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Sweden-Number/dlls/winecoreaudio.drv/coreaudio.c

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/*
* Unixlib for winecoreaudio driver.
*
* Copyright 2011 Andrew Eikum for CodeWeavers
* Copyright 2021 Huw Davies
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
*/
#if 0
#pragma makedep unix
#endif
#include "config.h"
#define LoadResource __carbon_LoadResource
#define CompareString __carbon_CompareString
#define GetCurrentThread __carbon_GetCurrentThread
#define GetCurrentProcess __carbon_GetCurrentProcess
#include <stdarg.h>
#include <errno.h>
#include <limits.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <fenv.h>
#include <unistd.h>
#include <libkern/OSAtomic.h>
#include <CoreAudio/CoreAudio.h>
#include <AudioToolbox/AudioFormat.h>
#include <AudioToolbox/AudioConverter.h>
#include <AudioUnit/AudioUnit.h>
#undef LoadResource
#undef CompareString
#undef GetCurrentThread
#undef GetCurrentProcess
#undef _CDECL
#include "ntstatus.h"
#define WIN32_NO_STATUS
#include "windef.h"
#include "winbase.h"
#include "winnls.h"
#include "winreg.h"
#include "winternl.h"
#include "mmdeviceapi.h"
#include "initguid.h"
#include "audioclient.h"
#include "wine/debug.h"
#include "wine/unixlib.h"
#include "unixlib.h"
WINE_DEFAULT_DEBUG_CHANNEL(coreaudio);
struct coreaudio_stream
{
OSSpinLock lock;
AudioComponentInstance unit;
AudioConverterRef converter;
AudioStreamBasicDescription dev_desc; /* audio unit format, not necessarily the same as fmt */
AudioDeviceID dev_id;
EDataFlow flow;
AUDCLNT_SHAREMODE share;
BOOL playing;
UINT32 period_ms, period_frames;
UINT32 bufsize_frames, resamp_bufsize_frames;
UINT32 lcl_offs_frames, held_frames, wri_offs_frames, tmp_buffer_frames;
UINT32 cap_bufsize_frames, cap_offs_frames, cap_held_frames;
UINT32 wrap_bufsize_frames;
UINT64 written_frames;
INT32 getbuf_last;
WAVEFORMATEX *fmt;
BYTE *local_buffer, *cap_buffer, *wrap_buffer, *resamp_buffer, *tmp_buffer;
SIZE_T local_buffer_size, tmp_buffer_size;
};
static HRESULT osstatus_to_hresult(OSStatus sc)
{
switch(sc){
case kAudioFormatUnsupportedDataFormatError:
case kAudioFormatUnknownFormatError:
case kAudioDeviceUnsupportedFormatError:
return AUDCLNT_E_UNSUPPORTED_FORMAT;
case kAudioHardwareBadDeviceError:
return AUDCLNT_E_DEVICE_INVALIDATED;
}
return E_FAIL;
}
/* copied from kernelbase */
static int muldiv( int a, int b, int c )
{
LONGLONG ret;
if (!c) return -1;
/* We want to deal with a positive divisor to simplify the logic. */
if (c < 0)
{
a = -a;
c = -c;
}
/* If the result is positive, we "add" to round. else, we subtract to round. */
if ((a < 0 && b < 0) || (a >= 0 && b >= 0))
ret = (((LONGLONG)a * b) + (c / 2)) / c;
else
ret = (((LONGLONG)a * b) - (c / 2)) / c;
if (ret > 2147483647 || ret < -2147483647) return -1;
return ret;
}
static AudioObjectPropertyScope get_scope(EDataFlow flow)
{
return (flow == eRender) ? kAudioDevicePropertyScopeOutput : kAudioDevicePropertyScopeInput;
}
static BOOL device_has_channels(AudioDeviceID device, EDataFlow flow)
{
AudioObjectPropertyAddress addr;
AudioBufferList *buffers;
BOOL ret = FALSE;
OSStatus sc;
UInt32 size;
int i;
addr.mSelector = kAudioDevicePropertyStreamConfiguration;
addr.mScope = get_scope(flow);
addr.mElement = 0;
sc = AudioObjectGetPropertyDataSize(device, &addr, 0, NULL, &size);
if(sc != noErr){
WARN("Unable to get _StreamConfiguration property size for device %u: %x\n",
(unsigned int)device, (int)sc);
return FALSE;
}
buffers = malloc(size);
if(!buffers) return FALSE;
sc = AudioObjectGetPropertyData(device, &addr, 0, NULL, &size, buffers);
if(sc != noErr){
WARN("Unable to get _StreamConfiguration property for device %u: %x\n",
(unsigned int)device, (int)sc);
free(buffers);
return FALSE;
}
for(i = 0; i < buffers->mNumberBuffers; i++){
if(buffers->mBuffers[i].mNumberChannels > 0){
ret = TRUE;
break;
}
}
free(buffers);
return ret;
}
static NTSTATUS get_endpoint_ids(void *args)
{
struct get_endpoint_ids_params *params = args;
unsigned int num_devices, i, needed;
AudioDeviceID *devices, default_id;
AudioObjectPropertyAddress addr;
struct endpoint *endpoint;
UInt32 devsize, size;
struct endpoint_info
{
CFStringRef name;
AudioDeviceID id;
} *info;
OSStatus sc;
WCHAR *ptr;
params->num = 0;
params->default_idx = 0;
addr.mScope = kAudioObjectPropertyScopeGlobal;
addr.mElement = kAudioObjectPropertyElementMaster;
if(params->flow == eRender) addr.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
else if(params->flow == eCapture) addr.mSelector = kAudioHardwarePropertyDefaultInputDevice;
else{
params->result = E_INVALIDARG;
return STATUS_SUCCESS;
}
size = sizeof(default_id);
sc = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr, 0, NULL, &size, &default_id);
if(sc != noErr){
WARN("Getting _DefaultInputDevice property failed: %x\n", (int)sc);
default_id = -1;
}
addr.mSelector = kAudioHardwarePropertyDevices;
sc = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &addr, 0, NULL, &devsize);
if(sc != noErr){
WARN("Getting _Devices property size failed: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
num_devices = devsize / sizeof(AudioDeviceID);
devices = malloc(devsize);
info = malloc(num_devices * sizeof(*info));
if(!devices || !info){
free(info);
free(devices);
params->result = E_OUTOFMEMORY;
return STATUS_SUCCESS;
}
sc = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr, 0, NULL, &devsize, devices);
if(sc != noErr){
WARN("Getting _Devices property failed: %x\n", (int)sc);
free(info);
free(devices);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
addr.mSelector = kAudioObjectPropertyName;
addr.mScope = get_scope(params->flow);
addr.mElement = 0;
for(i = 0; i < num_devices; i++){
if(!device_has_channels(devices[i], params->flow)) continue;
size = sizeof(CFStringRef);
sc = AudioObjectGetPropertyData(devices[i], &addr, 0, NULL, &size, &info[params->num].name);
if(sc != noErr){
WARN("Unable to get _Name property for device %u: %x\n",
(unsigned int)devices[i], (int)sc);
continue;
}
info[params->num++].id = devices[i];
}
free(devices);
needed = sizeof(*endpoint) * params->num;
endpoint = params->endpoints;
ptr = (WCHAR *)(endpoint + params->num);
for(i = 0; i < params->num; i++){
SIZE_T len = CFStringGetLength(info[i].name);
needed += (len + 1) * sizeof(WCHAR);
if(needed <= params->size){
endpoint->name = ptr;
CFStringGetCharacters(info[i].name, CFRangeMake(0, len), (UniChar*)endpoint->name);
ptr[len] = 0;
endpoint->id = info[i].id;
endpoint++;
ptr += len + 1;
}
CFRelease(info[i].name);
if(info[i].id == default_id) params->default_idx = i;
}
free(info);
if(needed > params->size){
params->size = needed;
params->result = HRESULT_FROM_WIN32(ERROR_INSUFFICIENT_BUFFER);
}
else params->result = S_OK;
return STATUS_SUCCESS;
}
static WAVEFORMATEX *clone_format(const WAVEFORMATEX *fmt)
{
WAVEFORMATEX *ret;
size_t size;
if(fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
size = sizeof(WAVEFORMATEXTENSIBLE);
else
size = sizeof(WAVEFORMATEX);
ret = malloc(size);
if(!ret)
return NULL;
memcpy(ret, fmt, size);
ret->cbSize = size - sizeof(WAVEFORMATEX);
return ret;
}
static void silence_buffer(struct coreaudio_stream *stream, BYTE *buffer, UINT32 frames)
{
WAVEFORMATEXTENSIBLE *fmtex = (WAVEFORMATEXTENSIBLE*)stream->fmt;
if((stream->fmt->wFormatTag == WAVE_FORMAT_PCM ||
(stream->fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))) &&
stream->fmt->wBitsPerSample == 8)
memset(buffer, 128, frames * stream->fmt->nBlockAlign);
else
memset(buffer, 0, frames * stream->fmt->nBlockAlign);
}
/* CA is pulling data from us */
static OSStatus ca_render_cb(void *user, AudioUnitRenderActionFlags *flags,
const AudioTimeStamp *ts, UInt32 bus, UInt32 nframes,
AudioBufferList *data)
{
struct coreaudio_stream *stream = user;
UINT32 to_copy_bytes, to_copy_frames, chunk_bytes, lcl_offs_bytes;
OSSpinLockLock(&stream->lock);
if(stream->playing){
lcl_offs_bytes = stream->lcl_offs_frames * stream->fmt->nBlockAlign;
to_copy_frames = min(nframes, stream->held_frames);
to_copy_bytes = to_copy_frames * stream->fmt->nBlockAlign;
chunk_bytes = (stream->bufsize_frames - stream->lcl_offs_frames) * stream->fmt->nBlockAlign;
if(to_copy_bytes > chunk_bytes){
memcpy(data->mBuffers[0].mData, stream->local_buffer + lcl_offs_bytes, chunk_bytes);
memcpy(((BYTE *)data->mBuffers[0].mData) + chunk_bytes, stream->local_buffer, to_copy_bytes - chunk_bytes);
}else
memcpy(data->mBuffers[0].mData, stream->local_buffer + lcl_offs_bytes, to_copy_bytes);
stream->lcl_offs_frames += to_copy_frames;
stream->lcl_offs_frames %= stream->bufsize_frames;
stream->held_frames -= to_copy_frames;
}else
to_copy_bytes = to_copy_frames = 0;
if(nframes > to_copy_frames)
silence_buffer(stream, ((BYTE *)data->mBuffers[0].mData) + to_copy_bytes, nframes - to_copy_frames);
OSSpinLockUnlock(&stream->lock);
return noErr;
}
static void ca_wrap_buffer(BYTE *dst, UINT32 dst_offs, UINT32 dst_bytes,
BYTE *src, UINT32 src_bytes)
{
UINT32 chunk_bytes = dst_bytes - dst_offs;
if(chunk_bytes < src_bytes){
memcpy(dst + dst_offs, src, chunk_bytes);
memcpy(dst, src + chunk_bytes, src_bytes - chunk_bytes);
}else
memcpy(dst + dst_offs, src, src_bytes);
}
/* we need to trigger CA to pull data from the device and give it to us
*
* raw data from CA is stored in cap_buffer, possibly via wrap_buffer
*
* raw data is resampled from cap_buffer into resamp_buffer in period-size
* chunks and copied to local_buffer
*/
static OSStatus ca_capture_cb(void *user, AudioUnitRenderActionFlags *flags,
const AudioTimeStamp *ts, UInt32 bus, UInt32 nframes,
AudioBufferList *data)
{
struct coreaudio_stream *stream = user;
AudioBufferList list;
OSStatus sc;
UINT32 cap_wri_offs_frames;
OSSpinLockLock(&stream->lock);
cap_wri_offs_frames = (stream->cap_offs_frames + stream->cap_held_frames) % stream->cap_bufsize_frames;
list.mNumberBuffers = 1;
list.mBuffers[0].mNumberChannels = stream->fmt->nChannels;
list.mBuffers[0].mDataByteSize = nframes * stream->fmt->nBlockAlign;
if(!stream->playing || cap_wri_offs_frames + nframes > stream->cap_bufsize_frames){
if(stream->wrap_bufsize_frames < nframes){
free(stream->wrap_buffer);
stream->wrap_buffer = malloc(list.mBuffers[0].mDataByteSize);
stream->wrap_bufsize_frames = nframes;
}
list.mBuffers[0].mData = stream->wrap_buffer;
}else
list.mBuffers[0].mData = stream->cap_buffer + cap_wri_offs_frames * stream->fmt->nBlockAlign;
sc = AudioUnitRender(stream->unit, flags, ts, bus, nframes, &list);
if(sc != noErr){
OSSpinLockUnlock(&stream->lock);
return sc;
}
if(stream->playing){
if(list.mBuffers[0].mData == stream->wrap_buffer){
ca_wrap_buffer(stream->cap_buffer,
cap_wri_offs_frames * stream->fmt->nBlockAlign,
stream->cap_bufsize_frames * stream->fmt->nBlockAlign,
stream->wrap_buffer, list.mBuffers[0].mDataByteSize);
}
stream->cap_held_frames += list.mBuffers[0].mDataByteSize / stream->fmt->nBlockAlign;
if(stream->cap_held_frames > stream->cap_bufsize_frames){
stream->cap_offs_frames += stream->cap_held_frames % stream->cap_bufsize_frames;
stream->cap_offs_frames %= stream->cap_bufsize_frames;
stream->cap_held_frames = stream->cap_bufsize_frames;
}
}
OSSpinLockUnlock(&stream->lock);
return noErr;
}
static AudioComponentInstance get_audiounit(EDataFlow dataflow, AudioDeviceID adevid)
{
AudioComponentInstance unit;
AudioComponent comp;
AudioComponentDescription desc;
OSStatus sc;
memset(&desc, 0, sizeof(desc));
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
if(!(comp = AudioComponentFindNext(NULL, &desc))){
WARN("AudioComponentFindNext failed\n");
return NULL;
}
sc = AudioComponentInstanceNew(comp, &unit);
if(sc != noErr){
WARN("AudioComponentInstanceNew failed: %x\n", (int)sc);
return NULL;
}
if(dataflow == eCapture){
UInt32 enableio;
enableio = 1;
sc = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input, 1, &enableio, sizeof(enableio));
if(sc != noErr){
WARN("Couldn't enable I/O on input element: %x\n", (int)sc);
AudioComponentInstanceDispose(unit);
return NULL;
}
enableio = 0;
sc = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output, 0, &enableio, sizeof(enableio));
if(sc != noErr){
WARN("Couldn't disable I/O on output element: %x\n", (int)sc);
AudioComponentInstanceDispose(unit);
return NULL;
}
}
sc = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0, &adevid, sizeof(adevid));
if(sc != noErr){
WARN("Couldn't set audio unit device\n");
AudioComponentInstanceDispose(unit);
return NULL;
}
return unit;
}
static void dump_adesc(const char *aux, AudioStreamBasicDescription *desc)
{
TRACE("%s: mSampleRate: %f\n", aux, desc->mSampleRate);
TRACE("%s: mBytesPerPacket: %u\n", aux, (unsigned int)desc->mBytesPerPacket);
TRACE("%s: mFramesPerPacket: %u\n", aux, (unsigned int)desc->mFramesPerPacket);
TRACE("%s: mBytesPerFrame: %u\n", aux, (unsigned int)desc->mBytesPerFrame);
TRACE("%s: mChannelsPerFrame: %u\n", aux, (unsigned int)desc->mChannelsPerFrame);
TRACE("%s: mBitsPerChannel: %u\n", aux, (unsigned int)desc->mBitsPerChannel);
}
static HRESULT ca_get_audiodesc(AudioStreamBasicDescription *desc,
const WAVEFORMATEX *fmt)
{
const WAVEFORMATEXTENSIBLE *fmtex = (const WAVEFORMATEXTENSIBLE *)fmt;
desc->mFormatFlags = 0;
if(fmt->wFormatTag == WAVE_FORMAT_PCM ||
(fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))){
desc->mFormatID = kAudioFormatLinearPCM;
if(fmt->wBitsPerSample > 8)
desc->mFormatFlags = kAudioFormatFlagIsSignedInteger;
}else if(fmt->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
(fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))){
desc->mFormatID = kAudioFormatLinearPCM;
desc->mFormatFlags = kAudioFormatFlagIsFloat;
}else if(fmt->wFormatTag == WAVE_FORMAT_MULAW ||
(fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_MULAW))){
desc->mFormatID = kAudioFormatULaw;
}else if(fmt->wFormatTag == WAVE_FORMAT_ALAW ||
(fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_ALAW))){
desc->mFormatID = kAudioFormatALaw;
}else
return AUDCLNT_E_UNSUPPORTED_FORMAT;
desc->mSampleRate = fmt->nSamplesPerSec;
desc->mBytesPerPacket = fmt->nBlockAlign;
desc->mFramesPerPacket = 1;
desc->mBytesPerFrame = fmt->nBlockAlign;
desc->mChannelsPerFrame = fmt->nChannels;
desc->mBitsPerChannel = fmt->wBitsPerSample;
desc->mReserved = 0;
return S_OK;
}
static HRESULT ca_setup_audiounit(EDataFlow dataflow, AudioComponentInstance unit,
const WAVEFORMATEX *fmt, AudioStreamBasicDescription *dev_desc,
AudioConverterRef *converter)
{
OSStatus sc;
HRESULT hr;
if(dataflow == eCapture){
AudioStreamBasicDescription desc;
UInt32 size;
Float64 rate;
fenv_t fenv;
BOOL fenv_stored = TRUE;
hr = ca_get_audiodesc(&desc, fmt);
if(FAILED(hr))
return hr;
dump_adesc("requested", &desc);
/* input-only units can't perform sample rate conversion, so we have to
* set up our own AudioConverter to support arbitrary sample rates. */
size = sizeof(*dev_desc);
sc = AudioUnitGetProperty(unit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 1, dev_desc, &size);
if(sc != noErr){
WARN("Couldn't get unit format: %x\n", (int)sc);
return osstatus_to_hresult(sc);
}
dump_adesc("hardware", dev_desc);
rate = dev_desc->mSampleRate;
*dev_desc = desc;
dev_desc->mSampleRate = rate;
dump_adesc("final", dev_desc);
sc = AudioUnitSetProperty(unit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 1, dev_desc, sizeof(*dev_desc));
if(sc != noErr){
WARN("Couldn't set unit format: %x\n", (int)sc);
return osstatus_to_hresult(sc);
}
/* AudioConverterNew requires divide-by-zero SSE exceptions to be masked */
if(feholdexcept(&fenv)){
WARN("Failed to store fenv state\n");
fenv_stored = FALSE;
}
sc = AudioConverterNew(dev_desc, &desc, converter);
if(fenv_stored && fesetenv(&fenv))
WARN("Failed to restore fenv state\n");
if(sc != noErr){
WARN("Couldn't create audio converter: %x\n", (int)sc);
return osstatus_to_hresult(sc);
}
}else{
hr = ca_get_audiodesc(dev_desc, fmt);
if(FAILED(hr))
return hr;
dump_adesc("final", dev_desc);
sc = AudioUnitSetProperty(unit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, dev_desc, sizeof(*dev_desc));
if(sc != noErr){
WARN("Couldn't set format: %x\n", (int)sc);
return osstatus_to_hresult(sc);
}
}
return S_OK;
}
static NTSTATUS create_stream(void *args)
{
struct create_stream_params *params = args;
struct coreaudio_stream *stream = calloc(1, sizeof(*stream));
AURenderCallbackStruct input;
OSStatus sc;
if(!stream){
params->result = E_OUTOFMEMORY;
return STATUS_SUCCESS;
}
stream->fmt = clone_format(params->fmt);
if(!stream->fmt){
params->result = E_OUTOFMEMORY;
goto end;
}
stream->period_ms = params->period / 10000;
stream->period_frames = muldiv(params->period, stream->fmt->nSamplesPerSec, 10000000);
stream->dev_id = params->dev_id;
stream->flow = params->flow;
stream->share = params->share;
stream->bufsize_frames = muldiv(params->duration, stream->fmt->nSamplesPerSec, 10000000);
if(params->share == AUDCLNT_SHAREMODE_EXCLUSIVE)
stream->bufsize_frames -= stream->bufsize_frames % stream->period_frames;
if(!(stream->unit = get_audiounit(stream->flow, stream->dev_id))){
params->result = AUDCLNT_E_DEVICE_INVALIDATED;
goto end;
}
params->result = ca_setup_audiounit(stream->flow, stream->unit, stream->fmt, &stream->dev_desc, &stream->converter);
if(FAILED(params->result)) goto end;
input.inputProcRefCon = stream;
if(stream->flow == eCapture){
input.inputProc = ca_capture_cb;
sc = AudioUnitSetProperty(stream->unit, kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Output, 1, &input, sizeof(input));
}else{
input.inputProc = ca_render_cb;
sc = AudioUnitSetProperty(stream->unit, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &input, sizeof(input));
}
if(sc != noErr){
WARN("Couldn't set callback: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
goto end;
}
sc = AudioUnitInitialize(stream->unit);
if(sc != noErr){
WARN("Couldn't initialize: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
goto end;
}
/* we play audio continuously because AudioOutputUnitStart sometimes takes
* a while to return */
sc = AudioOutputUnitStart(stream->unit);
if(sc != noErr){
WARN("Unit failed to start: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
goto end;
}
stream->local_buffer_size = stream->bufsize_frames * stream->fmt->nBlockAlign;
if(NtAllocateVirtualMemory(GetCurrentProcess(), (void **)&stream->local_buffer, 0, &stream->local_buffer_size,
MEM_COMMIT, PAGE_READWRITE)){
params->result = E_OUTOFMEMORY;
goto end;
}
silence_buffer(stream, stream->local_buffer, stream->bufsize_frames);
if(stream->flow == eCapture){
stream->cap_bufsize_frames = muldiv(params->duration, stream->dev_desc.mSampleRate, 10000000);
stream->cap_buffer = malloc(stream->cap_bufsize_frames * stream->fmt->nBlockAlign);
}
params->result = S_OK;
end:
if(FAILED(params->result)){
if(stream->converter) AudioConverterDispose(stream->converter);
if(stream->unit) AudioComponentInstanceDispose(stream->unit);
free(stream->fmt);
free(stream);
} else
params->stream = stream;
return STATUS_SUCCESS;
}
static NTSTATUS release_stream( void *args )
{
struct release_stream_params *params = args;
struct coreaudio_stream *stream = params->stream;
if(stream->unit){
AudioOutputUnitStop(stream->unit);
AudioComponentInstanceDispose(stream->unit);
}
if(stream->converter) AudioConverterDispose(stream->converter);
free(stream->resamp_buffer);
free(stream->wrap_buffer);
free(stream->cap_buffer);
if(stream->local_buffer)
NtFreeVirtualMemory(GetCurrentProcess(), (void **)&stream->local_buffer,
&stream->local_buffer_size, MEM_RELEASE);
if(stream->tmp_buffer)
NtFreeVirtualMemory(GetCurrentProcess(), (void **)&stream->tmp_buffer,
&stream->tmp_buffer_size, MEM_RELEASE);
free(stream->fmt);
params->result = S_OK;
return STATUS_SUCCESS;
}
static DWORD ca_channel_layout_to_channel_mask(const AudioChannelLayout *layout)
{
int i;
DWORD mask = 0;
for (i = 0; i < layout->mNumberChannelDescriptions; ++i) {
switch (layout->mChannelDescriptions[i].mChannelLabel) {
default: FIXME("Unhandled channel 0x%x\n",
(unsigned int)layout->mChannelDescriptions[i].mChannelLabel); break;
case kAudioChannelLabel_Left: mask |= SPEAKER_FRONT_LEFT; break;
case kAudioChannelLabel_Mono:
case kAudioChannelLabel_Center: mask |= SPEAKER_FRONT_CENTER; break;
case kAudioChannelLabel_Right: mask |= SPEAKER_FRONT_RIGHT; break;
case kAudioChannelLabel_LeftSurround: mask |= SPEAKER_BACK_LEFT; break;
case kAudioChannelLabel_CenterSurround: mask |= SPEAKER_BACK_CENTER; break;
case kAudioChannelLabel_RightSurround: mask |= SPEAKER_BACK_RIGHT; break;
case kAudioChannelLabel_LFEScreen: mask |= SPEAKER_LOW_FREQUENCY; break;
case kAudioChannelLabel_LeftSurroundDirect: mask |= SPEAKER_SIDE_LEFT; break;
case kAudioChannelLabel_RightSurroundDirect: mask |= SPEAKER_SIDE_RIGHT; break;
case kAudioChannelLabel_TopCenterSurround: mask |= SPEAKER_TOP_CENTER; break;
case kAudioChannelLabel_VerticalHeightLeft: mask |= SPEAKER_TOP_FRONT_LEFT; break;
case kAudioChannelLabel_VerticalHeightCenter: mask |= SPEAKER_TOP_FRONT_CENTER; break;
case kAudioChannelLabel_VerticalHeightRight: mask |= SPEAKER_TOP_FRONT_RIGHT; break;
case kAudioChannelLabel_TopBackLeft: mask |= SPEAKER_TOP_BACK_LEFT; break;
case kAudioChannelLabel_TopBackCenter: mask |= SPEAKER_TOP_BACK_CENTER; break;
case kAudioChannelLabel_TopBackRight: mask |= SPEAKER_TOP_BACK_RIGHT; break;
case kAudioChannelLabel_LeftCenter: mask |= SPEAKER_FRONT_LEFT_OF_CENTER; break;
case kAudioChannelLabel_RightCenter: mask |= SPEAKER_FRONT_RIGHT_OF_CENTER; break;
}
}
return mask;
}
/* For most hardware on Windows, users must choose a configuration with an even
* number of channels (stereo, quad, 5.1, 7.1). Users can then disable
* channels, but those channels are still reported to applications from
* GetMixFormat! Some applications behave badly if given an odd number of
* channels (e.g. 2.1). Here, we find the nearest configuration that Windows
* would report for a given channel layout. */
static void convert_channel_layout(const AudioChannelLayout *ca_layout, WAVEFORMATEXTENSIBLE *fmt)
{
DWORD ca_mask = ca_channel_layout_to_channel_mask(ca_layout);
TRACE("Got channel mask for CA: 0x%x\n", ca_mask);
if (ca_layout->mNumberChannelDescriptions == 1)
{
fmt->Format.nChannels = 1;
fmt->dwChannelMask = ca_mask;
return;
}
/* compare against known configurations and find smallest configuration
* which is a superset of the given speakers */
if (ca_layout->mNumberChannelDescriptions <= 2 &&
(ca_mask & ~KSAUDIO_SPEAKER_STEREO) == 0)
{
fmt->Format.nChannels = 2;
fmt->dwChannelMask = KSAUDIO_SPEAKER_STEREO;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 4 &&
(ca_mask & ~KSAUDIO_SPEAKER_QUAD) == 0)
{
fmt->Format.nChannels = 4;
fmt->dwChannelMask = KSAUDIO_SPEAKER_QUAD;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 4 &&
(ca_mask & ~KSAUDIO_SPEAKER_SURROUND) == 0)
{
fmt->Format.nChannels = 4;
fmt->dwChannelMask = KSAUDIO_SPEAKER_SURROUND;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 6 &&
(ca_mask & ~KSAUDIO_SPEAKER_5POINT1) == 0)
{
fmt->Format.nChannels = 6;
fmt->dwChannelMask = KSAUDIO_SPEAKER_5POINT1;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 6 &&
(ca_mask & ~KSAUDIO_SPEAKER_5POINT1_SURROUND) == 0)
{
fmt->Format.nChannels = 6;
fmt->dwChannelMask = KSAUDIO_SPEAKER_5POINT1_SURROUND;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 8 &&
(ca_mask & ~KSAUDIO_SPEAKER_7POINT1) == 0)
{
fmt->Format.nChannels = 8;
fmt->dwChannelMask = KSAUDIO_SPEAKER_7POINT1;
return;
}
if (ca_layout->mNumberChannelDescriptions <= 8 &&
(ca_mask & ~KSAUDIO_SPEAKER_7POINT1_SURROUND) == 0)
{
fmt->Format.nChannels = 8;
fmt->dwChannelMask = KSAUDIO_SPEAKER_7POINT1_SURROUND;
return;
}
/* oddball format, report truthfully */
fmt->Format.nChannels = ca_layout->mNumberChannelDescriptions;
fmt->dwChannelMask = ca_mask;
}
static DWORD get_channel_mask(unsigned int channels)
{
switch(channels){
case 0:
return 0;
case 1:
return KSAUDIO_SPEAKER_MONO;
case 2:
return KSAUDIO_SPEAKER_STEREO;
case 3:
return KSAUDIO_SPEAKER_STEREO | SPEAKER_LOW_FREQUENCY;
case 4:
return KSAUDIO_SPEAKER_QUAD; /* not _SURROUND */
case 5:
return KSAUDIO_SPEAKER_QUAD | SPEAKER_LOW_FREQUENCY;
case 6:
return KSAUDIO_SPEAKER_5POINT1; /* not 5POINT1_SURROUND */
case 7:
return KSAUDIO_SPEAKER_5POINT1 | SPEAKER_BACK_CENTER;
case 8:
return KSAUDIO_SPEAKER_7POINT1_SURROUND; /* Vista deprecates 7POINT1 */
}
FIXME("Unknown speaker configuration: %u\n", channels);
return 0;
}
static NTSTATUS get_mix_format(void *args)
{
struct get_mix_format_params *params = args;
AudioObjectPropertyAddress addr;
AudioChannelLayout *layout;
AudioBufferList *buffers;
Float64 rate;
UInt32 size;
OSStatus sc;
int i;
params->fmt->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
addr.mScope = get_scope(params->flow);
addr.mElement = 0;
addr.mSelector = kAudioDevicePropertyPreferredChannelLayout;
sc = AudioObjectGetPropertyDataSize(params->dev_id, &addr, 0, NULL, &size);
if(sc == noErr){
layout = malloc(size);
sc = AudioObjectGetPropertyData(params->dev_id, &addr, 0, NULL, &size, layout);
if(sc == noErr){
TRACE("Got channel layout: {tag: 0x%x, bitmap: 0x%x, num_descs: %u}\n",
(unsigned int)layout->mChannelLayoutTag, (unsigned int)layout->mChannelBitmap,
(unsigned int)layout->mNumberChannelDescriptions);
if(layout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions){
convert_channel_layout(layout, params->fmt);
}else{
WARN("Haven't implemented support for this layout tag: 0x%x, guessing at layout\n",
(unsigned int)layout->mChannelLayoutTag);
params->fmt->Format.nChannels = 0;
}
}else{
TRACE("Unable to get _PreferredChannelLayout property: %x, guessing at layout\n", (int)sc);
params->fmt->Format.nChannels = 0;
}
free(layout);
}else{
TRACE("Unable to get size for _PreferredChannelLayout property: %x, guessing at layout\n", (int)sc);
params->fmt->Format.nChannels = 0;
}
if(params->fmt->Format.nChannels == 0){
addr.mScope = get_scope(params->flow);
addr.mElement = 0;
addr.mSelector = kAudioDevicePropertyStreamConfiguration;
sc = AudioObjectGetPropertyDataSize(params->dev_id, &addr, 0, NULL, &size);
if(sc != noErr){
WARN("Unable to get size for _StreamConfiguration property: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
buffers = malloc(size);
if(!buffers){
params->result = E_OUTOFMEMORY;
return STATUS_SUCCESS;
}
sc = AudioObjectGetPropertyData(params->dev_id, &addr, 0, NULL, &size, buffers);
if(sc != noErr){
free(buffers);
WARN("Unable to get _StreamConfiguration property: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
for(i = 0; i < buffers->mNumberBuffers; ++i)
params->fmt->Format.nChannels += buffers->mBuffers[i].mNumberChannels;
free(buffers);
params->fmt->dwChannelMask = get_channel_mask(params->fmt->Format.nChannels);
}
addr.mSelector = kAudioDevicePropertyNominalSampleRate;
size = sizeof(Float64);
sc = AudioObjectGetPropertyData(params->dev_id, &addr, 0, NULL, &size, &rate);
if(sc != noErr){
WARN("Unable to get _NominalSampleRate property: %x\n", (int)sc);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
params->fmt->Format.nSamplesPerSec = rate;
params->fmt->Format.wBitsPerSample = 32;
params->fmt->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
params->fmt->Format.nBlockAlign = (params->fmt->Format.wBitsPerSample *
params->fmt->Format.nChannels) / 8;
params->fmt->Format.nAvgBytesPerSec = params->fmt->Format.nSamplesPerSec *
params->fmt->Format.nBlockAlign;
params->fmt->Samples.wValidBitsPerSample = params->fmt->Format.wBitsPerSample;
params->fmt->Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
params->result = S_OK;
return STATUS_SUCCESS;
}
static NTSTATUS is_format_supported(void *args)
{
struct is_format_supported_params *params = args;
const WAVEFORMATEXTENSIBLE *fmtex = (const WAVEFORMATEXTENSIBLE *)params->fmt_in;
AudioStreamBasicDescription dev_desc;
AudioConverterRef converter;
AudioComponentInstance unit;
params->result = S_OK;
if(!params->fmt_in || (params->share == AUDCLNT_SHAREMODE_SHARED && !params->fmt_out))
params->result = E_POINTER;
else if(params->share != AUDCLNT_SHAREMODE_SHARED && params->share != AUDCLNT_SHAREMODE_EXCLUSIVE)
params->result = E_INVALIDARG;
else if(params->fmt_in->wFormatTag == WAVE_FORMAT_EXTENSIBLE){
if(params->fmt_in->cbSize < sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX))
params->result = E_INVALIDARG;
else if(params->fmt_in->nAvgBytesPerSec == 0 || params->fmt_in->nBlockAlign == 0 ||
fmtex->Samples.wValidBitsPerSample > params->fmt_in->wBitsPerSample)
params->result = E_INVALIDARG;
else if(fmtex->Samples.wValidBitsPerSample < params->fmt_in->wBitsPerSample)
goto unsupported;
else if(params->share == AUDCLNT_SHAREMODE_EXCLUSIVE &&
(fmtex->dwChannelMask == 0 || fmtex->dwChannelMask & SPEAKER_RESERVED))
goto unsupported;
}
if(FAILED(params->result)) return STATUS_SUCCESS;
if(params->fmt_in->nBlockAlign != params->fmt_in->nChannels * params->fmt_in->wBitsPerSample / 8 ||
params->fmt_in->nAvgBytesPerSec != params->fmt_in->nBlockAlign * params->fmt_in->nSamplesPerSec)
goto unsupported;
if(params->fmt_in->nChannels == 0){
params->result = AUDCLNT_E_UNSUPPORTED_FORMAT;
return STATUS_SUCCESS;
}
unit = get_audiounit(params->flow, params->dev_id);
converter = NULL;
params->result = ca_setup_audiounit(params->flow, unit, params->fmt_in, &dev_desc, &converter);
AudioComponentInstanceDispose(unit);
if(FAILED(params->result)) goto unsupported;
if(converter) AudioConverterDispose(converter);
params->result = S_OK;
return STATUS_SUCCESS;
unsupported:
if(params->fmt_out){
struct get_mix_format_params get_mix_params =
{
.flow = params->flow,
.dev_id = params->dev_id,
.fmt = params->fmt_out,
};
get_mix_format(&get_mix_params);
params->result = get_mix_params.result;
if(SUCCEEDED(params->result)) params->result = S_FALSE;
}
else params->result = AUDCLNT_E_UNSUPPORTED_FORMAT;
return STATUS_SUCCESS;
}
static UINT buf_ptr_diff(UINT left, UINT right, UINT bufsize)
{
if(left <= right)
return right - left;
return bufsize - (left - right);
}
/* place data from cap_buffer into provided AudioBufferList */
static OSStatus feed_cb(AudioConverterRef converter, UInt32 *nframes, AudioBufferList *data,
AudioStreamPacketDescription **packets, void *user)
{
struct coreaudio_stream *stream = user;
*nframes = min(*nframes, stream->cap_held_frames);
if(!*nframes){
data->mBuffers[0].mData = NULL;
data->mBuffers[0].mDataByteSize = 0;
data->mBuffers[0].mNumberChannels = stream->fmt->nChannels;
return noErr;
}
data->mBuffers[0].mDataByteSize = *nframes * stream->fmt->nBlockAlign;
data->mBuffers[0].mNumberChannels = stream->fmt->nChannels;
if(stream->cap_offs_frames + *nframes > stream->cap_bufsize_frames){
UINT32 chunk_frames = stream->cap_bufsize_frames - stream->cap_offs_frames;
if(stream->wrap_bufsize_frames < *nframes){
free(stream->wrap_buffer);
stream->wrap_buffer = malloc(data->mBuffers[0].mDataByteSize);
stream->wrap_bufsize_frames = *nframes;
}
memcpy(stream->wrap_buffer, stream->cap_buffer + stream->cap_offs_frames * stream->fmt->nBlockAlign,
chunk_frames * stream->fmt->nBlockAlign);
memcpy(stream->wrap_buffer + chunk_frames * stream->fmt->nBlockAlign, stream->cap_buffer,
(*nframes - chunk_frames) * stream->fmt->nBlockAlign);
data->mBuffers[0].mData = stream->wrap_buffer;
}else
data->mBuffers[0].mData = stream->cap_buffer + stream->cap_offs_frames * stream->fmt->nBlockAlign;
stream->cap_offs_frames += *nframes;
stream->cap_offs_frames %= stream->cap_bufsize_frames;
stream->cap_held_frames -= *nframes;
if(packets)
*packets = NULL;
return noErr;
}
static void capture_resample(struct coreaudio_stream *stream)
{
UINT32 resamp_period_frames = muldiv(stream->period_frames, stream->dev_desc.mSampleRate,
stream->fmt->nSamplesPerSec);
OSStatus sc;
/* the resampling process often needs more source frames than we'd
* guess from a straight conversion using the sample rate ratio. so
* only convert if we have extra source data. */
while(stream->cap_held_frames > resamp_period_frames * 2){
AudioBufferList converted_list;
UInt32 wanted_frames = stream->period_frames;
converted_list.mNumberBuffers = 1;
converted_list.mBuffers[0].mNumberChannels = stream->fmt->nChannels;
converted_list.mBuffers[0].mDataByteSize = wanted_frames * stream->fmt->nBlockAlign;
if(stream->resamp_bufsize_frames < wanted_frames){
free(stream->resamp_buffer);
stream->resamp_buffer = malloc(converted_list.mBuffers[0].mDataByteSize);
stream->resamp_bufsize_frames = wanted_frames;
}
converted_list.mBuffers[0].mData = stream->resamp_buffer;
sc = AudioConverterFillComplexBuffer(stream->converter, feed_cb,
stream, &wanted_frames, &converted_list, NULL);
if(sc != noErr){
WARN("AudioConverterFillComplexBuffer failed: %x\n", (int)sc);
break;
}
ca_wrap_buffer(stream->local_buffer,
stream->wri_offs_frames * stream->fmt->nBlockAlign,
stream->bufsize_frames * stream->fmt->nBlockAlign,
stream->resamp_buffer, wanted_frames * stream->fmt->nBlockAlign);
stream->wri_offs_frames += wanted_frames;
stream->wri_offs_frames %= stream->bufsize_frames;
if(stream->held_frames + wanted_frames > stream->bufsize_frames){
stream->lcl_offs_frames += buf_ptr_diff(stream->lcl_offs_frames, stream->wri_offs_frames,
stream->bufsize_frames);
stream->held_frames = stream->bufsize_frames;
}else
stream->held_frames += wanted_frames;
}
}
static NTSTATUS get_buffer_size(void *args)
{
struct get_buffer_size_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
*params->frames = stream->bufsize_frames;
OSSpinLockUnlock(&stream->lock);
params->result = S_OK;
return STATUS_SUCCESS;
}
static HRESULT ca_get_max_stream_latency(struct coreaudio_stream *stream, UInt32 *max)
{
AudioObjectPropertyAddress addr;
AudioStreamID *ids;
UInt32 size;
OSStatus sc;
int nstreams, i;
addr.mScope = get_scope(stream->flow);
addr.mElement = 0;
addr.mSelector = kAudioDevicePropertyStreams;
sc = AudioObjectGetPropertyDataSize(stream->dev_id, &addr, 0, NULL, &size);
if(sc != noErr){
WARN("Unable to get size for _Streams property: %x\n", (int)sc);
return osstatus_to_hresult(sc);
}
ids = malloc(size);
if(!ids)
return E_OUTOFMEMORY;
sc = AudioObjectGetPropertyData(stream->dev_id, &addr, 0, NULL, &size, ids);
if(sc != noErr){
WARN("Unable to get _Streams property: %x\n", (int)sc);
free(ids);
return osstatus_to_hresult(sc);
}
nstreams = size / sizeof(AudioStreamID);
*max = 0;
addr.mSelector = kAudioStreamPropertyLatency;
for(i = 0; i < nstreams; ++i){
UInt32 latency;
size = sizeof(latency);
sc = AudioObjectGetPropertyData(ids[i], &addr, 0, NULL, &size, &latency);
if(sc != noErr){
WARN("Unable to get _Latency property: %x\n", (int)sc);
continue;
}
if(latency > *max)
*max = latency;
}
free(ids);
return S_OK;
}
static NTSTATUS get_latency(void *args)
{
struct get_latency_params *params = args;
struct coreaudio_stream *stream = params->stream;
UInt32 latency, stream_latency, size;
AudioObjectPropertyAddress addr;
OSStatus sc;
OSSpinLockLock(&stream->lock);
addr.mScope = get_scope(stream->flow);
addr.mSelector = kAudioDevicePropertyLatency;
addr.mElement = 0;
size = sizeof(latency);
sc = AudioObjectGetPropertyData(stream->dev_id, &addr, 0, NULL, &size, &latency);
if(sc != noErr){
WARN("Couldn't get _Latency property: %x\n", (int)sc);
OSSpinLockUnlock(&stream->lock);
params->result = osstatus_to_hresult(sc);
return STATUS_SUCCESS;
}
params->result = ca_get_max_stream_latency(stream, &stream_latency);
if(FAILED(params->result)){
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
latency += stream_latency;
/* pretend we process audio in Period chunks, so max latency includes
* the period time */
*params->latency = muldiv(latency, 10000000, stream->fmt->nSamplesPerSec)
+ stream->period_ms * 10000;
OSSpinLockUnlock(&stream->lock);
params->result = S_OK;
return STATUS_SUCCESS;
}
static UINT32 get_current_padding_nolock(struct coreaudio_stream *stream)
{
if(stream->flow == eCapture) capture_resample(stream);
return stream->held_frames;
}
static NTSTATUS get_current_padding(void *args)
{
struct get_current_padding_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
*params->padding = get_current_padding_nolock(stream);
OSSpinLockUnlock(&stream->lock);
params->result = S_OK;
return STATUS_SUCCESS;
}
static NTSTATUS start(void *args)
{
struct start_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
if(stream->playing)
params->result = AUDCLNT_E_NOT_STOPPED;
else{
stream->playing = TRUE;
params->result = S_OK;
}
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS stop(void *args)
{
struct stop_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
if(!stream->playing)
params->result = S_FALSE;
else{
stream->playing = FALSE;
params->result = S_OK;
}
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS reset(void *args)
{
struct reset_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
if(stream->playing)
params->result = AUDCLNT_E_NOT_STOPPED;
else if(stream->getbuf_last)
params->result = AUDCLNT_E_BUFFER_OPERATION_PENDING;
else{
if(stream->flow == eRender)
stream->written_frames = 0;
else
stream->written_frames += stream->held_frames;
stream->held_frames = 0;
stream->lcl_offs_frames = 0;
stream->wri_offs_frames = 0;
stream->cap_offs_frames = 0;
stream->cap_held_frames = 0;
params->result = S_OK;
}
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS get_render_buffer(void *args)
{
struct get_render_buffer_params *params = args;
struct coreaudio_stream *stream = params->stream;
UINT32 pad;
OSSpinLockLock(&stream->lock);
pad = get_current_padding_nolock(stream);
if(stream->getbuf_last){
params->result = AUDCLNT_E_OUT_OF_ORDER;
goto end;
}
if(!params->frames){
params->result = S_OK;
goto end;
}
if(pad + params->frames > stream->bufsize_frames){
params->result = AUDCLNT_E_BUFFER_TOO_LARGE;
goto end;
}
if(stream->wri_offs_frames + params->frames > stream->bufsize_frames){
if(stream->tmp_buffer_frames < params->frames){
NtFreeVirtualMemory(GetCurrentProcess(), (void **)&stream->tmp_buffer,
&stream->tmp_buffer_size, MEM_RELEASE);
stream->tmp_buffer_size = params->frames * stream->fmt->nBlockAlign;
if(NtAllocateVirtualMemory(GetCurrentProcess(), (void **)&stream->tmp_buffer, 0,
&stream->tmp_buffer_size, MEM_COMMIT, PAGE_READWRITE)){
stream->tmp_buffer_frames = 0;
params->result = E_OUTOFMEMORY;
goto end;
}
stream->tmp_buffer_frames = params->frames;
}
*params->data = stream->tmp_buffer;
stream->getbuf_last = -params->frames;
}else{
*params->data = stream->local_buffer + stream->wri_offs_frames * stream->fmt->nBlockAlign;
stream->getbuf_last = params->frames;
}
silence_buffer(stream, *params->data, params->frames);
params->result = S_OK;
end:
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS release_render_buffer(void *args)
{
struct release_render_buffer_params *params = args;
struct coreaudio_stream *stream = params->stream;
BYTE *buffer;
OSSpinLockLock(&stream->lock);
if(!params->frames){
stream->getbuf_last = 0;
params->result = S_OK;
}else if(!stream->getbuf_last)
params->result = AUDCLNT_E_OUT_OF_ORDER;
else if(params->frames > (stream->getbuf_last >= 0 ? stream->getbuf_last : -stream->getbuf_last))
params->result = AUDCLNT_E_INVALID_SIZE;
else{
if(stream->getbuf_last >= 0)
buffer = stream->local_buffer + stream->wri_offs_frames * stream->fmt->nBlockAlign;
else
buffer = stream->tmp_buffer;
if(params->flags & AUDCLNT_BUFFERFLAGS_SILENT)
silence_buffer(stream, buffer, params->frames);
if(stream->getbuf_last < 0)
ca_wrap_buffer(stream->local_buffer,
stream->wri_offs_frames * stream->fmt->nBlockAlign,
stream->bufsize_frames * stream->fmt->nBlockAlign,
buffer, params->frames * stream->fmt->nBlockAlign);
stream->wri_offs_frames += params->frames;
stream->wri_offs_frames %= stream->bufsize_frames;
stream->held_frames += params->frames;
stream->written_frames += params->frames;
stream->getbuf_last = 0;
params->result = S_OK;
}
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS get_capture_buffer(void *args)
{
struct get_capture_buffer_params *params = args;
struct coreaudio_stream *stream = params->stream;
UINT32 chunk_bytes, chunk_frames;
LARGE_INTEGER stamp, freq;
OSSpinLockLock(&stream->lock);
if(stream->getbuf_last){
params->result = AUDCLNT_E_OUT_OF_ORDER;
goto end;
}
capture_resample(stream);
*params->frames = 0;
if(stream->held_frames < stream->period_frames){
params->result = AUDCLNT_S_BUFFER_EMPTY;
goto end;
}
*params->flags = 0;
chunk_frames = stream->bufsize_frames - stream->lcl_offs_frames;
if(chunk_frames < stream->period_frames){
chunk_bytes = chunk_frames * stream->fmt->nBlockAlign;
if(!stream->tmp_buffer){
stream->tmp_buffer_size = stream->period_frames * stream->fmt->nBlockAlign;
NtAllocateVirtualMemory(GetCurrentProcess(), (void **)&stream->tmp_buffer, 0,
&stream->tmp_buffer_size, MEM_COMMIT, PAGE_READWRITE);
}
*params->data = stream->tmp_buffer;
memcpy(*params->data, stream->local_buffer + stream->lcl_offs_frames * stream->fmt->nBlockAlign,
chunk_bytes);
memcpy(*params->data + chunk_bytes, stream->local_buffer,
stream->period_frames * stream->fmt->nBlockAlign - chunk_bytes);
}else
*params->data = stream->local_buffer + stream->lcl_offs_frames * stream->fmt->nBlockAlign;
stream->getbuf_last = *params->frames = stream->period_frames;
if(params->devpos)
*params->devpos = stream->written_frames;
if(params->qpcpos){ /* fixme: qpc of recording time */
NtQueryPerformanceCounter(&stamp, &freq);
*params->qpcpos = (stamp.QuadPart * (INT64)10000000) / freq.QuadPart;
}
params->result = S_OK;
end:
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS release_capture_buffer(void *args)
{
struct release_capture_buffer_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
if(!params->done){
stream->getbuf_last = 0;
params->result = S_OK;
}else if(!stream->getbuf_last)
params->result = AUDCLNT_E_OUT_OF_ORDER;
else if(stream->getbuf_last != params->done)
params->result = AUDCLNT_E_INVALID_SIZE;
else{
stream->written_frames += params->done;
stream->held_frames -= params->done;
stream->lcl_offs_frames += params->done;
stream->lcl_offs_frames %= stream->bufsize_frames;
stream->getbuf_last = 0;
params->result = S_OK;
}
OSSpinLockUnlock(&stream->lock);
return STATUS_SUCCESS;
}
static NTSTATUS get_next_packet_size(void *args)
{
struct get_next_packet_size_params *params = args;
struct coreaudio_stream *stream = params->stream;
OSSpinLockLock(&stream->lock);
capture_resample(stream);
if(stream->held_frames >= stream->period_frames)
*params->frames = stream->period_frames;
else
*params->frames = 0;
OSSpinLockUnlock(&stream->lock);
params->result = S_OK;
return STATUS_SUCCESS;
}
static NTSTATUS get_position(void *args)
{
struct get_position_params *params = args;
struct coreaudio_stream *stream = params->stream;
LARGE_INTEGER stamp, freq;
OSSpinLockLock(&stream->lock);
*params->pos = stream->written_frames - stream->held_frames;
if(stream->share == AUDCLNT_SHAREMODE_SHARED)
*params->pos *= stream->fmt->nBlockAlign;
if(params->qpctime){
NtQueryPerformanceCounter(&stamp, &freq);
*params->qpctime = (stamp.QuadPart * (INT64)10000000) / freq.QuadPart;
}
OSSpinLockUnlock(&stream->lock);
params->result = S_OK;
return STATUS_SUCCESS;
}
static NTSTATUS get_frequency(void *args)
{
struct get_frequency_params *params = args;
struct coreaudio_stream *stream = params->stream;
if(stream->share == AUDCLNT_SHAREMODE_SHARED)
*params->freq = (UINT64)stream->fmt->nSamplesPerSec * stream->fmt->nBlockAlign;
else
*params->freq = stream->fmt->nSamplesPerSec;
params->result = S_OK;
return STATUS_SUCCESS;
}
static NTSTATUS is_started(void *args)
{
struct is_started_params *params = args;
struct coreaudio_stream *stream = params->stream;
if(stream->playing)
params->result = S_OK;
else
params->result = S_FALSE;
return STATUS_SUCCESS;
}
static NTSTATUS set_volumes(void *args)
{
struct set_volumes_params *params = args;
struct coreaudio_stream *stream = params->stream;
Float32 level = 1.0, tmp;
OSStatus sc;
UINT32 i;
if(params->channel >= stream->fmt->nChannels || params->channel < -1){
ERR("Incorrect channel %d\n", params->channel);
return STATUS_SUCCESS;
}
if(params->channel == -1){
for(i = 0; i < stream->fmt->nChannels; ++i){
tmp = params->master_volume * params->volumes[i] * params->session_volumes[i];
level = tmp < level ? tmp : level;
}
}else
level = params->master_volume * params->volumes[params->channel] *
params->session_volumes[params->channel];
sc = AudioUnitSetParameter(stream->unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, level, 0);
if(sc != noErr)
WARN("Couldn't set volume: %x\n", (int)sc);
return STATUS_SUCCESS;
}
unixlib_entry_t __wine_unix_call_funcs[] =
{
get_endpoint_ids,
create_stream,
release_stream,
start,
stop,
reset,
get_render_buffer,
release_render_buffer,
get_capture_buffer,
release_capture_buffer,
get_mix_format,
is_format_supported,
get_buffer_size,
get_latency,
get_current_padding,
get_next_packet_size,
get_position,
get_frequency,
is_started,
set_volumes,
midi_init,
midi_release,
midi_out_message,
midi_in_message,
midi_notify_wait,
};
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