* [Qemu-devel] [PATCH v2 1/9] audio: add mixeng option (documentation)
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 2/9] audio: make mixeng optional Kővágó, Zoltán
` (6 subsequent siblings)
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
This will allow us to disable mixeng when we use a decent backend.
Disabling mixeng have a few advantages:
* we no longer convert the audio output from one format to another, when
the underlying audio system would just convert it to a third format.
We no longer convert, only the underlying system, when needed.
* the underlying system probably has better resampling and sample format
converting methods anyway...
* we may support formats that the mixeng currently does not support (S24
or float samples, more than two channels)
* when using an audio server (like pulseaudio) different sound card
outputs will show up as separate streams, even if we use only one
backend
Disadvantages:
* audio capturing no longer works (wavcapture, and vnc audio extension)
* some backends only support a single playback stream or very picky
about the audio format. In this case we can't disable mixeng.
However mixeng is not removed, only made optional, so this shouldn't be
a big concern.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
qapi/audio.json | 5 +++++
qemu-options.hx | 6 ++++++
2 files changed, 11 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index 30613da..2852dcc 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -184,6 +184,10 @@
#
# General audio backend options that are used for both playback and recording.
#
+# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
+# When set to off, fixed-settings must be also off. Not every backend
+# compatible with the off setting (default on)
+#
# @fixed-settings: #optional use fixed settings for host input/output. When
# off, frequency, channels and format must not be specified
# (default on)
@@ -207,6 +211,7 @@
##
{ 'struct': 'AudiodevPerDirectionOptions',
'data': {
+ '*mixeng': 'bool',
'*fixed-settings': 'bool',
'*frequency': 'int',
'*channels': 'int',
diff --git a/qemu-options.hx b/qemu-options.hx
index 09d19a7..4fffb6a 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -329,6 +329,7 @@ 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.mixeng= use mixeng 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"
@@ -397,6 +398,11 @@ Identifies the audio backend.
Sets the timer @var{period} used by the audio subsystem in microseconds.
Default is 10000 (10 ms).
+@item in|out.mixeng=on|off
+Use QEMU's mixing engine to mix all streams inside QEMU. When off,
+@var{fixed-settings} must be off too. Not every backend is fully
+compatible with the off setting. Default is on.
+
@item in|out.fixed-settings=on|off
Use fixed settings for host audio. When off, it will change based on
how the guest opens the sound card. In this case you must not specify
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 2/9] audio: make mixeng optional
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 1/9] audio: add mixeng option (documentation) Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 3/9] paaudio: get/put_buffer functions Kővágó, Zoltán
` (5 subsequent siblings)
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Implementation of the previously added mixeng option.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 70 +++++++++++++++++++++++++++++++++++++++++++++-----
audio/audio_template.h | 22 +++++++++++-----
2 files changed, 79 insertions(+), 13 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index cb1987b..e048e37 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -793,32 +793,46 @@ 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 (hw->s->dev->out->mixeng) {
+ 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 (hw->s->dev->in->mixeng) {
+ 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)
@@ -1038,6 +1052,26 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
+ if (!s->dev->out->mixeng) {
+ 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;
@@ -1174,6 +1208,17 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
+ if (!s->dev->in->mixeng) {
+ 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, min;
@@ -1654,6 +1699,11 @@ CaptureVoiceOut *AUD_add_capture(
s = QTAILQ_FIRST(&audio_states);
}
+ if (!s->dev->out->mixeng) {
+ dolog("Can't capture with mixeng disabled\n");
+ goto err0;
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1811,9 +1861,13 @@ QemuOptsList qemu_audiodev_opts = {
static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
Error **errp)
{
+ if (!pdo->has_mixeng) {
+ pdo->has_mixeng = true;
+ pdo->mixeng = true;
+ }
if (!pdo->has_fixed_settings) {
pdo->has_fixed_settings = true;
- pdo->fixed_settings = true;
+ pdo->fixed_settings = pdo->mixeng;
}
if (!pdo->fixed_settings &&
(pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
@@ -1821,6 +1875,10 @@ static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
"You can't use frequency, channels or format with fixed-settings=off");
return;
}
+ if (!pdo->mixeng && pdo->fixed_settings) {
+ error_setg(errp, "You can't use fixed-settings without mixeng");
+ return;
+ }
if (!pdo->has_frequency) {
pdo->has_frequency = true;
@@ -1832,7 +1890,7 @@ static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
}
if (!pdo->has_voices) {
pdo->has_voices = true;
- pdo->voices = 1;
+ pdo->voices = pdo->mixeng ? 1 : INT_MAX;
}
if (!pdo->has_format) {
pdo->has_format = true;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 762efea..f3cbc66 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,15 +78,19 @@ 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;
- if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
- samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ if (hw->s->dev->TYPE->mixeng) {
+ size_t samples;
+ if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
+
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample)*samples);
+ HWBUF->size = samples;
} else {
- samples = 1024; /* todo better default */
+ HWBUF = NULL;
}
-
- HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample)*samples);
- HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -105,6 +109,10 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
+ if (!sw->s->dev->TYPE->mixeng) {
+ return 0;
+ }
+
samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc (AUDIO_FUNC, samples, sizeof (struct st_sample));
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 3/9] paaudio: get/put_buffer functions
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 1/9] audio: add mixeng option (documentation) Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 2/9] audio: make mixeng optional Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 4/9] audio: support more than two channels in volume setting Kővágó, Zoltán
` (4 subsequent siblings)
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 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 14bf258..8f19995 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -96,6 +96,59 @@ 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;
@@ -134,6 +187,32 @@ 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;
@@ -668,12 +747,16 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.fini_out = qpa_fini_out,
.write = qpa_write,
.buffer_size_out = qpa_buffer_size_out,
+ .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,
.buffer_size_in = qpa_buffer_size_in,
+ .get_buffer_in = qpa_get_buffer_in,
+ .put_buffer_in = qpa_put_buffer_in,
.volume_in = qpa_volume_in
};
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 4/9] audio: support more than two channels in volume setting
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
` (2 preceding siblings ...)
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 3/9] paaudio: get/put_buffer functions Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 5/9] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
` (3 subsequent siblings)
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 28 ++++++++++++++++++++--------
audio/audio.h | 10 ++++++++++
audio/audio_int.h | 4 ++--
audio/paaudio.c | 20 ++++++++++++--------
audio/spiceaudio.c | 14 ++++++++------
5 files changed, 52 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e048e37..a33f574 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1817,30 +1817,42 @@ 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 a48411a..06f15e3 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -127,6 +127,16 @@ 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 6f2f309..8b53c52 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -158,7 +158,7 @@ struct audio_pcm_ops {
* size may be smaller */
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);
@@ -167,7 +167,7 @@ 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 8f19995..c7875da 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -534,20 +534,22 @@ 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);
@@ -574,20 +576,22 @@ 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 48ca4e6..833e76f 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -177,13 +177,14 @@ 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);
}
@@ -264,13 +265,14 @@ 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);
}
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 5/9] audio: replace shift in audio_pcm_info with bytes_per_frame
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
` (3 preceding siblings ...)
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 4/9] audio: support more than two channels in volume setting Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 6/9] audio: basic support for multichannel audio Kővágó, Zoltán
` (2 subsequent siblings)
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 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 | 10 +++----
audio/audio.c | 76 ++++++++++++++++++++++++-------------------------
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 | 5 ++--
audio/wavaudio.c | 6 ++--
10 files changed, 67 insertions(+), 67 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 44f28f4..d37722c 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -613,7 +613,7 @@ 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);
@@ -657,7 +657,7 @@ 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;
}
@@ -823,7 +823,7 @@ 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) {
@@ -849,8 +849,8 @@ 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 a33f574..4873683 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -267,26 +267,27 @@ 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:
sign = 1;
case AUDIO_FORMAT_U16:
bits = 16;
- shift = 1;
+ mul = 2;
break;
case AUDIO_FORMAT_S32:
sign = 1;
case AUDIO_FORMAT_U32:
bits = 32;
- shift = 2;
+ mul = 4;
break;
default:
@@ -297,9 +298,8 @@ 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);
}
@@ -310,26 +310,25 @@ 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;
}
}
@@ -339,14 +338,13 @@ 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;
}
}
@@ -529,7 +527,7 @@ 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);
@@ -578,7 +576,7 @@ 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;
}
@@ -613,7 +611,7 @@ 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;
}
/*
@@ -687,7 +685,7 @@ 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;
@@ -731,13 +729,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %d ret %d total sw %d\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
@@ -837,7 +835,7 @@ 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)
@@ -952,10 +950,10 @@ 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)
@@ -978,10 +976,11 @@ 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,
@@ -1000,7 +999,7 @@ 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;
@@ -1030,10 +1029,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
size_t size, decr, proc;
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
- 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;
@@ -1181,16 +1181,16 @@ 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);
@@ -1198,7 +1198,7 @@ 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;
@@ -1268,7 +1268,7 @@ 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;
@@ -1321,7 +1321,7 @@ 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;
@@ -1357,7 +1357,7 @@ 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;
@@ -1745,11 +1745,11 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc(AUDIO_FUNC, hw->mix_buf->size,
- 1 << hw->info.shift);
+ hw->info.bytes_per_frame);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
"(%zu samples, each %d bytes)\n",
- hw->mix_buf->size, 1 << hw->info.shift);
+ hw->mix_buf->size, hw->info.bytes_per_frame);
goto err3;
}
@@ -2058,14 +2058,14 @@ 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, get_ticks_per_sec());
- 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)", 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 8b53c52..0c0289d 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -42,8 +42,7 @@ 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 7fdf469..e5959a2 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -225,7 +225,7 @@ 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) {
@@ -234,7 +234,7 @@ 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 6a10b67..31d356d 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;
- ds->samples = bc.dwBufferBytes >> hw->info.shift;
+ ds->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 6fcd82e..285c729 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -321,8 +321,8 @@ 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 0618b60..32fbf64 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -86,7 +86,7 @@ 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 667bf4d..f884414 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -510,16 +510,16 @@ 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);
}
- oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = oss->samples << hw->info.shift;
+ hw->size_emul = oss->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap (
NULL,
hw->size_emul,
@@ -654,12 +654,12 @@ 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);
}
- oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->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 833e76f..f42d822 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,8 +131,9 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
out->fpos = 0;
}
- *size = audio_rate_get_bytes(&hw->info, &out->rate,
- (out->fsize - out->fpos) << hw->info.shift);
+ *size = audio_rate_get_bytes(
+ &hw->info, &out->rate,
+ (out->fsize - out->fpos) * hw->info.bytes_per_frame);
return out->frame + out->fpos;
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 5b35173..487a1bb 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -40,14 +40,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
- assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+ assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %zu bytes failed\nReaons: %s\n",
bytes, strerror(errno));
}
- wav->total_samples += bytes >> hw->info.shift;
+ wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@@ -130,7 +130,7 @@ 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) {
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 6/9] audio: basic support for multichannel audio
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
` (4 preceding siblings ...)
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 5/9] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 7/9] paaudio: channel-map option Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 8/9] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 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 d37722c..b86f741 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -499,13 +499,6 @@ 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 (pdo->buffer_count) {
if (pdo->buffer_len) {
int64_t req = pdo->buffer_len * pdo->buffer_count;
diff --git a/audio/audio.c b/audio/audio.c
index 4873683..83ac08c 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -210,7 +210,7 @@ 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) {
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 7/9] paaudio: channel-map option
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
` (5 preceding siblings ...)
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 6/9] audio: basic support for multichannel audio Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 8/9] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
Add an option to change the channel map used by pulseaudio. If not
specified, falls back to an OSS compatible channel map.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 18 ++++++++++++++----
qapi/audio.json | 5 ++++-
qemu-options.hx | 9 +++++++++
3 files changed, 27 insertions(+), 5 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index c7875da..52aa412 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -336,17 +336,27 @@ static pa_stream *qpa_simple_new (
pa_stream_direction_t dir,
const char *dev,
const pa_sample_spec *ss,
- const pa_channel_map *map,
+ const char *map,
const pa_buffer_attr *attr,
int *rerror)
{
int r;
pa_stream *stream;
pa_stream_flags_t flags;
+ pa_channel_map pa_map;
pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new(c->context, name, ss, map);
+ if (map && !pa_channel_map_parse(&pa_map, map)) {
+ dolog("Invalid channel map specified: '%s'\n", map);
+ map = NULL;
+ }
+ if (!map) {
+ pa_channel_map_init_extend(&pa_map, ss->channels,
+ PA_CHANNEL_MAP_OSS);
+ }
+
+ stream = pa_stream_new(c->context, name, ss, &pa_map);
if (!stream) {
goto fail;
}
@@ -423,7 +433,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
&ba, /* buffering attributes */
&error
);
@@ -471,7 +481,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
NULL, /* buffering attributes */
&error
);
diff --git a/qapi/audio.json b/qapi/audio.json
index 2852dcc..f683a55 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -119,11 +119,14 @@
#
# @name: #optional name of the sink/source to use
#
+# @channel-map: #optional channel map to use (default: OSS compatible map)
+#
# Since: 2.5
##
{ 'struct': 'AudiodevPaPerDirectionOptions',
'data': {
- '*name': 'str' } }
+ '*name': 'str',
+ '*channel-map': 'str' } }
##
# @AudiodevPaOptions
diff --git a/qemu-options.hx b/qemu-options.hx
index 4fffb6a..9d17a04 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -365,6 +365,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
"-audiodev pa,id=id[,prop[=value][,...]]\n"
" server= PulseAudio server address\n"
" sink|source.name= sink/source device name\n"
+ " sink|source.channel-map= channel map to use\n"
#endif
#ifdef CONFIG_SDL
"-audiodev sdl,id=id[,prop[=value][,...]]\n"
@@ -518,6 +519,14 @@ Sets the PulseAudio @var{server} to connect to.
@item sink|source.name=@var{sink}
Use the specified sink/source for playback/recording.
+@item sink|source.channel-map=@var{map}
+Use the specified channel map. The default is an OSS compatible
+channel map. Do not forget to escape commas inside the map:
+
+@example
+-audiodev pa,id=example,sink.channel-map=front-left,,front-right
+@end example
+
@end table
@item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread* [Qemu-devel] [PATCH v2 8/9] usb-audio: do not count on avail bytes actually available
2015-08-21 20:36 [Qemu-devel] [PATCH v2 0/9] audio: multi channel audio support Kővágó, Zoltán
` (6 preceding siblings ...)
2015-08-21 20:36 ` [Qemu-devel] [PATCH v2 7/9] paaudio: channel-map option Kővágó, Zoltán
@ 2015-08-21 20:36 ` Kővágó, Zoltán
7 siblings, 0 replies; 9+ messages in thread
From: Kővágó, Zoltán @ 2015-08-21 20:36 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 | 30 ++++++++++++++++++------------
1 file changed, 18 insertions(+), 12 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 48ac992..ddd80e0 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -317,27 +317,28 @@ 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;
}
- 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;
}
@@ -369,16 +370,21 @@ 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;
+ }
}
}
--
2.5.0
^ permalink raw reply related [flat|nested] 9+ messages in thread