 ++++++++++++++++++++---------------------
audio/coreaudio.c | 4 +--
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/audio_int.h b/audio/audio_int.h
index 9176db249b23..5ba20783463a 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/dsound_template.h b/audio/dsound_template.h
index 9f10b688df57..7a15f91ce563 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/alsaaudio.c b/audio/alsaaudio.c
index cfe42284a6aa..eddf013a537c 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 f1c145dfcdeb..c00f4deddd3d 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/coreaudio.c b/audio/coreaudio.c
index 1427c9f622d9..66f0f459cf09 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/dsoundaudio.c b/audio/dsoundaudio.c
index d4a4757445b0..c265c0094b9f 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 ec8a287f3689..ff99b253ff0b 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 0c4451e972de..c43faeeea4aa 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 6ed7f7a79e39..b6b5da4812f2 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 2cc8b137bad2..b5128c8a173d 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 %zu bytes failed\nReaons: %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) {
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 20/26] audio: basic support for multichannel audio
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (18 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 19/26] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 21/26] paaudio: channel-map option Gerd Hoffmann
` (6 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 00998152d6e1c25b3194b9b71e27f14c4d26f396.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.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 eddf013a537c..f37ce1ce8570 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 c00f4deddd3d..7fc3aa9d1637 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) {
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 21/26] paaudio: channel-map option
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (19 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 20/26] audio: basic support for multichannel audio Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 22/26] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
` (5 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: 66670d43bd932be0919668cec6a8cd172bfb8383.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.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 d195b1caa8d8..20402b071804 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 0535eff7948f..07003808cba7 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 734a3b5d6569..6aee778896b8 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}][,...]]
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 22/26] usb-audio: do not count on avail bytes actually available
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (20 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 21/26] paaudio: channel-map option Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 23/26] usb-audio: support more than two channels of audio Gerd Hoffmann
` (4 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
This assumption is no longer true when mixeng is turned off.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 90b5ec62109a69ee7c28d95b367e40dc41ad658a.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
hw/usb/dev-audio.c | 30 ++++++++++++++++++------------
1 file changed, 18 insertions(+), 12 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index ae42e5a2f1d0..74c99b1f1204 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) {
- return;
- }
- data = streambuf_get(&s->out.buf);
+ while (avail) {
+ size_t written, len;
+
+ data = streambuf_get(&s->out.buf, &len);
if (!data) {
return;
}
- AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE);
- avail -= USBAUDIO_PACKET_SIZE;
+
+ written = AUD_write(s->out.voice, data, len);
+ avail -= written;
+ s->out.buf.cons += written;
+
+ if (written < len) {
+ return;
+ }
}
}
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 23/26] usb-audio: support more than two channels of audio
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (21 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 22/26] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 24/26] usbaudio: change playback counters to 64 bit Gerd Hoffmann
` (3 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
This commit adds support for 5.1 and 7.1 audio playback. This commit
adds a new property to usb-audio:
* multi=on|off
Whether to enable the 5.1 and 7.1 audio support. When off (default)
it continues to emulate the old stereo-only device. When on, it
emulates a slightly different audio device that supports 5.1 and 7.1
audio.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 17a517c599bf8e906f41430b97bbe07ebe7eeef5.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
hw/usb/dev-audio.c | 419 +++++++++++++++++++++++++++++++++++++++------
1 file changed, 366 insertions(+), 53 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 74c99b1f1204..e42bdfbdc101 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -37,11 +37,15 @@
#include "desc.h"
#include "audio/audio.h"
+static void usb_audio_reinit(USBDevice *dev, unsigned channels);
+
#define USBAUDIO_VENDOR_NUM 0x46f4 /* CRC16() of "QEMU" */
#define USBAUDIO_PRODUCT_NUM 0x0002
#define DEV_CONFIG_VALUE 1 /* The one and only */
+#define USBAUDIO_MAX_CHANNELS(s) (s->multi ? 8 : 2)
+
/* Descriptor subtypes for AC interfaces */
#define DST_AC_HEADER 1
#define DST_AC_INPUT_TERMINAL 2
@@ -80,6 +84,27 @@ static const USBDescStrings usb_audio_stringtable = {
[STRING_REAL_STREAM] = "Audio Output - 48 kHz Stereo",
};
+/*
+ * A USB audio device supports an arbitrary number of alternate
+ * interface settings for each interface. Each corresponds to a block
+ * diagram of parameterized blocks. This can thus refer to things like
+ * number of channels, data rates, or in fact completely different
+ * block diagrams. Alternative setting 0 is always the null block diagram,
+ * which is used by a disabled device.
+ */
+enum usb_audio_altset {
+ ALTSET_OFF = 0x00, /* No endpoint */
+ ALTSET_STEREO = 0x01, /* Single endpoint */
+ ALTSET_51 = 0x02,
+ ALTSET_71 = 0x03,
+};
+
+static unsigned altset_channels[] = {
+ [ALTSET_STEREO] = 2,
+ [ALTSET_51] = 6,
+ [ALTSET_71] = 8,
+};
+
#define U16(x) ((x) & 0xff), (((x) >> 8) & 0xff)
#define U24(x) U16(x), (((x) >> 16) & 0xff)
#define U32(x) U24(x), (((x) >> 24) & 0xff)
@@ -87,7 +112,8 @@ static const USBDescStrings usb_audio_stringtable = {
/*
* A Basic Audio Device uses these specific values
*/
-#define USBAUDIO_PACKET_SIZE 192
+#define USBAUDIO_PACKET_SIZE_BASE 96
+#define USBAUDIO_PACKET_SIZE(channels) (USBAUDIO_PACKET_SIZE_BASE * channels)
#define USBAUDIO_SAMPLE_RATE 48000
#define USBAUDIO_PACKET_INTERVAL 1
@@ -121,7 +147,7 @@ static const USBDescIface desc_iface[] = {
0x01, /* u8 bTerminalID */
U16(0x0101), /* u16 wTerminalType */
0x00, /* u8 bAssocTerminal */
- 0x02, /* u16 bNrChannels */
+ 0x02, /* u8 bNrChannels */
U16(0x0003), /* u16 wChannelConfig */
0x00, /* u8 iChannelNames */
STRING_INPUT_TERMINAL, /* u8 iTerminal */
@@ -156,14 +182,14 @@ static const USBDescIface desc_iface[] = {
},
},{
.bInterfaceNumber = 1,
- .bAlternateSetting = 0,
+ .bAlternateSetting = ALTSET_OFF,
.bNumEndpoints = 0,
.bInterfaceClass = USB_CLASS_AUDIO,
.bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
.iInterface = STRING_NULL_STREAM,
},{
.bInterfaceNumber = 1,
- .bAlternateSetting = 1,
+ .bAlternateSetting = ALTSET_STEREO,
.bNumEndpoints = 1,
.bInterfaceClass = USB_CLASS_AUDIO,
.bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
@@ -199,7 +225,7 @@ static const USBDescIface desc_iface[] = {
{
.bEndpointAddress = USB_DIR_OUT | 0x01,
.bmAttributes = 0x0d,
- .wMaxPacketSize = USBAUDIO_PACKET_SIZE,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2),
.bInterval = 1,
.is_audio = 1,
/* Stereo Headphone Class-specific
@@ -247,17 +273,274 @@ static const USBDesc desc_audio = {
.str = usb_audio_stringtable,
};
-/*
- * A USB audio device supports an arbitrary number of alternate
- * interface settings for each interface. Each corresponds to a block
- * diagram of parameterized blocks. This can thus refer to things like
- * number of channels, data rates, or in fact completely different
- * block diagrams. Alternative setting 0 is always the null block diagram,
- * which is used by a disabled device.
- */
-enum usb_audio_altset {
- ALTSET_OFF = 0x00, /* No endpoint */
- ALTSET_ON = 0x01, /* Single endpoint */
+/* multi channel compatible desc */
+
+static const USBDescIface desc_iface_multi[] = {
+ {
+ .bInterfaceNumber = 0,
+ .bNumEndpoints = 0,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceProtocol = 0x04,
+ .iInterface = STRING_USBAUDIO_CONTROL,
+ .ndesc = 4,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-Specific AC Interface Header Descriptor */
+ .data = (uint8_t[]) {
+ 0x09, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_HEADER, /* u8 bDescriptorSubtype */
+ U16(0x0100), /* u16 bcdADC */
+ U16(0x38), /* u16 wTotalLength */
+ 0x01, /* u8 bInCollection */
+ 0x01, /* u8 baInterfaceNr */
+ }
+ },{
+ /* Generic Stereo Input Terminal ID1 Descriptor */
+ .data = (uint8_t[]) {
+ 0x0c, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_INPUT_TERMINAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalID */
+ U16(0x0101), /* u16 wTerminalType */
+ 0x00, /* u8 bAssocTerminal */
+ 0x08, /* u8 bNrChannels */
+ U16(0x063f), /* u16 wChannelConfig */
+ 0x00, /* u8 iChannelNames */
+ STRING_INPUT_TERMINAL, /* u8 iTerminal */
+ }
+ },{
+ /* Generic Stereo Feature Unit ID2 Descriptor */
+ .data = (uint8_t[]) {
+ 0x19, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_FEATURE_UNIT, /* u8 bDescriptorSubtype */
+ 0x02, /* u8 bUnitID */
+ 0x01, /* u8 bSourceID */
+ 0x02, /* u8 bControlSize */
+ U16(0x0001), /* u16 bmaControls(0) */
+ U16(0x0002), /* u16 bmaControls(1) */
+ U16(0x0002), /* u16 bmaControls(2) */
+ U16(0x0002), /* u16 bmaControls(3) */
+ U16(0x0002), /* u16 bmaControls(4) */
+ U16(0x0002), /* u16 bmaControls(5) */
+ U16(0x0002), /* u16 bmaControls(6) */
+ U16(0x0002), /* u16 bmaControls(7) */
+ U16(0x0002), /* u16 bmaControls(8) */
+ STRING_FEATURE_UNIT, /* u8 iFeature */
+ }
+ },{
+ /* Headphone Ouptut Terminal ID3 Descriptor */
+ .data = (uint8_t[]) {
+ 0x09, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_OUTPUT_TERMINAL, /* u8 bDescriptorSubtype */
+ 0x03, /* u8 bUnitID */
+ U16(0x0301), /* u16 wTerminalType (SPK) */
+ 0x00, /* u8 bAssocTerminal */
+ 0x02, /* u8 bSourceID */
+ STRING_OUTPUT_TERMINAL, /* u8 iTerminal */
+ }
+ }
+ },
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_OFF,
+ .bNumEndpoints = 0,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_NULL_STREAM,
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_STEREO,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x02, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_51,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x06, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(6),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_71,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x08, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(8),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ }
+};
+
+static const USBDescDevice desc_device_multi = {
+ .bcdUSB = 0x0100,
+ .bMaxPacketSize0 = 64,
+ .bNumConfigurations = 1,
+ .confs = (USBDescConfig[]) {
+ {
+ .bNumInterfaces = 2,
+ .bConfigurationValue = DEV_CONFIG_VALUE,
+ .iConfiguration = STRING_CONFIG,
+ .bmAttributes = USB_CFG_ATT_ONE | USB_CFG_ATT_SELFPOWER,
+ .bMaxPower = 0x32,
+ .nif = ARRAY_SIZE(desc_iface_multi),
+ .ifs = desc_iface_multi,
+ }
+ },
+};
+
+static const USBDesc desc_audio_multi = {
+ .id = {
+ .idVendor = USBAUDIO_VENDOR_NUM,
+ .idProduct = USBAUDIO_PRODUCT_NUM,
+ .bcdDevice = 0,
+ .iManufacturer = STRING_MANUFACTURER,
+ .iProduct = STRING_PRODUCT,
+ .iSerialNumber = STRING_SERIALNUMBER,
+ },
+ .full = &desc_device_multi,
+ .str = usb_audio_stringtable,
};
/*
@@ -300,10 +583,11 @@ struct streambuf {
uint32_t cons;
};
-static void streambuf_init(struct streambuf *buf, uint32_t size)
+static void streambuf_init(struct streambuf *buf, uint32_t size,
+ uint32_t channels)
{
g_free(buf->data);
- buf->size = size - (size % USBAUDIO_PACKET_SIZE);
+ buf->size = size - (size % USBAUDIO_PACKET_SIZE(channels));
buf->data = g_malloc(buf->size);
buf->prod = 0;
buf->cons = 0;
@@ -315,21 +599,21 @@ static void streambuf_fini(struct streambuf *buf)
buf->data = NULL;
}
-static int streambuf_put(struct streambuf *buf, USBPacket *p)
+static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
{
uint32_t free = buf->size - (buf->prod - buf->cons);
- if (free < USBAUDIO_PACKET_SIZE) {
+ if (free < USBAUDIO_PACKET_SIZE(channels)) {
return 0;
}
- if (p->iov.size != USBAUDIO_PACKET_SIZE) {
+ if (p->iov.size != USBAUDIO_PACKET_SIZE(channels)) {
return 0;
}
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
- USBAUDIO_PACKET_SIZE);
- buf->prod += USBAUDIO_PACKET_SIZE;
- return USBAUDIO_PACKET_SIZE;
+ USBAUDIO_PACKET_SIZE(channels));
+ buf->prod += USBAUDIO_PACKET_SIZE(channels);
+ return USBAUDIO_PACKET_SIZE(channels);
}
static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
@@ -357,14 +641,15 @@ typedef struct USBAudioState {
enum usb_audio_altset altset;
struct audsettings as;
SWVoiceOut *voice;
- bool mute;
- uint8_t vol[2];
+ Volume vol;
struct streambuf buf;
+ uint32_t channels;
} out;
/* properties */
uint32_t debug;
- uint32_t buffer;
+ uint32_t buffer_user, buffer;
+ bool multi;
} USBAudioState;
#define TYPE_USB_AUDIO "usb-audio"
@@ -397,10 +682,15 @@ static int usb_audio_set_output_altset(USBAudioState *s, int altset)
{
switch (altset) {
case ALTSET_OFF:
- streambuf_init(&s->out.buf, s->buffer);
AUD_set_active_out(s->out.voice, false);
break;
- case ALTSET_ON:
+ case ALTSET_STEREO:
+ case ALTSET_51:
+ case ALTSET_71:
+ if (s->out.channels != altset_channels[altset]) {
+ usb_audio_reinit(USB_DEVICE(s), altset_channels[altset]);
+ }
+ streambuf_init(&s->out.buf, s->buffer, s->out.channels);
AUD_set_active_out(s->out.voice, true);
break;
default:
@@ -431,33 +721,33 @@ static int usb_audio_get_control(USBAudioState *s, uint8_t attrib,
switch (aid) {
case ATTRIB_ID(MUTE_CONTROL, CR_GET_CUR, 0x0200):
- data[0] = s->out.mute;
+ data[0] = s->out.vol.mute;
ret = 1;
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_CUR, 0x0200):
- if (cn < 2) {
- uint16_t vol = (s->out.vol[cn] * 0x8800 + 127) / 255 + 0x8000;
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
+ uint16_t vol = (s->out.vol.vol[cn] * 0x8800 + 127) / 255 + 0x8000;
data[0] = vol;
data[1] = vol >> 8;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MIN, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x01;
data[1] = 0x80;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MAX, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x00;
data[1] = 0x08;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_RES, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x88;
data[1] = 0x00;
ret = 2;
@@ -479,16 +769,17 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib,
switch (aid) {
case ATTRIB_ID(MUTE_CONTROL, CR_SET_CUR, 0x0200):
- s->out.mute = data[0] & 1;
+ s->out.vol.mute = data[0] & 1;
set_vol = true;
ret = 0;
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_SET_CUR, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
uint16_t vol = data[0] + (data[1] << 8);
if (s->debug) {
- fprintf(stderr, "usb-audio: vol %04x\n", (uint16_t)vol);
+ fprintf(stderr, "usb-audio: cn %d vol %04x\n", cn,
+ (uint16_t)vol);
}
vol -= 0x8000;
@@ -497,7 +788,7 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib,
vol = 255;
}
- s->out.vol[cn] = vol;
+ s->out.vol.vol[cn] = vol;
set_vol = true;
ret = 0;
}
@@ -506,11 +797,14 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib,
if (set_vol) {
if (s->debug) {
- fprintf(stderr, "usb-audio: mute %d, lvol %3d, rvol %3d\n",
- s->out.mute, s->out.vol[0], s->out.vol[1]);
+ int i;
+ fprintf(stderr, "usb-audio: mute %d", s->out.vol.mute);
+ for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) {
+ fprintf(stderr, ", vol[%d] %3d", i, s->out.vol.vol[i]);
+ }
+ fprintf(stderr, "\n");
}
- AUD_set_volume_out(s->out.voice, s->out.mute,
- s->out.vol[0], s->out.vol[1]);
+ audio_set_volume_out(s->out.voice, &s->out.vol);
}
return ret;
@@ -603,7 +897,7 @@ static void usb_audio_handle_dataout(USBAudioState *s, USBPacket *p)
return;
}
- streambuf_put(&s->out.buf, p);
+ streambuf_put(&s->out.buf, p, s->out.channels);
if (p->actual_length < p->iov.size && s->debug > 1) {
fprintf(stderr, "usb-audio: output overrun (%zd bytes)\n",
p->iov.size - p->actual_length);
@@ -645,6 +939,9 @@ static void usb_audio_unrealize(USBDevice *dev, Error **errp)
static void usb_audio_realize(USBDevice *dev, Error **errp)
{
USBAudioState *s = USB_AUDIO(dev);
+ int i;
+
+ dev->usb_desc = s->multi ? &desc_audio_multi : &desc_audio;
usb_desc_create_serial(dev);
usb_desc_init(dev);
@@ -652,18 +949,35 @@ static void usb_audio_realize(USBDevice *dev, Error **errp)
AUD_register_card(TYPE_USB_AUDIO, &s->card);
s->out.altset = ALTSET_OFF;
- s->out.mute = false;
- s->out.vol[0] = 240; /* 0 dB */
- s->out.vol[1] = 240; /* 0 dB */
+ s->out.vol.mute = false;
+ for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) {
+ s->out.vol.vol[i] = 240; /* 0 dB */
+ }
+
+ usb_audio_reinit(dev, 2);
+}
+
+static void usb_audio_reinit(USBDevice *dev, unsigned channels)
+{
+ USBAudioState *s = USB_AUDIO(dev);
+
+ s->out.channels = channels;
+ if (!s->buffer_user) {
+ s->buffer = 32 * USBAUDIO_PACKET_SIZE(s->out.channels);
+ } else {
+ s->buffer = s->buffer_user;
+ }
+
+ s->out.vol.channels = s->out.channels;
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
- s->out.as.nchannels = 2;
+ s->out.as.nchannels = s->out.channels;
s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
- streambuf_init(&s->out.buf, s->buffer);
+ streambuf_init(&s->out.buf, s->buffer, s->out.channels);
s->out.voice = AUD_open_out(&s->card, s->out.voice, TYPE_USB_AUDIO,
s, output_callback, &s->out.as);
- AUD_set_volume_out(s->out.voice, s->out.mute, s->out.vol[0], s->out.vol[1]);
+ audio_set_volume_out(s->out.voice, &s->out.vol);
AUD_set_active_out(s->out.voice, 0);
}
@@ -675,8 +989,8 @@ static const VMStateDescription vmstate_usb_audio = {
static Property usb_audio_properties[] = {
DEFINE_AUDIO_PROPERTIES(USBAudioState, card),
DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0),
- DEFINE_PROP_UINT32("buffer", USBAudioState, buffer,
- 32 * USBAUDIO_PACKET_SIZE),
+ DEFINE_PROP_UINT32("buffer", USBAudioState, buffer_user, 0),
+ DEFINE_PROP_BOOL("multi", USBAudioState, multi, false),
DEFINE_PROP_END_OF_LIST(),
};
@@ -689,7 +1003,6 @@ static void usb_audio_class_init(ObjectClass *klass, void *data)
dc->props = usb_audio_properties;
set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
k->product_desc = "QEMU USB Audio Interface";
- k->usb_desc = &desc_audio;
k->realize = usb_audio_realize;
k->handle_reset = usb_audio_handle_reset;
k->handle_control = usb_audio_handle_control;
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 24/26] usbaudio: change playback counters to 64 bit
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (22 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 23/26] usb-audio: support more than two channels of audio Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 25/26] audio: fix buffer-length typo in documentation Gerd Hoffmann
` (2 subsequent siblings)
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann, Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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>
Message-id: c0a8d934c145bab55506e0623cc0206e2632a196.1568574965.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.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 e42bdfbdc101..ea604bbb8e4a 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;
}
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 25/26] audio: fix buffer-length typo in documentation
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (23 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 24/26] usbaudio: change playback counters to 64 bit Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 26/26] audio: fix ALSA period-length " Gerd Hoffmann
2019-09-19 15:15 ` [Qemu-devel] [PULL 00/26] Audio 20190919 patches Peter Maydell
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Stefan Hajnoczi, Markus Armbruster
From: Stefan Hajnoczi <stefanha@redhat.com>
Fixes: f0b3d811529 ("audio: -audiodev command line option: documentation")
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Message-Id: <20190918095335.7646-2-stefanha@redhat.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
qemu-options.hx | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/qemu-options.hx b/qemu-options.hx
index 6aee778896b8..e9417e6561b8 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -440,7 +440,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
" in|out.format= sample format to use with fixed settings\n"
" valid values: s8, s16, s32, u8, u16, u32\n"
" in|out.voices= number of voices to use\n"
- " in|out.buffer-len= length of buffer in microseconds\n"
+ " in|out.buffer-length= length of buffer in microseconds\n"
"-audiodev none,id=id,[,prop[=value][,...]]\n"
" dummy driver that discards all output\n"
#ifdef CONFIG_AUDIO_ALSA
@@ -531,7 +531,7 @@ Valid values are: @code{s8}, @code{s16}, @code{s32}, @code{u8},
@item in|out.voices=@var{voices}
Specify the number of @var{voices} to use. Default is 1.
-@item in|out.buffer=@var{usecs}
+@item in|out.buffer-length=@var{usecs}
Sets the size of the buffer in microseconds.
@end table
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* [Qemu-devel] [PULL 26/26] audio: fix ALSA period-length typo in documentation
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (24 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 25/26] audio: fix buffer-length typo in documentation Gerd Hoffmann
@ 2019-09-19 8:36 ` Gerd Hoffmann
2019-09-19 15:15 ` [Qemu-devel] [PULL 00/26] Audio 20190919 patches Peter Maydell
26 siblings, 0 replies; 31+ messages in thread
From: Gerd Hoffmann @ 2019-09-19 8:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Stefan Hajnoczi, Markus Armbruster
From: Stefan Hajnoczi <stefanha@redhat.com>
Fixes: f0b3d811529 ("audio: -audiodev command line option: documentation")
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Message-Id: <20190918095335.7646-4-stefanha@redhat.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
qemu-options.hx | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/qemu-options.hx b/qemu-options.hx
index e9417e6561b8..877edc677160 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -446,7 +446,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
#ifdef CONFIG_AUDIO_ALSA
"-audiodev alsa,id=id[,prop[=value][,...]]\n"
" in|out.dev= name of the audio device to use\n"
- " in|out.period-len= length of period in microseconds\n"
+ " in|out.period-length= length of period in microseconds\n"
" in|out.try-poll= attempt to use poll mode\n"
" threshold= threshold (in microseconds) when playback starts\n"
#endif
@@ -552,7 +552,7 @@ ALSA specific options are:
Specify the ALSA @var{device} to use for input and/or output. Default
is @code{default}.
-@item in|out.period-len=@var{usecs}
+@item in|out.period-length=@var{usecs}
Sets the period length in microseconds.
@item in|out.try-poll=on|off
--
2.18.1
^ permalink raw reply related [flat|nested] 31+ messages in thread
* Re: [Qemu-devel] [PULL 00/26] Audio 20190919 patches
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
` (25 preceding siblings ...)
2019-09-19 8:36 ` [Qemu-devel] [PULL 26/26] audio: fix ALSA period-length " Gerd Hoffmann
@ 2019-09-19 15:15 ` Peter Maydell
2019-09-19 15:28 ` Eric Blake
26 siblings, 1 reply; 31+ messages in thread
From: Peter Maydell @ 2019-09-19 15:15 UTC (permalink / raw)
To: Gerd Hoffmann; +Cc: QEMU Developers, Markus Armbruster
On Thu, 19 Sep 2019 at 09:38, Gerd Hoffmann <kraxel@redhat.com> wrote:
>
> The following changes since commit f8c3db33a5e863291182f8862ddf81618a7c6194:
>
> target/sparc: Switch to do_transaction_failed() hook (2019-09-17 12:01:00 +0100)
>
> are available in the Git repository at:
>
> git://git.kraxel.org/qemu tags/audio-20190919-pull-request
>
> for you to fetch changes up to cf0c1c2aa32db5d658c3c797ad995a6d571bad96:
>
> audio: fix ALSA period-length typo in documentation (2019-09-19 10:32:48 +0200)
>
> ----------------------------------------------------------------
> audio: make mixeng optional.
> audio: add surround sound support.
> audio: documentation fixes.
>
Hi; I'm afraid this fails to build on OSX/FreeBSD/OpenBSD/Windows,
with format string issues:
/Users/pm215/src/qemu-for-merges/audio/wavaudio.c:50:15: error: format
specifies type 'size_t' (aka 'unsigned long') but the argument has
type 'int64_t' (aka 'long long') [-Werror,-Wformat]
bytes, strerror(errno));
^~~~~
/Users/pm215/src/qemu-for-merges/audio/audio_int.h:257:52: note:
expanded from macro 'dolog'
#define dolog(fmt, ...) AUD_log(AUDIO_CAP, fmt, ## __VA_ARGS__)
^~~~~~~~~~~
thanks
-- PMM
^ permalink raw reply [flat|nested] 31+ messages in thread
* Re: [PULL 15/26] audio: add mixeng option (documentation)
2019-09-19 8:36 ` [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation) Gerd Hoffmann
@ 2019-09-19 15:27 ` Eric Blake
0 siblings, 0 replies; 31+ messages in thread
From: Eric Blake @ 2019-09-19 15:27 UTC (permalink / raw)
To: Gerd Hoffmann, qemu-devel
Cc: Markus Armbruster, Kővágó, Zoltán
[-- Attachment #1.1: Type: text/plain, Size: 1701 bytes --]
On 9/19/19 3:36 AM, Gerd Hoffmann wrote:
> From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
>
> 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.
>
> +++ 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-engineeng= use mixing engine to mix streams inside QEMU\n"
s/engineeng/engine/
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 31+ messages in thread
* Re: [Qemu-devel] [PULL 00/26] Audio 20190919 patches
2019-09-19 15:15 ` [Qemu-devel] [PULL 00/26] Audio 20190919 patches Peter Maydell
@ 2019-09-19 15:28 ` Eric Blake
2019-09-19 21:26 ` Zoltán Kővágó
0 siblings, 1 reply; 31+ messages in thread
From: Eric Blake @ 2019-09-19 15:28 UTC (permalink / raw)
To: Peter Maydell, Gerd Hoffmann; +Cc: QEMU Developers, Markus Armbruster
[-- Attachment #1.1: Type: text/plain, Size: 1343 bytes --]
On 9/19/19 10:15 AM, Peter Maydell wrote:
> On Thu, 19 Sep 2019 at 09:38, Gerd Hoffmann <kraxel@redhat.com> wrote:
>>
>> The following changes since commit f8c3db33a5e863291182f8862ddf81618a7c6194:
>>
>> target/sparc: Switch to do_transaction_failed() hook (2019-09-17 12:01:00 +0100)
>>
>> are available in the Git repository at:
>>
>> git://git.kraxel.org/qemu tags/audio-20190919-pull-request
>>
>> for you to fetch changes up to cf0c1c2aa32db5d658c3c797ad995a6d571bad96:
>>
>> audio: fix ALSA period-length typo in documentation (2019-09-19 10:32:48 +0200)
>>
>> ----------------------------------------------------------------
>> audio: make mixeng optional.
>> audio: add surround sound support.
>> audio: documentation fixes.
>>
>
> Hi; I'm afraid this fails to build on OSX/FreeBSD/OpenBSD/Windows,
> with format string issues:
>
> /Users/pm215/src/qemu-for-merges/audio/wavaudio.c:50:15: error: format
> specifies type 'size_t' (aka 'unsigned long') but the argument has
> type 'int64_t' (aka 'long long') [-Werror,-Wformat]
> bytes, strerror(errno));
> ^~~~~
As long as you spin a v2, it's also worth fixing a typo I found in 15/26.
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 31+ messages in thread
* Re: [Qemu-devel] [PULL 00/26] Audio 20190919 patches
2019-09-19 15:28 ` Eric Blake
@ 2019-09-19 21:26 ` Zoltán Kővágó
0 siblings, 0 replies; 31+ messages in thread
From: Zoltán Kővágó @ 2019-09-19 21:26 UTC (permalink / raw)
To: Eric Blake, Peter Maydell, Gerd Hoffmann
Cc: QEMU Developers, Markus Armbruster
On 2019-09-19 17:28, Eric Blake wrote:
> On 9/19/19 10:15 AM, Peter Maydell wrote:
>> On Thu, 19 Sep 2019 at 09:38, Gerd Hoffmann <kraxel@redhat.com> wrote:
>>>
>>> The following changes since commit f8c3db33a5e863291182f8862ddf81618a7c6194:
>>>
>>> target/sparc: Switch to do_transaction_failed() hook (2019-09-17 12:01:00 +0100)
>>>
>>> are available in the Git repository at:
>>>
>>> git://git.kraxel.org/qemu tags/audio-20190919-pull-request
>>>
>>> for you to fetch changes up to cf0c1c2aa32db5d658c3c797ad995a6d571bad96:
>>>
>>> audio: fix ALSA period-length typo in documentation (2019-09-19 10:32:48 +0200)
>>>
>>> ----------------------------------------------------------------
>>> audio: make mixeng optional.
>>> audio: add surround sound support.
>>> audio: documentation fixes.
>>>
>>
>> Hi; I'm afraid this fails to build on OSX/FreeBSD/OpenBSD/Windows,
>> with format string issues:
>>
>> /Users/pm215/src/qemu-for-merges/audio/wavaudio.c:50:15: error: format
>> specifies type 'size_t' (aka 'unsigned long') but the argument has
>> type 'int64_t' (aka 'long long') [-Werror,-Wformat]
>> bytes, strerror(errno));
>> ^~~~~
>
> As long as you spin a v2, it's also worth fixing a typo I found in 15/26.
>
Thanks, fixed in v4, along with the compilation problem.
^ permalink raw reply [flat|nested] 31+ messages in thread
end of thread, other threads:[~2019-09-19 21:43 UTC | newest]
Thread overview: 31+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2019-09-19 8:36 [Qemu-devel] [PULL 00/26] Audio 20190919 patches Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 01/26] audio: api for mixeng code free backends Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 02/26] alsaaudio: port to the new audio backend api Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 03/26] coreaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 04/26] dsoundaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 05/26] noaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 06/26] ossaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 07/26] paaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 08/26] sdlaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 09/26] spiceaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 10/26] wavaudio: " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 11/26] audio: remove remains of the old " Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 12/26] audio: unify input and output mixeng buffer management Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 13/26] audio: common rate control code for timer based outputs Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 14/26] audio: split ctl_* functions into enable_* and volume_* Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 15/26] audio: add mixeng option (documentation) Gerd Hoffmann
2019-09-19 15:27 ` Eric Blake
2019-09-19 8:36 ` [Qemu-devel] [PULL 16/26] audio: make mixeng optional Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 17/26] paaudio: get/put_buffer functions Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 18/26] audio: support more than two channels in volume setting Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 19/26] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 20/26] audio: basic support for multichannel audio Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 21/26] paaudio: channel-map option Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 22/26] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 23/26] usb-audio: support more than two channels of audio Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 24/26] usbaudio: change playback counters to 64 bit Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 25/26] audio: fix buffer-length typo in documentation Gerd Hoffmann
2019-09-19 8:36 ` [Qemu-devel] [PULL 26/26] audio: fix ALSA period-length " Gerd Hoffmann
2019-09-19 15:15 ` [Qemu-devel] [PULL 00/26] Audio 20190919 patches Peter Maydell
2019-09-19 15:28 ` Eric Blake
2019-09-19 21:26 ` Zoltán Kővágó
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