* [PULL 01/13] audio: fix parameter dereference before NULL check
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 02/13] audio: paaudio: fix connection and stream name Gerd Hoffmann
` (12 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann,
Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
This should fix Coverity issues CID 1405305 and 1405301.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 0eadcc88b8421bb86ce2d68ac70517f920c3ad6c.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_template.h | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 235d1acbbebb..1006d6b1eb8a 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -425,8 +425,8 @@ SW *glue (AUD_open_, TYPE) (
struct audsettings *as
)
{
- AudioState *s = card->state;
- AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
+ AudioState *s;
+ AudiodevPerDirectionOptions *pdo;
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -434,6 +434,9 @@ SW *glue (AUD_open_, TYPE) (
goto fail;
}
+ s = card->state;
+ pdo = glue(audio_get_pdo_, TYPE)(s->dev);
+
ldebug ("open %s, freq %d, nchannels %d, fmt %d\n",
name, as->freq, as->nchannels, as->fmt);
--
2.18.1
^ permalink raw reply related [flat|nested] 15+ messages in thread* [PULL 02/13] audio: paaudio: fix connection and stream name
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
2019-10-18 7:41 ` [PULL 01/13] audio: fix parameter dereference before NULL check Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 03/13] audio: paaudio: ability to specify " Gerd Hoffmann
` (11 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann,
Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
Connection name was previously erroneously set to the server socket
path, while connection names were simply "qemu". After this patch, the
connection name will be the vm name (falling back to "qemu" if not
specified), while stream names will be the audiodev's id.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3d139426031a400a68d440608ba5e43f0e116cd8.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/paaudio.c | 9 ++++++---
1 file changed, 6 insertions(+), 3 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index ed31f863f7fe..3e6580a5ee50 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,6 +2,7 @@
#include "qemu/osdep.h"
#include "qemu/module.h"
+#include "qemu-common.h"
#include "audio.h"
#include "qapi/opts-visitor.h"
@@ -338,7 +339,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
pa->stream = qpa_simple_new (
c,
- "qemu",
+ g->dev->id,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
@@ -387,7 +388,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
pa->stream = qpa_simple_new (
c,
- "qemu",
+ g->dev->id,
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
@@ -549,6 +550,7 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
/* common */
static void *qpa_conn_init(const char *server)
{
+ const char *vm_name;
PAConnection *c = g_malloc0(sizeof(PAConnection));
QTAILQ_INSERT_TAIL(&pa_conns, c, list);
@@ -557,8 +559,9 @@ static void *qpa_conn_init(const char *server)
goto fail;
}
+ vm_name = qemu_get_vm_name();
c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
- server);
+ vm_name ? vm_name : "qemu");
if (!c->context) {
goto fail;
}
--
2.18.1
^ permalink raw reply related [flat|nested] 15+ messages in thread* [PULL 03/13] audio: paaudio: ability to specify stream name
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
2019-10-18 7:41 ` [PULL 01/13] audio: fix parameter dereference before NULL check Gerd Hoffmann
2019-10-18 7:41 ` [PULL 02/13] audio: paaudio: fix connection and stream name Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 04/13] audio: add mixing-engine option (documentation) Gerd Hoffmann
` (10 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 UTC (permalink / raw)
To: qemu-devel
Cc: Kővágó, Zoltán, Gerd Hoffmann,
Markus Armbruster
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
This can be used to identify stream in tools like pavucontrol when one
creates multiple -audiodevs or runs multiple qemu instances.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 2d6e337c474ac84172d0809e6959c26b21d48120.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/paaudio.c | 4 ++--
qapi/audio.json | 6 ++++++
2 files changed, 8 insertions(+), 2 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 3e6580a5ee50..2cba492d8265 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -339,7 +339,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
pa->stream = qpa_simple_new (
c,
- g->dev->id,
+ ppdo->has_stream_name ? ppdo->stream_name : g->dev->id,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
@@ -388,7 +388,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
pa->stream = qpa_simple_new (
c,
- g->dev->id,
+ ppdo->has_stream_name ? ppdo->stream_name : g->dev->id,
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
diff --git a/qapi/audio.json b/qapi/audio.json
index 9fefdf5186dd..e45218f081f2 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -206,6 +206,11 @@
#
# @name: name of the sink/source to use
#
+# @stream-name: name of the PulseAudio stream created by qemu. Can be
+# used to identify the stream in PulseAudio when you
+# create multiple PulseAudio devices or run multiple qemu
+# instances (default: audiodev's id, since 4.2)
+#
# @latency: latency you want PulseAudio to achieve in microseconds
# (default 15000)
#
@@ -215,6 +220,7 @@
'base': 'AudiodevPerDirectionOptions',
'data': {
'*name': 'str',
+ '*stream-name': 'str',
'*latency': 'uint32' } }
##
--
2.18.1
^ permalink raw reply related [flat|nested] 15+ messages in thread* [PULL 04/13] audio: add mixing-engine option (documentation)
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (2 preceding siblings ...)
2019-10-18 7:41 ` [PULL 03/13] audio: paaudio: ability to specify " Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 05/13] audio: make mixeng optional Gerd Hoffmann
` (9 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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.
Originally thw two main use cases of the disabled option was: using
unsupported audio formats (5.1 and 7.1 audio) and having different
pulseaudio streams per audio frontend. Since we can have multiple
-audiodevs, the latter is not that important, so currently you only need
this option if you want to use 5.1 or 7.1 audio (implemented in a later
patch), otherwise it's probably better to stick to the old and tried
mixeng, since it's less picky about the backends.
The ideal solution would be to port as much as possible to gstreamer,
but this is currently out of scope:
https://wiki.qemu.org/Internships/ProjectIdeas/AudioGStreamer
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 5765186a7aadd51a72bc7d3e804307f0ee8a34ce.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
qapi/audio.json | 6 ++++++
qemu-options.hx | 15 +++++++++++++++
2 files changed, 21 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index e45218f081f2..83312b23391e 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -11,6 +11,11 @@
# 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 and
+# convert audio formats when not supported by the backend. When
+# set to off, fixed-settings must be also off (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 +36,7 @@
##
{ 'struct': 'AudiodevPerDirectionOptions',
'data': {
+ '*mixing-engine': 'bool',
'*fixed-settings': 'bool',
'*frequency': 'uint32',
'*channels': 'uint32',
diff --git a/qemu-options.hx b/qemu-options.hx
index 793d70ff9388..996b6fba7461 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
" specifies the audio backend to use\n"
" id= identifier of the backend\n"
" timer-period= timer period in microseconds\n"
+ " in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n"
" in|out.fixed-settings= use fixed settings for host audio\n"
" in|out.frequency= frequency to use with fixed settings\n"
" in|out.channels= number of channels to use with fixed settings\n"
@@ -493,6 +494,10 @@ output's property with @code{out.@var{prop}}. For example:
-audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified
@end example
+NOTE: parameter validation is known to be incomplete, in many cases
+specifying an invalid option causes QEMU to print an error message and
+continue emulation without sound.
+
Valid global options are:
@table @option
@@ -503,6 +508,16 @@ 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 and convert
+audio formats when not supported by the backend. When off,
+@var{fixed-settings} must be off too. Note that disabling this option
+means that the selected backend must support multiple streams and the
+audio formats used by the virtual cards, otherwise you'll get no sound.
+It's not recommended to disable this option unless you want to use 5.1
+or 7.1 audio, as mixing engine only supports mono and stereo audio.
+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] 15+ messages in thread* [PULL 05/13] audio: make mixeng optional
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (3 preceding siblings ...)
2019-10-18 7:41 ` [PULL 04/13] audio: add mixing-engine option (documentation) Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 06/13] paaudio: get/put_buffer functions Gerd Hoffmann
` (8 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: c05bc258889ed289e8ee1bdbcc5e84174ec221e7.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/audio_template.h | 24 ++++++++++-----
audio/audio.c | 70 ++++++++++++++++++++++++++++++++++++++----
2 files changed, 80 insertions(+), 14 deletions(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 1006d6b1eb8a..3287d7075e6a 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));
@@ -328,9 +336,9 @@ static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
HW *hw;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
- if (pdo->fixed_settings) {
+ if (!pdo->mixing_engine || pdo->fixed_settings) {
hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
- if (hw) {
+ if (!pdo->mixing_engine || hw) {
return hw;
}
}
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] 15+ messages in thread* [PULL 06/13] paaudio: get/put_buffer functions
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (4 preceding siblings ...)
2019-10-18 7:41 ` [PULL 05/13] audio: make mixeng optional Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 07/13] audio: support more than two channels in volume setting Gerd Hoffmann
` (7 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: d03d30138b9b5a9681cc90cbfbfec0a197cac88c.1570996490.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 2cba492d8265..77cd0bc0d026 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -99,6 +99,59 @@ 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;
@@ -137,6 +190,32 @@ 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;
@@ -701,11 +780,15 @@ 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] 15+ messages in thread* [PULL 07/13] audio: support more than two channels in volume setting
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (5 preceding siblings ...)
2019-10-18 7:41 ` [PULL 06/13] paaudio: get/put_buffer functions Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 08/13] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
` (6 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: 5d3dd2ee3baaa62805e79c3901abb7415ae32461.1570996490.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 77cd0bc0d026..292c8c9ff4c0 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -532,20 +532,22 @@ 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);
@@ -572,20 +574,22 @@ 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] 15+ messages in thread* [PULL 08/13] audio: replace shift in audio_pcm_info with bytes_per_frame
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (6 preceding siblings ...)
2019-10-18 7:41 ` [PULL 07/13] audio: support more than two channels in volume setting Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 09/13] audio: basic support for multichannel audio Gerd Hoffmann
` (5 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.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 47efdc1b1eab..e46d834bd3af 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
- assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+ assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
bytes, strerror(errno));
}
- wav->total_samples += bytes >> hw->info.shift;
+ wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
uint8_t rlen[4];
uint8_t dlen[4];
- uint32_t datalen = wav->total_samples << hw->info.shift;
+ uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {
--
2.18.1
^ permalink raw reply related [flat|nested] 15+ messages in thread* [PULL 09/13] audio: basic support for multichannel audio
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (7 preceding siblings ...)
2019-10-18 7:41 ` [PULL 08/13] audio: replace shift in audio_pcm_info with bytes_per_frame Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 10/13] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
` (4 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: 7e53be1f97e939ed3bb729ef39e76b775643118a.1570996490.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] 15+ messages in thread* [PULL 10/13] usb-audio: do not count on avail bytes actually available
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (8 preceding siblings ...)
2019-10-18 7:41 ` [PULL 09/13] audio: basic support for multichannel audio Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 11/13] usb-audio: support more than two channels of audio Gerd Hoffmann
` (3 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: d63f4d39a0ee7a2e4e7e4a2eb005ba79120eaf1d.1570996490.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] 15+ messages in thread* [PULL 11/13] usb-audio: support more than two channels of audio
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (9 preceding siblings ...)
2019-10-18 7:41 ` [PULL 10/13] usb-audio: do not count on avail bytes actually available Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 12/13] usbaudio: change playback counters to 64 bit Gerd Hoffmann
` (2 subsequent siblings)
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: 98e96606228afa907fa238eac26573d5af63434a.1570996490.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] 15+ messages in thread* [PULL 12/13] usbaudio: change playback counters to 64 bit
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (10 preceding siblings ...)
2019-10-18 7:41 ` [PULL 11/13] usb-audio: support more than two channels of audio Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 7:41 ` [PULL 13/13] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams Gerd Hoffmann
2019-10-18 15:36 ` [PULL 00/13] Audio 20191018 patches Peter Maydell
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: ff866985ed369f1e18ea7c70da6a7fce8e241deb.1570996490.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] 15+ messages in thread* [PULL 13/13] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (11 preceding siblings ...)
2019-10-18 7:41 ` [PULL 12/13] usbaudio: change playback counters to 64 bit Gerd Hoffmann
@ 2019-10-18 7:41 ` Gerd Hoffmann
2019-10-18 15:36 ` [PULL 00/13] Audio 20191018 patches Peter Maydell
13 siblings, 0 replies; 15+ messages in thread
From: Gerd Hoffmann @ 2019-10-18 7:41 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: 2900e462d27bd73277ae083d037c32b1b4451ee2.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
---
audio/paaudio.c | 50 ++++++++++++++++++++++++++++++++++++++++++++-----
1 file changed, 45 insertions(+), 5 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 292c8c9ff4c0..df541a72d3a9 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -339,17 +339,59 @@ 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 pa_buffer_attr *attr,
int *rerror)
{
int r;
- pa_stream *stream;
+ pa_stream *stream = NULL;
pa_stream_flags_t flags;
+ pa_channel_map map;
pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new(c->context, name, ss, map);
+ pa_channel_map_init(&map);
+ map.channels = ss->channels;
+
+ /*
+ * TODO: This currently expects the only frontend supporting more than 2
+ * channels is the usb-audio. We will need some means to set channel
+ * order when a new frontend gains multi-channel support.
+ */
+ switch (ss->channels) {
+ case 1:
+ map.map[0] = PA_CHANNEL_POSITION_MONO;
+ break;
+
+ case 2:
+ map.map[0] = PA_CHANNEL_POSITION_LEFT;
+ map.map[1] = PA_CHANNEL_POSITION_RIGHT;
+ break;
+
+ case 6:
+ map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
+ map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
+ map.map[2] = PA_CHANNEL_POSITION_CENTER;
+ map.map[3] = PA_CHANNEL_POSITION_LFE;
+ map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
+ map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
+ break;
+
+ case 8:
+ map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
+ map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
+ map.map[2] = PA_CHANNEL_POSITION_CENTER;
+ map.map[3] = PA_CHANNEL_POSITION_LFE;
+ map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
+ map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
+ map.map[6] = PA_CHANNEL_POSITION_SIDE_LEFT;
+ map.map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT;
+
+ default:
+ dolog("Internal error: unsupported channel count %d\n", ss->channels);
+ goto fail;
+ }
+
+ stream = pa_stream_new(c->context, name, ss, &map);
if (!stream) {
goto fail;
}
@@ -422,7 +464,6 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
&ba, /* buffering attributes */
&error
);
@@ -471,7 +512,6 @@ 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 */
&ba, /* buffering attributes */
&error
);
--
2.18.1
^ permalink raw reply related [flat|nested] 15+ messages in thread* Re: [PULL 00/13] Audio 20191018 patches
2019-10-18 7:41 [PULL 00/13] Audio 20191018 patches Gerd Hoffmann
` (12 preceding siblings ...)
2019-10-18 7:41 ` [PULL 13/13] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams Gerd Hoffmann
@ 2019-10-18 15:36 ` Peter Maydell
13 siblings, 0 replies; 15+ messages in thread
From: Peter Maydell @ 2019-10-18 15:36 UTC (permalink / raw)
To: Gerd Hoffmann; +Cc: QEMU Developers, Markus Armbruster
On Fri, 18 Oct 2019 at 08:43, Gerd Hoffmann <kraxel@redhat.com> wrote:
>
> The following changes since commit f22f553efffd083ff624be116726f843a39f1148:
>
> Merge remote-tracking branch 'remotes/rth/tags/pull-tcg-20191013' into staging (2019-10-17 16:48:56 +0100)
>
> are available in the Git repository at:
>
> git://git.kraxel.org/qemu tags/audio-20191018-pull-request
>
> for you to fetch changes up to 0cf13e367a99dd1abefc46ec94b4c1a80c678f61:
>
> paaudio: fix channel order for usb-audio 5.1 and 7.1 streams (2019-10-18 08:14:05 +0200)
>
> ----------------------------------------------------------------
> audio: bugfixes, pa connection and stream naming.
> audio: 5.1/7.1 support for alsa, pa and usb-audio.
>
Applied, thanks.
Please update the changelog at https://wiki.qemu.org/ChangeLog/4.2
for any user-visible changes.
-- PMM
^ permalink raw reply [flat|nested] 15+ messages in thread