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[Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/
From: |
Kővágó, Zoltán |
Subject: |
[Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/out pcm_ops |
Date: |
Sun, 23 Dec 2018 21:52:16 +0100 |
This patch removes the samples member from HWVoiceIn and HWVoiceOut.
Backends can specify buffer size via the newly added buffer_size_in and
buffer_size_out functions in audio_pcm_ops. They are optional, if not
defined qemu will fall back to some built-in constant.
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
Not sure if this is necessary. At first it seemed like a good idea to
have a function so that backends can compute the size on demand when
needed and things like that, but currently it's just a burden. The only
good feature is that it allows a backend to not define a function and
let the audio subsystem choose a default value, but the same could be
achieved by specifying that hw->samples = 0 means use a default value.
So if you guys agree, I'll remove this patch. Maybe add an -audiodev
parameter to change it, overriding whatever the backends supplies.
---
audio/alsaaudio.c | 20 ++++++++++++++++++--
audio/audio.c | 2 --
audio/audio_int.h | 5 +++--
audio/audio_template.h | 24 +++++++++++++++---------
audio/coreaudio.c | 10 +++++++++-
audio/dsound_template.h | 8 +++++++-
audio/dsoundaudio.c | 4 ++++
audio/noaudio.c | 2 --
audio/ossaudio.c | 22 +++++++++++++++++++---
audio/paaudio.c | 21 +++++++++++++++++----
audio/sdlaudio.c | 10 +++++++++-
audio/spiceaudio.c | 14 ++++++++++++--
audio/wavaudio.c | 1 -
13 files changed, 113 insertions(+), 30 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 56271b1174..672803e5c2 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -45,6 +45,7 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceOut;
@@ -52,6 +53,7 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -696,7 +698,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as,
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ alsa->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
@@ -704,6 +706,12 @@ static int alsa_init_out(HWVoiceOut *hw, struct
audsettings *as,
return 0;
}
+static size_t alsa_buffer_size_out(HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ return alsa->samples;
+}
+
#define VOICE_CTL_PAUSE 0
#define VOICE_CTL_PREPARE 1
#define VOICE_CTL_START 2
@@ -790,7 +798,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ alsa->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
@@ -798,6 +806,12 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
return 0;
}
+static size_t alsa_buffer_size_in(HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ return alsa->samples;
+}
+
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
@@ -915,11 +929,13 @@ static void alsa_audio_fini (void *opaque)
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
+ .buffer_size_out = alsa_buffer_size_out,
.write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
+ .buffer_size_in = alsa_buffer_size_in,
.read = alsa_read,
.ctl_in = alsa_ctl_in,
};
diff --git a/audio/audio.c b/audio/audio.c
index f195d8eb95..7db183b357 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1742,8 +1742,6 @@ CaptureVoiceOut *AUD_add_capture(
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
- /* XXX find a more elegant way */
- hw->samples = 4096 * 4;
audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index a5add3c2b8..598038d999 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -70,7 +70,6 @@ typedef struct HWVoiceOut {
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
- size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
int ctl_caps;
@@ -93,7 +92,6 @@ typedef struct HWVoiceIn {
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
- size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
int ctl_caps;
struct audio_pcm_ops *pcm_ops;
@@ -155,6 +153,8 @@ struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
+ /* get the optimal buffer size in samples; optional */
+ size_t (*buffer_size_out)(HWVoiceOut *hw);
/*
* get a buffer that after later can be passed to put_buffer_out; optional
* returns the buffer, and writes it's size to size (in bytes)
@@ -172,6 +172,7 @@ struct audio_pcm_ops {
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
+ size_t (*buffer_size_in)(HWVoiceIn *hw);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 83ffc62183..07ce9ce51f 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,7 +78,20 @@ 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;
+ size_t samples;
+ if (!hw->pcm_ops) {
+ /*
+ * We should only end up here when using wavcapture hmp command (and
not
+ * the wavcapture audio backend).
+ * It needs a lot of samples, otherwise you'll end up with "Could not
+ * mix X bytes into a capture buffer" warnings and a garbled capture.
+ */
+ samples = 4096 * 4;
+ } else if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
if (audio_bug(__func__, samples == 0)) {
dolog("Attempted to allocate empty buffer\n");
}
@@ -264,11 +277,6 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
goto err0;
}
- if (audio_bug(__func__, hw->samples <= 0)) {
- dolog("hw->samples=%zd\n", hw->samples);
- goto err1;
- }
-
#ifdef DAC
hw->clip = mixeng_clip
#else
@@ -288,9 +296,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
#endif
return hw;
- err1:
- glue (hw->pcm_ops->fini_, TYPE) (hw);
- err0:
+err0:
g_free (hw);
return NULL;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index a532e862dd..f4210d5784 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,6 +43,7 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
+ size_t samples;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -557,7 +558,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct
audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+ core->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
@@ -617,6 +618,12 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct
audsettings *as,
return 0;
}
+static size_t coreaudio_buffer_size_out(HWVoiceOut *hw)
+{
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+ return core->samples;
+}
+
static void coreaudio_fini_out (HWVoiceOut *hw)
{
OSStatus status;
@@ -693,6 +700,7 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
.write = coreaudio_write,
+ .buffer_size_out = coreaudio_buffer_size_out,
.get_buffer_out = coreaudio_get_buffer_out,
.put_buffer_out = coreaudio_put_buffer_out_nowrite,
.ctl_out = coreaudio_ctl_out
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index ff5a1f85fd..6a10b6751b 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -254,7 +254,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct
audsettings *as,
);
}
hw->size_emul = bc.dwBufferBytes;
- hw->samples = bc.dwBufferBytes >> hw->info.shift;
+ ds->samples = bc.dwBufferBytes >> hw->info.shift;
ds->s = s;
#ifdef DEBUG_DSOUND
@@ -268,6 +268,12 @@ static int dsound_init_out(HWVoiceOut *hw, struct
audsettings *as,
return -1;
}
+static size_t glue(dsound_buffer_size_, TYPE)(HWVOICE *hw)
+{
+ DSOUNDVOICE *ds = (DSOUNDVOICE *) hw;
+ return ds->samples;
+}
+
#undef NAME
#undef NAME2
#undef TYPE
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 7b3266aaf3..be6b8d8889 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -54,12 +54,14 @@ typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
dsound *s;
+ size_t samples;
} DSoundVoiceOut;
typedef struct {
HWVoiceIn hw;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
dsound *s;
+ size_t samples;
} DSoundVoiceIn;
static void dsound_log_hresult (HRESULT hr)
@@ -672,6 +674,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
.write = audio_generic_write,
+ .buffer_size_out = dsound_buffer_size_out,
.get_buffer_out = dsound_get_buffer_out,
.put_buffer_out = dsound_put_buffer_out,
.ctl_out = dsound_ctl_out,
@@ -679,6 +682,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
.read = audio_generic_read,
+ .buffer_size_in = dsound_buffer_size_in,
.get_buffer_in = dsound_get_buffer_in,
.put_buffer_in = dsound_put_buffer_in,
.ctl_in = dsound_ctl_in
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 6a3a1c418b..9b1dfb551d 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -58,7 +58,6 @@ static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void
*drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
- hw->samples = 1024;
return 0;
}
@@ -77,7 +76,6 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
- hw->samples = 1024;
return 0;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 08fe047f91..bc34e12de4 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -44,6 +44,7 @@ typedef struct OSSVoiceOut {
int nfrags;
int fragsize;
int mmapped;
+ size_t samples;
Audiodev *dev;
} OSSVoiceOut;
@@ -52,6 +53,7 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
+ size_t samples;
Audiodev *dev;
} OSSVoiceIn;
@@ -511,11 +513,11 @@ static int oss_init_out(HWVoiceOut *hw, struct
audsettings *as,
obt.nfrags * obt.fragsize, hw->info.align + 1);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = hw->samples << hw->info.shift;
+ hw->size_emul = oss->samples << hw->info.shift;
hw->buf_emul = mmap (
NULL,
hw->size_emul,
@@ -563,6 +565,12 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
return 0;
}
+static size_t oss_buffer_size_out(HWVoiceOut *hw)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ return oss->samples;
+}
+
static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
@@ -658,13 +666,19 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
obt.nfrags * obt.fragsize, hw->info.align + 1);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->fd = fd;
oss->dev = dev;
return 0;
}
+static size_t oss_buffer_size_in(HWVoiceIn *hw)
+{
+ OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ return oss->samples;
+}
+
static void oss_fini_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
@@ -753,6 +767,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.write = oss_write,
+ .buffer_size_out = oss_buffer_size_out,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
.ctl_out = oss_ctl_out,
@@ -760,6 +775,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.read = oss_read,
+ .buffer_size_in = oss_buffer_size_in,
.ctl_in = oss_ctl_in
};
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 392225c875..7cab3cff97 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -355,8 +355,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
- 46440);
+ pa->samples = audio_buffer_samples(g->dev->out, &obt_as, 46440);
return 0;
@@ -364,6 +363,13 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as,
return -1;
}
+static size_t qpa_buffer_size_out(HWVoiceOut *hw)
+{
+ PAVoiceOut *pa = (PAVoiceOut *) hw;
+ return pa->samples;
+}
+
+
static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
@@ -397,8 +403,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
- 46440);
+ pa->samples = audio_buffer_samples(g->dev->in, &obt_as, 46440);
return 0;
@@ -406,6 +411,12 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
return -1;
}
+static size_t qpa_buffer_size_in(HWVoiceIn *hw)
+{
+ PAVoiceIn *pa = (PAVoiceIn *) hw;
+ return pa->samples;
+}
+
static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
{
int err;
@@ -683,11 +694,13 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.write = qpa_write,
+ .buffer_size_out = qpa_buffer_size_out,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.read = qpa_read,
+ .buffer_size_in = qpa_buffer_size_in,
.ctl_in = qpa_ctl_in
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 685cbc83b8..4df35ce31a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -42,6 +42,7 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
+ size_t samples;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -363,7 +364,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings
*as,
obt_as.endianness = endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ sdl->samples = obt.samples;
s->initialized = 1;
s->exit = 0;
@@ -371,6 +372,12 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings
*as,
return 0;
}
+static size_t sdl_buffer_size_out(HWVoiceOut *hw)
+{
+ SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+ return sdl->samples;
+}
+
static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
@@ -439,6 +446,7 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
.write = sdl_write,
+ .buffer_size_out = sdl_buffer_size_out,
.get_buffer_out = sdl_get_buffer_out,
.put_buffer_out = sdl_put_buffer_out_nowrite,
.ctl_out = sdl_ctl_out,
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index d1605d3939..709245e453 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -133,7 +133,6 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings
*as,
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_OUT_SAMPLES;
out->active = 0;
out->sin.base.sif = &playback_sif.base;
@@ -144,6 +143,11 @@ static int line_out_init(HWVoiceOut *hw, struct
audsettings *as,
return 0;
}
+static size_t line_out_buffer_size(HWVoiceOut *hw)
+{
+ return LINE_OUT_SAMPLES;
+}
+
static void line_out_fini (HWVoiceOut *hw)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
@@ -248,7 +252,6 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_IN_SAMPLES;
in->active = 0;
in->sin.base.sif = &record_sif.base;
@@ -259,6 +262,11 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
return 0;
}
+static size_t line_in_buffer_size(HWVoiceIn *hw)
+{
+ return LINE_IN_SAMPLES;
+}
+
static void line_in_fini (HWVoiceIn *hw)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
@@ -329,6 +337,7 @@ static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
.write = audio_generic_write,
+ .buffer_size_out = line_out_buffer_size,
.get_buffer_out = line_out_get_buffer,
.put_buffer_out = line_out_put_buffer,
.ctl_out = line_out_ctl,
@@ -336,6 +345,7 @@ static struct audio_pcm_ops audio_callbacks = {
.init_in = line_in_init,
.fini_in = line_in_fini,
.read = line_in_read,
+ .buffer_size_in = line_in_buffer_size,
.ctl_in = line_in_ctl,
};
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 31db03aadb..0a0e76d2d9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -110,7 +110,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings
*as,
wav_as.endianness = 0;
audio_pcm_init_info (&hw->info, &wav_as);
- hw->samples = 1024;
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
--
2.20.1
- [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops, (continued)
- [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 28/52] audio: api for mixeng code free backends, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 36/52] spiceaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 32/52] noaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 31/52] dsoundaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 33/52] ossaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 39/52] audio: unify input and output mixeng buffer management, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 30/52] coreaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/out pcm_ops,
Kővágó, Zoltán <=
- [Qemu-devel] [PATCH v2 44/52] audio: make mixeng optional, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 46/52] audio: support more than two channels in volume setting, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 48/52] audio: basic support for multichannel audio, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 37/52] wavaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 35/52] sdlaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 38/52] audio: remove remains of the old backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 41/52] audio: common rate control code for timer based outputs, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 29/52] alsaaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 34/52] paaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 42/52] audio: split ctl_* functions into enable_* and volume_*, Kővágó, Zoltán, 2018/12/23