* [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support
@ 2019-10-13 19:57 Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 01/10] audio: add mixing-engine option (documentation) Kővágó, Zoltán
` (9 more replies)
0 siblings, 10 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:57 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Hi,
I've updated my mixeng-free patches. This time it's only documentation
fixes in the first patch, otherwise it's identical to v5.
Regards,
Zoltan
Kővágó, Zoltán (10):
audio: add mixing-engine 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
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
paaudio: fix channel order for usb-audio 5.1 and 7.1 streams
audio/alsaaudio.c | 18 +-
audio/audio.c | 176 ++++++++++-----
audio/audio.h | 10 +
audio/audio_int.h | 7 +-
audio/audio_template.h | 24 ++-
audio/coreaudio.c | 4 +-
audio/dsound_template.h | 10 +-
audio/dsoundaudio.c | 4 +-
audio/noaudio.c | 2 +-
audio/ossaudio.c | 14 +-
audio/paaudio.c | 153 ++++++++++++--
audio/spiceaudio.c | 17 +-
audio/wavaudio.c | 6 +-
hw/usb/dev-audio.c | 459 ++++++++++++++++++++++++++++++++++------
qapi/audio.json | 6 +
qemu-options.hx | 15 ++
16 files changed, 740 insertions(+), 185 deletions(-)
--
2.23.0
^ permalink raw reply [flat|nested] 11+ messages in thread
* [PATCH v6 01/10] audio: add mixing-engine option (documentation)
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
@ 2019-10-13 19:57 ` Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 02/10] audio: make mixeng optional Kővágó, Zoltán
` (8 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:57 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
This will allow us to disable mixeng when we use a decent backend.
Disabling mixeng have a few advantages:
* we no longer convert the audio output from one format to another, when
the underlying audio system would just convert it to a third format.
We no longer convert, only the underlying system, when needed.
* the underlying system probably has better resampling and sample format
converting methods anyway...
* we may support formats that the mixeng currently does not support (S24
or float samples, more than two channels)
* when using an audio server (like pulseaudio) different sound card
outputs will show up as separate streams, even if we use only one
backend
Disadvantages:
* audio capturing no longer works (wavcapture, and vnc audio extension)
* some backends only support a single playback stream or very picky
about the audio format. In this case we can't disable mixeng.
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>
---
Notes:
Changes from v1:
* renamed mixeng to mixing-engine
Changes from v5:
* better documentation
Changes from v1:
* renamed mixeng to mixing-engine
qapi/audio.json | 6 ++++++
qemu-options.hx | 15 +++++++++++++++
2 files changed, 21 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index 9fefdf5186..23bbbe4b6a 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 2a04ca6ac5..3f2aecc99c 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
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 02/10] audio: make mixeng optional
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 01/10] audio: add mixing-engine option (documentation) Kővágó, Zoltán
@ 2019-10-13 19:57 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 03/10] paaudio: get/put_buffer functions Kővágó, Zoltán
` (7 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:57 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Implementation of the previously added mixing-engine option.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
Changes from v4:
* audio_pcm_hw_add_* always returns a new HW (or fails) when not using mixeng
audio/audio.c | 70 ++++++++++++++++++++++++++++++++++++++----
audio/audio_template.h | 24 ++++++++++-----
2 files changed, 80 insertions(+), 14 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 7128ee98dc..d616a4af98 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -838,32 +838,46 @@ static void audio_timer (void *opaque)
*/
size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
+ HWVoiceOut *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_write(sw, buf, size);
+ if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
+ return audio_pcm_sw_write(sw, buf, size);
+ } else {
+ return hw->pcm_ops->write(hw, buf, size);
+ }
}
size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
+ HWVoiceIn *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_read(sw, buf, size);
+ if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
+ return audio_pcm_sw_read(sw, buf, size);
+ } else {
+ return hw->pcm_ops->read(hw, buf, size);
+ }
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
@@ -1090,6 +1104,26 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
+ if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ sw = hw->sw_head.lh_first;
+
+ if (hw->pending_disable) {
+ hw->enabled = 0;
+ hw->pending_disable = 0;
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, false);
+ }
+ }
+
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
size_t played, live, prev_rpos, free;
int nb_live, cleanup_required;
@@ -1227,6 +1261,17 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
+ if (!audio_get_pdo_in(s->dev)->mixing_engine) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ SWVoiceIn *sw = hw->sw_head.lh_first;
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
size_t captured = 0, min;
@@ -1751,6 +1796,11 @@ CaptureVoiceOut *AUD_add_capture(
s = audio_init(NULL, NULL);
}
+ if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+ dolog("Can't capture with mixeng disabled\n");
+ return NULL;
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1905,9 +1955,13 @@ void audio_create_pdos(Audiodev *dev)
static void audio_validate_per_direction_opts(
AudiodevPerDirectionOptions *pdo, Error **errp)
{
+ if (!pdo->has_mixing_engine) {
+ pdo->has_mixing_engine = true;
+ pdo->mixing_engine = true;
+ }
if (!pdo->has_fixed_settings) {
pdo->has_fixed_settings = true;
- pdo->fixed_settings = true;
+ pdo->fixed_settings = pdo->mixing_engine;
}
if (!pdo->fixed_settings &&
(pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
@@ -1915,6 +1969,10 @@ static void audio_validate_per_direction_opts(
"You can't use frequency, channels or format with fixed-settings=off");
return;
}
+ if (!pdo->mixing_engine && pdo->fixed_settings) {
+ error_setg(errp, "You can't use fixed-settings without mixeng");
+ return;
+ }
if (!pdo->has_frequency) {
pdo->has_frequency = true;
@@ -1926,7 +1984,7 @@ static void audio_validate_per_direction_opts(
}
if (!pdo->has_voices) {
pdo->has_voices = true;
- pdo->voices = 1;
+ pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
}
if (!pdo->has_format) {
pdo->has_format = true;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 235d1acbbe..3709a4431c 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;
}
}
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 03/10] paaudio: get/put_buffer functions
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 01/10] audio: add mixing-engine option (documentation) Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 02/10] audio: make mixeng optional Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 04/10] audio: support more than two channels in volume setting Kővágó, Zoltán
` (6 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This lets us avoid some buffer copying when using mixeng.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 83 insertions(+)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index ed31f863f7..6ccdf31415 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
};
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 04/10] audio: support more than two channels in volume setting
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (2 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 03/10] paaudio: get/put_buffer functions Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 05/10] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
` (5 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 30 ++++++++++++++++++++++--------
audio/audio.h | 10 ++++++++++
audio/audio_int.h | 4 ++--
audio/paaudio.c | 20 ++++++++++++--------
audio/spiceaudio.c | 14 ++++++++------
5 files changed, 54 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index d616a4af98..f1c145dfcd 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1891,31 +1891,45 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_out(sw, &vol);
+}
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
+ 255;
if (hw->pcm_ops->volume_out) {
- hw->pcm_ops->volume_out(hw, &sw->vol);
+ hw->pcm_ops->volume_out(hw, vol);
}
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_in(sw, &vol);
+}
+
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
+ 255;
if (hw->pcm_ops->volume_in) {
- hw->pcm_ops->volume_in(hw, &sw->vol);
+ hw->pcm_ops->volume_in(hw, vol);
}
}
}
diff --git a/audio/audio.h b/audio/audio.h
index c74abb8c47..0db3c7dd5e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -124,6 +124,16 @@ uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol);
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol);
+#define AUDIO_MAX_CHANNELS 16
+typedef struct Volume {
+ bool mute;
+ int channels;
+ uint8_t vol[AUDIO_MAX_CHANNELS];
+} Volume;
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol);
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol);
+
SWVoiceIn *AUD_open_in (
QEMUSoundCard *card,
SWVoiceIn *sw,
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 22a703c13e..9176db249b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -166,7 +166,7 @@ struct audio_pcm_ops {
*/
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
void (*enable_out)(HWVoiceOut *hw, bool enable);
- void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
+ void (*volume_out)(HWVoiceOut *hw, Volume *vol);
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
@@ -174,7 +174,7 @@ struct audio_pcm_ops {
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
void (*enable_in)(HWVoiceIn *hw, bool enable);
- void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
+ void (*volume_in)(HWVoiceIn *hw, Volume *vol);
};
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6ccdf31415..d195b1caa8 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -531,20 +531,22 @@ static void qpa_fini_in (HWVoiceIn *hw)
}
}
-static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void qpa_volume_out(HWVoiceOut *hw, Volume *vol)
{
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
@@ -571,20 +573,22 @@ static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
pa_threaded_mainloop_unlock(c->mainloop);
}
-static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void qpa_volume_in(HWVoiceIn *hw, Volume *vol)
{
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 9860f9c5e1..6ed7f7a79e 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -179,13 +179,14 @@ static void line_out_enable(HWVoiceOut *hw, bool enable)
}
#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
-static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void line_out_volume(HWVoiceOut *hw, Volume *vol)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_playback_set_volume(&out->sin, 2, svol);
spice_server_playback_set_mute(&out->sin, vol->mute);
}
@@ -262,13 +263,14 @@ static void line_in_enable(HWVoiceIn *hw, bool enable)
}
#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
-static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void line_in_volume(HWVoiceIn *hw, Volume *vol)
{
SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_record_set_volume(&in->sin, 2, svol);
spice_server_record_set_mute(&in->sin, vol->mute);
}
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 05/10] audio: replace shift in audio_pcm_info with bytes_per_frame
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (3 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 04/10] audio: support more than two channels in volume setting Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 06/10] audio: basic support for multichannel audio Kővágó, Zoltán
` (4 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples). But if we want to add support for surround sound,
this no longer holds true.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 11 +++---
audio/audio.c | 74 ++++++++++++++++++++---------------------
audio/audio_int.h | 3 +-
audio/coreaudio.c | 4 +--
audio/dsound_template.h | 10 +++---
audio/dsoundaudio.c | 4 +--
audio/noaudio.c | 2 +-
audio/ossaudio.c | 14 ++++----
audio/spiceaudio.c | 3 +-
audio/wavaudio.c | 6 ++--
10 files changed, 66 insertions(+), 65 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index cfe42284a6..eddf013a53 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -602,7 +602,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
size_t pos = 0;
- size_t len_frames = len >> hw->info.shift;
+ size_t len_frames = len / hw->info.bytes_per_frame;
while (len_frames) {
char *src = advance(buf, pos);
@@ -648,7 +648,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
}
}
- pos += written << hw->info.shift;
+ pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
@@ -802,7 +802,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
- nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
+ nread = snd_pcm_readi(
+ alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
@@ -828,8 +829,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
}
}
- pos += nread << hw->info.shift;
- len -= nread << hw->info.shift;
+ pos += nread * hw->info.bytes_per_frame;
+ len -= nread * hw->info.bytes_per_frame;
}
return pos;
diff --git a/audio/audio.c b/audio/audio.c
index f1c145dfcd..c00f4deddd 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
- int bits = 8, sign = 0, shift = 0;
+ int bits = 8, sign = 0, mul;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
case AUDIO_FORMAT_U8:
+ mul = 1;
break;
case AUDIO_FORMAT_S16:
@@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
- shift = 1;
+ mul = 2;
break;
case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
- shift = 2;
+ mul = 4;
break;
default:
@@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
- info->shift = (as->nchannels == 2) + shift;
- info->align = (1 << info->shift) - 1;
- info->bytes_per_second = info->freq << info->shift;
+ info->bytes_per_frame = as->nchannels * mul;
+ info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
@@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
}
if (info->sign) {
- memset (buf, 0x00, len << info->shift);
+ memset(buf, 0x00, len * info->bytes_per_frame);
}
else {
switch (info->bits) {
case 8:
- memset (buf, 0x80, len << info->shift);
+ memset(buf, 0x80, len * info->bytes_per_frame);
break;
case 16:
{
int i;
uint16_t *p = buf;
- int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
int i;
uint32_t *p = buf;
- int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
- uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+ uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
@@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
return 0;
}
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
@@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
/*
@@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
- size >> sw->info.shift,
+ size / sw->info.bytes_per_frame,
ret,
sw->total_hw_samples_mixed
);
#endif
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@@ -882,7 +880,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->mix_buf->size << sw->hw->info.shift;
+ return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+ live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
);
- return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static size_t audio_get_free(SWVoiceOut *sw)
@@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+ live, dead, (((int64_t) dead << 32) / sw->ratio) *
+ sw->info.bytes_per_frame);
#endif
- return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1047,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
while (n) {
size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
- size_t bytes = to_write << hw->info.shift;
+ size_t bytes = to_write * hw->info.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
@@ -1082,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
return clipped + live;
}
- decr = MIN(size >> hw->info.shift, live);
+ decr = MIN(size / hw->info.bytes_per_frame, live);
audio_pcm_hw_clip_out(hw, buf, decr);
- proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
- hw->info.shift;
+ proc = hw->pcm_ops->put_buffer_out(hw, buf,
+ decr * hw->info.bytes_per_frame) /
+ hw->info.bytes_per_frame;
live -= proc;
clipped += proc;
@@ -1234,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
while (samples) {
size_t proc;
- size_t size = samples << hw->info.shift;
+ size_t size = samples * hw->info.bytes_per_frame;
void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
- assert((size & hw->info.align) == 0);
+ assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
hw->pcm_ops->put_buffer_in(hw, buf, size);
break;
}
- proc = MIN(size >> hw->info.shift,
+ proc = MIN(size / hw->info.bytes_per_frame,
conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1251,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
samples -= proc;
conv += proc;
- hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+ hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
@@ -1325,7 +1325,7 @@ static void audio_run_capture (AudioState *s)
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
- to_capture << hw->info.shift);
+ to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
@@ -1378,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->conv_buf->size << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1414,7 +1414,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->mix_buf->size << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1833,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
- cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
+ cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
@@ -2153,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
- samples = (bytes - rate->bytes_sent) >> info->shift;
+ samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
if (samples < 0 || samples > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
audio_rate_start(rate);
samples = 0;
}
- ret = MIN(samples << info->shift, bytes_avail);
+ ret = MIN(samples * info->bytes_per_frame, bytes_avail);
rate->bytes_sent += ret;
return ret;
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9176db249b..5ba2078346 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -43,8 +43,7 @@ struct audio_pcm_info {
int sign;
int freq;
int nchannels;
- int align;
- int shift;
+ int bytes_per_frame;
int bytes_per_second;
int swap_endianness;
};
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 1427c9f622..66f0f459cf 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
- pending_frames = hw->pending_emul >> hw->info.shift;
+ pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
/* if there are not enough samples, set signal and return */
if (pending_frames < frameCount) {
@@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc(
return 0;
}
- len = frameCount << hw->info.shift;
+ len = frameCount * hw->info.bytes_per_frame;
while (len) {
size_t write_len;
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 9f10b688df..7a15f91ce5 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
- if ((p1p && *p1p && (*blen1p & info->align)) ||
- (p2p && *p2p && (*blen2p & info->align))) {
+ if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
+ (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
dolog("DirectSound returned misaligned buffer %ld %ld\n",
*blen1p, *blen2p);
glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
- if (bc.dwBufferBytes & hw->info.align) {
+ if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
dolog (
"GetCaps returned misaligned buffer size %ld, alignment %d\n",
- bc.dwBufferBytes, hw->info.align + 1
+ bc.dwBufferBytes, hw->info.bytes_per_frame
);
}
hw->size_emul = bc.dwBufferBytes;
- hw->samples = bc.dwBufferBytes >> hw->info.shift;
+ hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
ds->s = s;
#ifdef DEBUG_DSOUND
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index d4a4757445..c265c0094b 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
return;
}
- len1 = blen1 >> hw->info.shift;
- len2 = blen2 >> hw->info.shift;
+ len1 = blen1 / hw->info.bytes_per_frame;
+ len2 = blen2 / hw->info.bytes_per_frame;
#ifdef DEBUG_DSOUND
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
diff --git a/audio/noaudio.c b/audio/noaudio.c
index ec8a287f36..ff99b253ff 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
NoVoiceIn *no = (NoVoiceIn *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
- audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+ audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
return bytes;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 0c4451e972..c43faeeea4 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = hw->samples << hw->info.shift;
+ hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap(
NULL,
hw->size_emul,
@@ -644,12 +644,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->fd = fd;
oss->dev = dev;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ed7f7a79e..b6b5da4812 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
if (out->frame) {
*size = audio_rate_get_bytes(
- &hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift);
+ &hw->info, &out->rate,
+ (out->fsize - out->fpos) * hw->info.bytes_per_frame);
} else {
audio_rate_start(&out->rate);
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 47efdc1b1e..e46d834bd3 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
- assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+ assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
bytes, strerror(errno));
}
- wav->total_samples += bytes >> hw->info.shift;
+ wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
uint8_t rlen[4];
uint8_t dlen[4];
- uint32_t datalen = wav->total_samples << hw->info.shift;
+ uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 06/10] audio: basic support for multichannel audio
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (4 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 05/10] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 07/10] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
` (3 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Which currently only means removing some checks. Old code won't require
more than two channels, but new code will need it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 7 -------
audio/audio.c | 2 +-
2 files changed, 1 insertion(+), 8 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index eddf013a53..f37ce1ce85 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -493,13 +493,6 @@ static int alsa_open(bool in, struct alsa_params_req *req,
goto err;
}
- if (nchannels != 1 && nchannels != 2) {
- alsa_logerr2 (err, typ,
- "Can not handle obtained number of channels %d\n",
- nchannels);
- goto err;
- }
-
if (apdo->buffer_length) {
int dir = 0;
unsigned int btime = apdo->buffer_length;
diff --git a/audio/audio.c b/audio/audio.c
index c00f4deddd..7fc3aa9d16 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -242,7 +242,7 @@ static int audio_validate_settings (struct audsettings *as)
{
int invalid;
- invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid = as->nchannels < 1;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 07/10] usb-audio: do not count on avail bytes actually available
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (5 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 06/10] audio: basic support for multichannel audio Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 08/10] usb-audio: support more than two channels of audio Kővágó, Zoltán
` (2 subsequent siblings)
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This assumption is no longer true when mixeng is turned off.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 28 +++++++++++++++++-----------
1 file changed, 17 insertions(+), 11 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index ae42e5a2f1..74c99b1f12 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -319,30 +319,31 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p)
{
uint32_t free = buf->size - (buf->prod - buf->cons);
- if (!free) {
+ if (free < USBAUDIO_PACKET_SIZE) {
return 0;
}
if (p->iov.size != USBAUDIO_PACKET_SIZE) {
return 0;
}
- assert(free >= USBAUDIO_PACKET_SIZE);
+
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE);
buf->prod += USBAUDIO_PACKET_SIZE;
return USBAUDIO_PACKET_SIZE;
}
-static uint8_t *streambuf_get(struct streambuf *buf)
+static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
uint32_t used = buf->prod - buf->cons;
uint8_t *data;
if (!used) {
+ *len = 0;
return NULL;
}
- assert(used >= USBAUDIO_PACKET_SIZE);
data = buf->data + (buf->cons % buf->size);
- buf->cons += USBAUDIO_PACKET_SIZE;
+ *len = MIN(buf->prod - buf->cons,
+ buf->size - (buf->cons % buf->size));
return data;
}
@@ -374,16 +375,21 @@ static void output_callback(void *opaque, int avail)
USBAudioState *s = opaque;
uint8_t *data;
- for (;;) {
- if (avail < USBAUDIO_PACKET_SIZE) {
+ while (avail) {
+ size_t written, len;
+
+ data = streambuf_get(&s->out.buf, &len);
+ if (!data) {
return;
}
- data = streambuf_get(&s->out.buf);
- if (!data) {
+
+ written = AUD_write(s->out.voice, data, len);
+ avail -= written;
+ s->out.buf.cons += written;
+
+ if (written < len) {
return;
}
- AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE);
- avail -= USBAUDIO_PACKET_SIZE;
}
}
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 08/10] usb-audio: support more than two channels of audio
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (6 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 07/10] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 09/10] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 10/10] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams Kővágó, Zoltán
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
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>
---
Notes:
According to the spec the channel order is front left, front right,
center, lfe, surround left, surround right for 5.1 sound. But alsa uses
front left, front right, surround left, surround right, center, lfe.
This means if you simply use an alsa device as -audiodev, the surround
and center/lfe channels will be swapped. There is not much to do sort
of writing a mini mixeng or something like that, but you can easily add
a new device to /etc/asound.conf that swaps the channels:
pcm.swap {
type route
slave.pcm "default" # or whatever
slave.channels 6
ttable.0.0 1
ttable.1.1 1
ttable.2.4 1
ttable.3.5 1
ttable.4.2 1
ttable.5.3 1
}
and use -audiodev alsa,id=foo,out.mixing-engine=off,alsa-out.dev=swap,...
(due to how usb and usb-audio works, you'll probably need
alsa-out.try-poll=off and some playing with threshold)
Pulseaudio allows us to specify channel order, and we default to a
compatible order so it should work correctly without extra setup.
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 74c99b1f12..e42bdfbdc1 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;
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 09/10] usbaudio: change playback counters to 64 bit
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (7 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 08/10] usb-audio: support more than two channels of audio Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 10/10] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams Kővágó, Zoltán
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
With stereo playback, they need about 375 minutes of continuous audio
playback to overflow, which is usually not a problem (as stopping and
later resuming playback resets the counters). But with 7.1 audio, they
only need about 95 minutes to overflow.
After the overflow, the buf->prod % USBAUDIO_PACKET_SIZE(channels)
assertion no longer holds true, which will result in overflowing the
buffer. With 64 bit variables, it would take about 762000 years to
overflow.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 14 ++++++++------
1 file changed, 8 insertions(+), 6 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index e42bdfbdc1..ea604bbb8e 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -578,9 +578,9 @@ static const USBDesc desc_audio_multi = {
struct streambuf {
uint8_t *data;
- uint32_t size;
- uint32_t prod;
- uint32_t cons;
+ size_t size;
+ uint64_t prod;
+ uint64_t cons;
};
static void streambuf_init(struct streambuf *buf, uint32_t size,
@@ -601,7 +601,7 @@ static void streambuf_fini(struct streambuf *buf)
static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
{
- uint32_t free = buf->size - (buf->prod - buf->cons);
+ int64_t free = buf->size - (buf->prod - buf->cons);
if (free < USBAUDIO_PACKET_SIZE(channels)) {
return 0;
@@ -610,6 +610,8 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
return 0;
}
+ /* can happen if prod overflows */
+ assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE(channels));
buf->prod += USBAUDIO_PACKET_SIZE(channels);
@@ -618,10 +620,10 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
- uint32_t used = buf->prod - buf->cons;
+ int64_t used = buf->prod - buf->cons;
uint8_t *data;
- if (!used) {
+ if (used <= 0) {
*len = 0;
return NULL;
}
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
* [PATCH v6 10/10] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
` (8 preceding siblings ...)
2019-10-13 19:58 ` [PATCH v6 09/10] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
@ 2019-10-13 19:58 ` Kővágó, Zoltán
9 siblings, 0 replies; 11+ messages in thread
From: Kővágó, Zoltán @ 2019-10-13 19:58 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 50 ++++++++++++++++++++++++++++++++++++++++++++-----
1 file changed, 45 insertions(+), 5 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index d195b1caa8..6ff0d17537 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -338,17 +338,59 @@ 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;
}
@@ -421,7 +463,6 @@ 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
);
@@ -470,7 +511,6 @@ 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
);
--
2.23.0
^ permalink raw reply related [flat|nested] 11+ messages in thread
end of thread, other threads:[~2019-10-13 20:09 UTC | newest]
Thread overview: 11+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2019-10-13 19:57 [PATCH v6 00/10] Audio: Mixeng-free 5.1/7.1 audio support Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 01/10] audio: add mixing-engine option (documentation) Kővágó, Zoltán
2019-10-13 19:57 ` [PATCH v6 02/10] audio: make mixeng optional Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 03/10] paaudio: get/put_buffer functions Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 04/10] audio: support more than two channels in volume setting Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 05/10] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 06/10] audio: basic support for multichannel audio Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 07/10] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 08/10] usb-audio: support more than two channels of audio Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 09/10] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
2019-10-13 19:58 ` [PATCH v6 10/10] paaudio: fix channel order for usb-audio 5.1 and 7.1 streams Kővágó, Zoltán
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