qemu-e2k/audio/coreaudio.m

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/*
* QEMU OS X CoreAudio audio driver
*
* Copyright (c) 2005 Mike Kronenberg
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "qemu/osdep.h"
#include <CoreAudio/CoreAudio.h>
#include <pthread.h> /* pthread_X */
#include "qemu/main-loop.h"
#include "qemu/module.h"
#include "audio.h"
#define AUDIO_CAP "coreaudio"
#include "audio_int.h"
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t buf_mutex;
AudioDeviceID outputDeviceID;
int frameSizeSetting;
uint32_t bufferCount;
UInt32 audioDevicePropertyBufferFrameSize;
AudioDeviceIOProcID ioprocid;
bool enabled;
} coreaudioVoiceOut;
#if !defined(MAC_OS_VERSION_12_0) \
|| (MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_VERSION_12_0)
#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster
#endif
static const AudioObjectPropertyAddress voice_addr = {
kAudioHardwarePropertyDefaultOutputDevice,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMain
};
static OSStatus coreaudio_get_voice(AudioDeviceID *id)
{
UInt32 size = sizeof(*id);
return AudioObjectGetPropertyData(kAudioObjectSystemObject,
&voice_addr,
0,
NULL,
&size,
id);
}
static OSStatus coreaudio_get_framesizerange(AudioDeviceID id,
AudioValueRange *framerange)
{
UInt32 size = sizeof(*framerange);
AudioObjectPropertyAddress addr = {
kAudioDevicePropertyBufferFrameSizeRange,
kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMain
};
return AudioObjectGetPropertyData(id,
&addr,
0,
NULL,
&size,
framerange);
}
static OSStatus coreaudio_get_framesize(AudioDeviceID id, UInt32 *framesize)
{
UInt32 size = sizeof(*framesize);
AudioObjectPropertyAddress addr = {
kAudioDevicePropertyBufferFrameSize,
kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMain
};
return AudioObjectGetPropertyData(id,
&addr,
0,
NULL,
&size,
framesize);
}
static OSStatus coreaudio_set_framesize(AudioDeviceID id, UInt32 *framesize)
{
UInt32 size = sizeof(*framesize);
AudioObjectPropertyAddress addr = {
kAudioDevicePropertyBufferFrameSize,
kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMain
};
return AudioObjectSetPropertyData(id,
&addr,
0,
NULL,
size,
framesize);
}
static OSStatus coreaudio_set_streamformat(AudioDeviceID id,
AudioStreamBasicDescription *d)
{
UInt32 size = sizeof(*d);
AudioObjectPropertyAddress addr = {
kAudioDevicePropertyStreamFormat,
kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMain
};
return AudioObjectSetPropertyData(id,
&addr,
0,
NULL,
size,
d);
}
static OSStatus coreaudio_get_isrunning(AudioDeviceID id, UInt32 *result)
{
UInt32 size = sizeof(*result);
AudioObjectPropertyAddress addr = {
kAudioDevicePropertyDeviceIsRunning,
kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMain
};
return AudioObjectGetPropertyData(id,
&addr,
0,
NULL,
&size,
result);
}
static void coreaudio_logstatus (OSStatus status)
{
const char *str = "BUG";
switch (status) {
case kAudioHardwareNoError:
str = "kAudioHardwareNoError";
break;
case kAudioHardwareNotRunningError:
str = "kAudioHardwareNotRunningError";
break;
case kAudioHardwareUnspecifiedError:
str = "kAudioHardwareUnspecifiedError";
break;
case kAudioHardwareUnknownPropertyError:
str = "kAudioHardwareUnknownPropertyError";
break;
case kAudioHardwareBadPropertySizeError:
str = "kAudioHardwareBadPropertySizeError";
break;
case kAudioHardwareIllegalOperationError:
str = "kAudioHardwareIllegalOperationError";
break;
case kAudioHardwareBadDeviceError:
str = "kAudioHardwareBadDeviceError";
break;
case kAudioHardwareBadStreamError:
str = "kAudioHardwareBadStreamError";
break;
case kAudioHardwareUnsupportedOperationError:
str = "kAudioHardwareUnsupportedOperationError";
break;
case kAudioDeviceUnsupportedFormatError:
str = "kAudioDeviceUnsupportedFormatError";
break;
case kAudioDevicePermissionsError:
str = "kAudioDevicePermissionsError";
break;
default:
AUD_log (AUDIO_CAP, "Reason: status code %" PRId32 "\n", (int32_t)status);
return;
}
AUD_log (AUDIO_CAP, "Reason: %s\n", str);
}
static void G_GNUC_PRINTF (2, 3) coreaudio_logerr (
OSStatus status,
const char *fmt,
...
)
{
va_list ap;
va_start (ap, fmt);
AUD_log (AUDIO_CAP, fmt, ap);
va_end (ap);
coreaudio_logstatus (status);
}
static void G_GNUC_PRINTF (3, 4) coreaudio_logerr2 (
OSStatus status,
const char *typ,
const char *fmt,
...
)
{
va_list ap;
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
coreaudio_logstatus (status);
}
#define coreaudio_playback_logerr(status, ...) \
coreaudio_logerr2(status, "playback", __VA_ARGS__)
static int coreaudio_buf_lock (coreaudioVoiceOut *core, const char *fn_name)
{
int err;
err = pthread_mutex_lock (&core->buf_mutex);
if (err) {
dolog ("Could not lock voice for %s\nReason: %s\n",
fn_name, strerror (err));
return -1;
}
return 0;
}
static int coreaudio_buf_unlock (coreaudioVoiceOut *core, const char *fn_name)
{
int err;
err = pthread_mutex_unlock (&core->buf_mutex);
if (err) {
dolog ("Could not unlock voice for %s\nReason: %s\n",
fn_name, strerror (err));
return -1;
}
return 0;
}
#define COREAUDIO_WRAPPER_FUNC(name, ret_type, args_decl, args) \
static ret_type glue(coreaudio_, name)args_decl \
{ \
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw; \
ret_type ret; \
\
if (coreaudio_buf_lock(core, "coreaudio_" #name)) { \
return 0; \
} \
\
ret = glue(audio_generic_, name)args; \
\
coreaudio_buf_unlock(core, "coreaudio_" #name); \
return ret; \
}
COREAUDIO_WRAPPER_FUNC(buffer_get_free, size_t, (HWVoiceOut *hw), (hw))
COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
(hw, size))
audio: fix bug 1858488 The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-23 08:49:39 +01:00
COREAUDIO_WRAPPER_FUNC(put_buffer_out, size_t,
(HWVoiceOut *hw, void *buf, size_t size),
(hw, buf, size))
COREAUDIO_WRAPPER_FUNC(write, size_t, (HWVoiceOut *hw, void *buf, size_t size),
(hw, buf, size))
#undef COREAUDIO_WRAPPER_FUNC
/*
* callback to feed audiooutput buffer. called without iothread lock.
* allowed to lock "buf_mutex", but disallowed to have any other locks.
*/
static OSStatus audioDeviceIOProc(
AudioDeviceID inDevice,
const AudioTimeStamp *inNow,
const AudioBufferList *inInputData,
const AudioTimeStamp *inInputTime,
AudioBufferList *outOutputData,
const AudioTimeStamp *inOutputTime,
void *hwptr)
{
UInt32 frameCount, pending_frames;
void *out = outOutputData->mBuffers[0].mData;
HWVoiceOut *hw = hwptr;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
size_t len;
if (coreaudio_buf_lock (core, "audioDeviceIOProc")) {
inInputTime = 0;
return 0;
}
if (inDevice != core->outputDeviceID) {
coreaudio_buf_unlock (core, "audioDeviceIOProc(old device)");
return 0;
}
frameCount = core->audioDevicePropertyBufferFrameSize;
pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
/* if there are not enough samples, set signal and return */
if (pending_frames < frameCount) {
inInputTime = 0;
coreaudio_buf_unlock (core, "audioDeviceIOProc(empty)");
return 0;
}
len = frameCount * hw->info.bytes_per_frame;
while (len) {
size_t write_len, start;
start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
assert(start < hw->size_emul);
write_len = MIN(MIN(hw->pending_emul, len),
hw->size_emul - start);
memcpy(out, hw->buf_emul + start, write_len);
hw->pending_emul -= write_len;
len -= write_len;
out += write_len;
}
coreaudio_buf_unlock (core, "audioDeviceIOProc");
return 0;
}
static OSStatus init_out_device(coreaudioVoiceOut *core)
{
OSStatus status;
AudioValueRange frameRange;
AudioStreamBasicDescription streamBasicDescription = {
.mBitsPerChannel = core->hw.info.bits,
.mBytesPerFrame = core->hw.info.bytes_per_frame,
.mBytesPerPacket = core->hw.info.bytes_per_frame,
.mChannelsPerFrame = core->hw.info.nchannels,
.mFormatFlags = kLinearPCMFormatFlagIsFloat,
.mFormatID = kAudioFormatLinearPCM,
.mFramesPerPacket = 1,
.mSampleRate = core->hw.info.freq
};
status = coreaudio_get_voice(&core->outputDeviceID);
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not get default output Device\n");
return status;
}
if (core->outputDeviceID == kAudioDeviceUnknown) {
dolog ("Could not initialize playback - Unknown Audiodevice\n");
return status;
}
/* get minimum and maximum buffer frame sizes */
status = coreaudio_get_framesizerange(core->outputDeviceID,
&frameRange);
if (status == kAudioHardwareBadObjectError) {
return 0;
}
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not get device buffer frame range\n");
return status;
}
if (frameRange.mMinimum > core->frameSizeSetting) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
} else if (frameRange.mMaximum < core->frameSizeSetting) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
} else {
core->audioDevicePropertyBufferFrameSize = core->frameSizeSetting;
}
/* set Buffer Frame Size */
status = coreaudio_set_framesize(core->outputDeviceID,
&core->audioDevicePropertyBufferFrameSize);
if (status == kAudioHardwareBadObjectError) {
return 0;
}
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not set device buffer frame size %" PRIu32 "\n",
(uint32_t)core->audioDevicePropertyBufferFrameSize);
return status;
}
/* get Buffer Frame Size */
status = coreaudio_get_framesize(core->outputDeviceID,
&core->audioDevicePropertyBufferFrameSize);
if (status == kAudioHardwareBadObjectError) {
return 0;
}
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not get device buffer frame size\n");
return status;
}
core->hw.samples = core->bufferCount * core->audioDevicePropertyBufferFrameSize;
/* set Samplerate */
status = coreaudio_set_streamformat(core->outputDeviceID,
&streamBasicDescription);
if (status == kAudioHardwareBadObjectError) {
return 0;
}
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not set samplerate %lf\n",
streamBasicDescription.mSampleRate);
core->outputDeviceID = kAudioDeviceUnknown;
return status;
}
/*
* set Callback.
*
* On macOS 11.3.1, Core Audio calls AudioDeviceIOProc after calling an
* internal function named HALB_Mutex::Lock(), which locks a mutex in
* HALB_IOThread::Entry(void*). HALB_Mutex::Lock() is also called in
* AudioObjectGetPropertyData, which is called by coreaudio driver.
* Therefore, the specified callback must be designed to avoid a deadlock
* with the callers of AudioObjectGetPropertyData.
*/
core->ioprocid = NULL;
status = AudioDeviceCreateIOProcID(core->outputDeviceID,
audioDeviceIOProc,
&core->hw,
&core->ioprocid);
if (status == kAudioHardwareBadDeviceError) {
return 0;
}
if (status != kAudioHardwareNoError || core->ioprocid == NULL) {
coreaudio_playback_logerr (status, "Could not set IOProc\n");
core->outputDeviceID = kAudioDeviceUnknown;
return status;
}
return 0;
}
static void fini_out_device(coreaudioVoiceOut *core)
{
OSStatus status;
UInt32 isrunning;
/* stop playback */
status = coreaudio_get_isrunning(core->outputDeviceID, &isrunning);
if (status != kAudioHardwareBadObjectError) {
if (status != kAudioHardwareNoError) {
coreaudio_logerr(status,
"Could not determine whether Device is playing\n");
}
if (isrunning) {
status = AudioDeviceStop(core->outputDeviceID, core->ioprocid);
if (status != kAudioHardwareBadDeviceError && status != kAudioHardwareNoError) {
coreaudio_logerr(status, "Could not stop playback\n");
}
}
}
/* remove callback */
status = AudioDeviceDestroyIOProcID(core->outputDeviceID,
core->ioprocid);
if (status != kAudioHardwareBadDeviceError && status != kAudioHardwareNoError) {
coreaudio_logerr(status, "Could not remove IOProc\n");
}
core->outputDeviceID = kAudioDeviceUnknown;
}
static void update_device_playback_state(coreaudioVoiceOut *core)
{
OSStatus status;
UInt32 isrunning;
status = coreaudio_get_isrunning(core->outputDeviceID, &isrunning);
if (status != kAudioHardwareNoError) {
if (status != kAudioHardwareBadObjectError) {
coreaudio_logerr(status,
"Could not determine whether Device is playing\n");
}
return;
}
if (core->enabled) {
/* start playback */
if (!isrunning) {
status = AudioDeviceStart(core->outputDeviceID, core->ioprocid);
if (status != kAudioHardwareBadDeviceError && status != kAudioHardwareNoError) {
coreaudio_logerr (status, "Could not resume playback\n");
}
}
} else {
/* stop playback */
if (isrunning) {
status = AudioDeviceStop(core->outputDeviceID,
core->ioprocid);
if (status != kAudioHardwareBadDeviceError && status != kAudioHardwareNoError) {
coreaudio_logerr(status, "Could not pause playback\n");
}
}
}
}
/* called without iothread lock. */
static OSStatus handle_voice_change(
AudioObjectID in_object_id,
UInt32 in_number_addresses,
const AudioObjectPropertyAddress *in_addresses,
void *in_client_data)
{
coreaudioVoiceOut *core = in_client_data;
qemu_mutex_lock_iothread();
if (core->outputDeviceID) {
fini_out_device(core);
}
if (!init_out_device(core)) {
update_device_playback_state(core);
}
qemu_mutex_unlock_iothread();
return 0;
}
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
int err;
Audiodev *dev = drv_opaque;
AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out;
struct audsettings obt_as;
/* create mutex */
err = pthread_mutex_init(&core->buf_mutex, NULL);
if (err) {
dolog("Could not create mutex\nReason: %s\n", strerror (err));
return -1;
}
obt_as = *as;
as = &obt_as;
as->fmt = AUDIO_FORMAT_F32;
audio_pcm_init_info (&hw->info, as);
core->frameSizeSetting = audio_buffer_frames(
qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610);
core->bufferCount = cpdo->has_buffer_count ? cpdo->buffer_count : 4;
status = AudioObjectAddPropertyListener(kAudioObjectSystemObject,
&voice_addr, handle_voice_change,
core);
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr (status,
"Could not listen to voice property change\n");
return -1;
}
if (init_out_device(core)) {
status = AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
&voice_addr,
handle_voice_change,
core);
if (status != kAudioHardwareNoError) {
coreaudio_playback_logerr(status,
"Could not remove voice property change listener\n");
}
return -1;
}
return 0;
}
static void coreaudio_fini_out (HWVoiceOut *hw)
{
OSStatus status;
int err;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
status = AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
&voice_addr,
handle_voice_change,
core);
if (status != kAudioHardwareNoError) {
coreaudio_logerr(status, "Could not remove voice property change listener\n");
}
fini_out_device(core);
/* destroy mutex */
err = pthread_mutex_destroy(&core->buf_mutex);
if (err) {
dolog("Could not destroy mutex\nReason: %s\n", strerror (err));
}
}
static void coreaudio_enable_out(HWVoiceOut *hw, bool enable)
{
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
core->enabled = enable;
update_device_playback_state(core);
}
static void *coreaudio_audio_init(Audiodev *dev, Error **errp)
{
return dev;
}
static void coreaudio_audio_fini (void *opaque)
{
}
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
audio: fix bug 1858488 The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-23 08:49:39 +01:00
/* wrapper for audio_generic_write */
.write = coreaudio_write,
/* wrapper for audio_generic_buffer_get_free */
.buffer_get_free = coreaudio_buffer_get_free,
audio: fix bug 1858488 The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-23 08:49:39 +01:00
/* wrapper for audio_generic_get_buffer_out */
.get_buffer_out = coreaudio_get_buffer_out,
audio: fix bug 1858488 The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-23 08:49:39 +01:00
/* wrapper for audio_generic_put_buffer_out */
.put_buffer_out = coreaudio_put_buffer_out,
.enable_out = coreaudio_enable_out
};
static struct audio_driver coreaudio_audio_driver = {
.name = "coreaudio",
.descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
.init = coreaudio_audio_init,
.fini = coreaudio_audio_fini,
.pcm_ops = &coreaudio_pcm_ops,
.max_voices_out = 1,
.max_voices_in = 0,
.voice_size_out = sizeof (coreaudioVoiceOut),
.voice_size_in = 0
};
static void register_audio_coreaudio(void)
{
audio_driver_register(&coreaudio_audio_driver);
}
type_init(register_audio_coreaudio);