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Commit 46928d38 authored by Andrew Eikum's avatar Andrew Eikum Committed by Alexandre Julliard
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winecoreaudio.drv: Use AUHAL API instead of AudioQueue.

parent ef1342ba
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Inline Side-by-side
Showing with 603 additions and 578 deletions
dlls/winecoreaudio.drv/mmdevdrv.c
View file @ 46928d38
......@@ -20,8 +20,37 @@
#define COBJMACROS
#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 <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
#undef DPRINTF
#include "windef.h"
#include "winbase.h"
#include "winnls.h"
......@@ -41,42 +70,35 @@
#include "audioclient.h"
#include "audiopolicy.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 <unistd.h>
WINE_DEFAULT_DEBUG_CHANNEL(coreaudio);
#include <libkern/OSAtomic.h>
#include <CoreAudio/CoreAudio.h>
#include <AudioToolbox/AudioQueue.h>
#include <AudioToolbox/AudioFormat.h>
#ifndef HAVE_AUDIOUNIT_AUDIOCOMPONENT_H
/* Define new AudioComponent Manager functions for OSX 10.5 */
typedef Component AudioComponent;
typedef ComponentDescription AudioComponentDescription;
typedef ComponentInstance AudioComponentInstance;
WINE_DEFAULT_DEBUG_CHANNEL(coreaudio);
static inline AudioComponent AudioComponentFindNext(AudioComponent ac, AudioComponentDescription *desc)
{
return FindNextComponent(ac, desc);
}
static inline OSStatus AudioComponentInstanceNew(AudioComponent ac, AudioComponentInstance *aci)
{
return OpenAComponent(ac, aci);
}
static inline OSStatus AudioComponentInstanceDispose(AudioComponentInstance aci)
{
return CloseComponent(aci);
}
#endif
#define NULL_PTR_ERR MAKE_HRESULT(SEVERITY_ERROR, FACILITY_WIN32, RPC_X_NULL_REF_POINTER)
static const REFERENCE_TIME DefaultPeriod = 100000;
static const REFERENCE_TIME MinimumPeriod = 50000;
typedef struct _QueuedBufInfo {
Float64 start_sampletime;
UINT64 start_pos;
UINT32 len_frames;
struct list entry;
} QueuedBufInfo;
typedef struct _AQBuffer {
AudioQueueBufferRef buf;
struct list entry;
BOOL used;
} AQBuffer;
struct ACImpl;
typedef struct ACImpl ACImpl;
......@@ -128,37 +150,29 @@ struct ACImpl {
HANDLE event;
float *vols;
BOOL initted;
AudioDeviceID adevid;
AudioQueueRef aqueue;
AudioObjectPropertyScope scope;
AudioConverterRef converter;
AudioComponentInstance unit;
AudioStreamBasicDescription dev_desc; /* audio unit format, not necessarily the same as fmt */
HANDLE timer;
UINT32 period_ms, bufsize_frames, inbuf_frames, read_offs_bytes, period_frames;
UINT64 last_time, written_frames;
AudioQueueBufferRef public_buffer;
UINT32 getbuf_last;
int playing;
BYTE *tmp_buffer, *capture_buf;
Float64 highest_sampletime, next_sampletime;
UINT32 period_ms, bufsize_frames, period_frames;
UINT64 written_frames;
UINT32 lcl_offs_frames, wri_offs_frames, held_frames, tmp_buffer_frames;
UINT32 cap_bufsize_frames, cap_offs_frames, cap_held_frames, wrap_bufsize_frames, resamp_bufsize_frames;
INT32 getbuf_last;
BOOL playing;
BYTE *cap_buffer, *wrap_buffer, *resamp_buffer, *local_buffer, *tmp_buffer;
AudioSession *session;
AudioSessionWrapper *session_wrapper;
struct list entry;
struct list avail_buffers;
struct list queued_buffers; /* either in avail, queued or public_buffer */
struct list queued_bufinfos;
OSSpinLock lock;
};
enum PlayingStates {
StateStopped = 0,
StatePlaying,
StateInTransition
};
static const IAudioClientVtbl AudioClient_Vtbl;
static const IAudioRenderClientVtbl AudioRenderClient_Vtbl;
static const IAudioCaptureClientVtbl AudioCaptureClient_Vtbl;
......@@ -593,6 +607,62 @@ static BOOL get_deviceid_by_guid(GUID *guid, AudioDeviceID *id, EDataFlow *flow)
return FALSE;
}
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: %lx\n", 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: %lx\n", 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: %lx\n", 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;
}
HRESULT WINAPI AUDDRV_GetAudioEndpoint(GUID *guid, IMMDevice *dev, IAudioClient **out)
{
ACImpl *This;
......@@ -639,156 +709,19 @@ HRESULT WINAPI AUDDRV_GetAudioEndpoint(GUID *guid, IMMDevice *dev, IAudioClient
This->parent = dev;
IMMDevice_AddRef(This->parent);
list_init(&This->avail_buffers);
list_init(&This->queued_buffers);
list_init(&This->queued_bufinfos);
This->adevid = adevid;
if(!(This->unit = get_audiounit(This->dataflow, This->adevid))){
HeapFree(GetProcessHeap(), 0, This);
return AUDCLNT_E_DEVICE_INVALIDATED;
}
*out = &This->IAudioClient_iface;
IAudioClient_AddRef(&This->IAudioClient_iface);
return S_OK;
}
/* current position from start of stream */
#define BUFPOS_ABSOLUTE 1
/* current position from start of this buffer */
#define BUFPOS_RELATIVE 2
static UINT64 get_current_aqbuffer_position(ACImpl *This, int mode)
{
struct list *head;
QueuedBufInfo *bufinfo;
UINT64 ret;
head = list_head(&This->queued_bufinfos);
if(!head){
TRACE("No buffers queued\n");
if(mode == BUFPOS_ABSOLUTE)
return This->written_frames;
return 0;
}
bufinfo = LIST_ENTRY(head, QueuedBufInfo, entry);
if(This->playing == StatePlaying){
AudioTimeStamp tstamp;
OSStatus sc;
/* AudioQueueGetCurrentTime() is brain damaged. The returned
* mSampleTime member jumps backwards seemingly at random, so
* we record the highest sampletime and use that during these
* anomalies.
*
* It also behaves poorly when the queue is paused, jumping
* forwards during the pause and backwards again after resuming.
* So we record the sampletime when the queue is paused and use
* that. */
sc = AudioQueueGetCurrentTime(This->aqueue, NULL, &tstamp, NULL);
if(sc != noErr){
if(sc != kAudioQueueErr_InvalidRunState)
WARN("Unable to get current time: %lx\n", sc);
if(mode == BUFPOS_ABSOLUTE)
return This->highest_sampletime;
return 0;
}
if(!(tstamp.mFlags & kAudioTimeStampSampleTimeValid)){
FIXME("SampleTime not valid: %lx\n", tstamp.mFlags);
return 0;
}
if(tstamp.mSampleTime > This->highest_sampletime)
This->highest_sampletime = tstamp.mSampleTime;
}
while(This->highest_sampletime > bufinfo->start_sampletime + bufinfo->len_frames){
This->inbuf_frames -= bufinfo->len_frames;
list_remove(&bufinfo->entry);
HeapFree(GetProcessHeap(), 0, bufinfo);
head = list_head(&This->queued_bufinfos);
if(!head){
TRACE("No buffers queued\n");
if(mode == BUFPOS_ABSOLUTE)
return This->written_frames;
return 0;
}
bufinfo = LIST_ENTRY(head, QueuedBufInfo, entry);
}
if(This->highest_sampletime < bufinfo->start_sampletime)
ret = 0;
else
ret = This->highest_sampletime - bufinfo->start_sampletime;
if(mode == BUFPOS_ABSOLUTE){
ret = This->written_frames - (bufinfo->len_frames - ret);
while((head = list_next(&This->queued_bufinfos, &bufinfo->entry))){
bufinfo = LIST_ENTRY(head, QueuedBufInfo, entry);
ret -= bufinfo->len_frames;
}
}
TRACE("%llu frames (%s)\n", ret,
mode == BUFPOS_ABSOLUTE ? "absolute" : "relative");
return ret;
}
static void avail_update(ACImpl *This)
{
AQBuffer *buf, *next;
OSStatus sc;
if(This->dataflow == eCapture){
DWORD bufsize_bytes = This->bufsize_frames * This->fmt->nBlockAlign;
DWORD inbuf_bytes = This->inbuf_frames * This->fmt->nBlockAlign;
LIST_FOR_EACH_ENTRY_SAFE(buf, next, &This->queued_buffers, AQBuffer, entry){
DWORD buffer_bytes = buf->buf->mAudioDataByteSize, to_copy_bytes;
if(buf->used)
break;
to_copy_bytes = bufsize_bytes - (This->read_offs_bytes + inbuf_bytes) % bufsize_bytes;
if(buffer_bytes <= to_copy_bytes){
memcpy(This->capture_buf + (This->read_offs_bytes + inbuf_bytes) % bufsize_bytes,
buf->buf->mAudioData, buffer_bytes);
}else{
memcpy(This->capture_buf + (This->read_offs_bytes + inbuf_bytes) % bufsize_bytes,
buf->buf->mAudioData, to_copy_bytes);
memcpy(This->capture_buf, ((char *)buf->buf->mAudioData) + to_copy_bytes,
buffer_bytes - to_copy_bytes);
}
if(inbuf_bytes + buffer_bytes > bufsize_bytes){
This->read_offs_bytes += inbuf_bytes + buffer_bytes;
This->read_offs_bytes %= bufsize_bytes;
inbuf_bytes = bufsize_bytes;
}else
inbuf_bytes += buffer_bytes;
buf->used = TRUE;
list_remove(&buf->entry);
list_add_tail(&This->queued_buffers, &buf->entry);
sc = AudioQueueEnqueueBuffer(This->aqueue, buf->buf, 0, NULL);
if(sc != noErr)
WARN("EnqueueBuffer gave: %lx\n", sc);
}
This->inbuf_frames = inbuf_bytes / This->fmt->nBlockAlign;
}else{
LIST_FOR_EACH_ENTRY_SAFE(buf, next, &This->queued_buffers, AQBuffer, entry){
if(buf->used)
break;
list_remove(&buf->entry);
list_add_tail(&This->avail_buffers, &buf->entry);
}
}
}
static HRESULT WINAPI AudioClient_QueryInterface(IAudioClient *iface,
REFIID riid, void **ppv)
{
......@@ -837,31 +770,10 @@ static ULONG WINAPI AudioClient_Release(IAudioClient *iface)
WaitForSingleObject(event, INFINITE);
CloseHandle(event);
}
if(This->aqueue){
AQBuffer *buf, *next;
QueuedBufInfo *bufinfo, *bufinfo2;
if(This->public_buffer){
buf = This->public_buffer->mUserData;
list_add_tail(&This->avail_buffers, &buf->entry);
}
IAudioClient_Stop(iface);
AudioQueueStop(This->aqueue, 1);
/* Stopped synchronously, all buffers returned. */
list_move_tail(&This->avail_buffers, &This->queued_buffers);
LIST_FOR_EACH_ENTRY_SAFE(buf, next, &This->avail_buffers, AQBuffer, entry){
AudioQueueFreeBuffer(This->aqueue, buf->buf);
HeapFree(GetProcessHeap(), 0, buf);
}
LIST_FOR_EACH_ENTRY_SAFE(bufinfo, bufinfo2, &This->queued_bufinfos,
QueuedBufInfo, entry)
HeapFree(GetProcessHeap(), 0, bufinfo);
AudioQueueDispose(This->aqueue, 1);
}
AudioOutputUnitStop(This->unit);
AudioComponentInstanceDispose(This->unit);
if(This->converter)
AudioConverterDispose(This->converter);
if(This->session){
EnterCriticalSection(&g_sessions_lock);
list_remove(&This->entry);
......@@ -869,7 +781,10 @@ static ULONG WINAPI AudioClient_Release(IAudioClient *iface)
}
HeapFree(GetProcessHeap(), 0, This->vols);
HeapFree(GetProcessHeap(), 0, This->tmp_buffer);
HeapFree(GetProcessHeap(), 0, This->capture_buf);
HeapFree(GetProcessHeap(), 0, This->cap_buffer);
HeapFree(GetProcessHeap(), 0, This->local_buffer);
free(This->wrap_buffer);
HeapFree(GetProcessHeap(), 0, This->resamp_buffer);
CoTaskMemFree(This->fmt);
IMMDevice_Release(This->parent);
IUnknown_Release(This->pUnkFTMarshal);
......@@ -999,84 +914,6 @@ static HRESULT ca_get_audiodesc(AudioStreamBasicDescription *desc,
return S_OK;
}
/* We can't use debug printing or {Enter,Leave}CriticalSection from
* OSX callback threads. We may use OSSpinLock.
* OSSpinLock is not a recursive lock, so don't call
* synchronized functions while holding the lock. */
static void ca_out_buffer_cb(void *user, AudioQueueRef aqueue,
AudioQueueBufferRef buffer)
{
AQBuffer *buf = buffer->mUserData;
buf->used = FALSE;
}
static void ca_in_buffer_cb(void *user, AudioQueueRef aqueue,
AudioQueueBufferRef buffer, const AudioTimeStamp *start,
UInt32 ndesc, const AudioStreamPacketDescription *descs)
{
AQBuffer *buf = buffer->mUserData;
buf->used = FALSE;
/* let's update inbuf_frames synchronously without OSAddAtomic */
}
static HRESULT ca_setup_aqueue(AudioDeviceID did, EDataFlow flow,
const WAVEFORMATEX *fmt, void *user, AudioQueueRef *aqueue)
{
AudioStreamBasicDescription desc;
AudioObjectPropertyAddress addr;
CFStringRef uid;
OSStatus sc;
HRESULT hr;
UInt32 size;
addr.mScope = kAudioObjectPropertyScopeGlobal;
addr.mElement = 0;
addr.mSelector = kAudioDevicePropertyDeviceUID;
size = sizeof(uid);
sc = AudioObjectGetPropertyData(did, &addr, 0, NULL, &size, &uid);
if(sc != noErr){
WARN("Unable to get _DeviceUID property: %lx\n", sc);
return osstatus_to_hresult(sc);
}
hr = ca_get_audiodesc(&desc, fmt);
if(FAILED(hr)){
CFRelease(uid);
return hr;
}
if(flow == eRender)
sc = AudioQueueNewOutput(&desc, ca_out_buffer_cb, user, NULL, NULL, 0,
aqueue);
else if(flow == eCapture)
sc = AudioQueueNewInput(&desc, ca_in_buffer_cb, user, NULL, NULL, 0,
aqueue);
else{
CFRelease(uid);
return E_UNEXPECTED;
}
if(sc != noErr){
WARN("Unable to create AudioQueue: %lx\n", sc);
CFRelease(uid);
return osstatus_to_hresult(sc);
}
sc = AudioQueueSetProperty(*aqueue, kAudioQueueProperty_CurrentDevice,
&uid, sizeof(uid));
if(sc != noErr){
WARN("Unable to change AQueue device: %lx\n", sc);
CFRelease(uid);
return osstatus_to_hresult(sc);
}
CFRelease(uid);
return S_OK;
}
static void session_init_vols(AudioSession *session, UINT channels)
{
if(session->channel_count < channels){
......@@ -1159,6 +996,299 @@ static HRESULT get_audio_session(const GUID *sessionguid,
return S_OK;
}
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);
}
static void silence_buffer(ACImpl *This, BYTE *buffer, UINT32 frames)
{
WAVEFORMATEXTENSIBLE *fmtex = (WAVEFORMATEXTENSIBLE*)This->fmt;
if((This->fmt->wFormatTag == WAVE_FORMAT_PCM ||
(This->fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))) &&
This->fmt->wBitsPerSample == 8)
memset(buffer, 128, frames * This->fmt->nBlockAlign);
else
memset(buffer, 0, frames * This->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)
{
ACImpl *This = user;
UINT32 to_copy_bytes, to_copy_frames, chunk_bytes, lcl_offs_bytes;
OSSpinLockLock(&This->lock);
if(This->playing){
lcl_offs_bytes = This->lcl_offs_frames * This->fmt->nBlockAlign;
to_copy_frames = min(nframes, This->held_frames);
to_copy_bytes = to_copy_frames * This->fmt->nBlockAlign;
chunk_bytes = (This->bufsize_frames - This->lcl_offs_frames) * This->fmt->nBlockAlign;
if(to_copy_bytes > chunk_bytes){
memcpy(data->mBuffers[0].mData, This->local_buffer + lcl_offs_bytes, chunk_bytes);
memcpy(((BYTE *)data->mBuffers[0].mData) + chunk_bytes, This->local_buffer, to_copy_bytes - chunk_bytes);
}else
memcpy(data->mBuffers[0].mData, This->local_buffer + lcl_offs_bytes, to_copy_bytes);
This->lcl_offs_frames += to_copy_frames;
This->lcl_offs_frames %= This->bufsize_frames;
This->held_frames -= to_copy_frames;
}else
to_copy_bytes = to_copy_frames = 0;
if(nframes > to_copy_frames)
silence_buffer(This, ((BYTE *)data->mBuffers[0].mData) + to_copy_bytes, nframes - to_copy_frames);
OSSpinLockUnlock(&This->lock);
return noErr;
}
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)
{
ACImpl *This = user;
*nframes = min(*nframes, This->cap_held_frames);
if(!*nframes){
data->mBuffers[0].mData = NULL;
data->mBuffers[0].mDataByteSize = 0;
data->mBuffers[0].mNumberChannels = This->fmt->nChannels;
return noErr;
}
data->mBuffers[0].mDataByteSize = *nframes * This->fmt->nBlockAlign;
data->mBuffers[0].mNumberChannels = This->fmt->nChannels;
if(This->cap_offs_frames + *nframes > This->cap_bufsize_frames){
UINT32 chunk_frames = This->cap_bufsize_frames - This->cap_offs_frames;
if(This->wrap_bufsize_frames < *nframes){
free(This->wrap_buffer);
This->wrap_buffer = malloc(data->mBuffers[0].mDataByteSize);
This->wrap_bufsize_frames = *nframes;
}
memcpy(This->wrap_buffer, This->cap_buffer + This->cap_offs_frames * This->fmt->nBlockAlign,
chunk_frames * This->fmt->nBlockAlign);
memcpy(This->wrap_buffer + chunk_frames * This->fmt->nBlockAlign, This->cap_buffer,
(*nframes - chunk_frames) * This->fmt->nBlockAlign);
data->mBuffers[0].mData = This->wrap_buffer;
}else
data->mBuffers[0].mData = This->cap_buffer + This->cap_offs_frames * This->fmt->nBlockAlign;
This->cap_offs_frames += *nframes;
This->cap_offs_frames %= This->cap_bufsize_frames;
This->cap_held_frames -= *nframes;
if(packets)
*packets = NULL;
return noErr;
}
static void capture_resample(ACImpl *This)
{
UINT32 resamp_period_frames = MulDiv(This->period_frames, This->dev_desc.mSampleRate, This->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(This->cap_held_frames > resamp_period_frames * 2){
AudioBufferList converted_list;
UInt32 wanted_frames = This->period_frames;
converted_list.mNumberBuffers = 1;
converted_list.mBuffers[0].mNumberChannels = This->fmt->nChannels;
converted_list.mBuffers[0].mDataByteSize = wanted_frames * This->fmt->nBlockAlign;
if(This->resamp_bufsize_frames < wanted_frames){
HeapFree(GetProcessHeap(), 0, This->resamp_buffer);
This->resamp_buffer = HeapAlloc(GetProcessHeap(), 0, converted_list.mBuffers[0].mDataByteSize);
This->resamp_bufsize_frames = wanted_frames;
}
converted_list.mBuffers[0].mData = This->resamp_buffer;
sc = AudioConverterFillComplexBuffer(This->converter, feed_cb,
This, &wanted_frames, &converted_list, NULL);
if(sc != noErr){
WARN("AudioConverterFillComplexBuffer failed: %lx\n", sc);
break;
}
ca_wrap_buffer(This->local_buffer,
This->wri_offs_frames * This->fmt->nBlockAlign,
This->bufsize_frames * This->fmt->nBlockAlign,
This->resamp_buffer, wanted_frames * This->fmt->nBlockAlign);
This->wri_offs_frames += wanted_frames;
This->wri_offs_frames %= This->bufsize_frames;
if(This->held_frames + wanted_frames > This->bufsize_frames){
This->lcl_offs_frames += buf_ptr_diff(This->lcl_offs_frames,
This->wri_offs_frames, This->bufsize_frames);
This->held_frames = This->bufsize_frames;
}else
This->held_frames += wanted_frames;
}
}
/* 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)
{
ACImpl *This = user;
AudioBufferList list;
OSStatus sc;
UINT32 cap_wri_offs_frames;
OSSpinLockLock(&This->lock);
cap_wri_offs_frames = (This->cap_offs_frames + This->cap_held_frames) % This->cap_bufsize_frames;
list.mNumberBuffers = 1;
list.mBuffers[0].mNumberChannels = This->fmt->nChannels;
list.mBuffers[0].mDataByteSize = nframes * This->fmt->nBlockAlign;
if(!This->playing || cap_wri_offs_frames + nframes > This->cap_bufsize_frames){
if(This->wrap_bufsize_frames < nframes){
free(This->wrap_buffer);
This->wrap_buffer = malloc(list.mBuffers[0].mDataByteSize);
This->wrap_bufsize_frames = nframes;
}
list.mBuffers[0].mData = This->wrap_buffer;
}else
list.mBuffers[0].mData = This->cap_buffer + cap_wri_offs_frames * This->fmt->nBlockAlign;
sc = AudioUnitRender(This->unit, flags, ts, bus, nframes, &list);
if(sc != noErr){
OSSpinLockUnlock(&This->lock);
return sc;
}
if(This->playing){
if(list.mBuffers[0].mData == This->wrap_buffer){
ca_wrap_buffer(This->cap_buffer,
cap_wri_offs_frames * This->fmt->nBlockAlign,
This->cap_bufsize_frames * This->fmt->nBlockAlign,
This->wrap_buffer, list.mBuffers[0].mDataByteSize);
}
This->cap_held_frames += list.mBuffers[0].mDataByteSize / This->fmt->nBlockAlign;
if(This->cap_held_frames > This->cap_bufsize_frames){
This->cap_offs_frames += This->cap_held_frames % This->cap_bufsize_frames;
This->cap_offs_frames %= This->cap_bufsize_frames;
This->cap_held_frames = This->cap_bufsize_frames;
}
}
OSSpinLockUnlock(&This->lock);
return noErr;
}
static void dump_adesc(const char *aux, AudioStreamBasicDescription *desc)
{
TRACE("%s: mSampleRate: %f\n", aux, desc->mSampleRate);
TRACE("%s: mBytesPerPacket: %ld\n", aux, desc->mBytesPerPacket);
TRACE("%s: mFramesPerPacket: %ld\n", aux, desc->mFramesPerPacket);
TRACE("%s: mBytesPerFrame: %ld\n", aux, desc->mBytesPerFrame);
TRACE("%s: mChannelsPerFrame: %ld\n", aux, desc->mChannelsPerFrame);
TRACE("%s: mBitsPerChannel: %ld\n", aux, desc->mBitsPerChannel);
}
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;
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: %lx\n", 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: %lx\n", sc);
return osstatus_to_hresult(sc);
}
sc = AudioConverterNew(dev_desc, &desc, converter);
if(sc != noErr){
WARN("Couldn't create audio converter: %lx\n", 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: %lx\n", sc);
return osstatus_to_hresult(sc);
}
}
return S_OK;
}
static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
AUDCLNT_SHAREMODE mode, DWORD flags, REFERENCE_TIME duration,
REFERENCE_TIME period, const WAVEFORMATEX *fmt,
......@@ -1222,21 +1352,13 @@ static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
OSSpinLockLock(&This->lock);
if(This->aqueue){
if(This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_ALREADY_INITIALIZED;
}
hr = ca_setup_aqueue(This->adevid, This->dataflow, fmt, This, &This->aqueue);
if(FAILED(hr)){
OSSpinLockUnlock(&This->lock);
return hr;
}
This->fmt = clone_format(fmt);
if(!This->fmt){
AudioQueueDispose(This->aqueue, 1);
This->aqueue = NULL;
OSSpinLockUnlock(&This->lock);
return E_OUTOFMEMORY;
}
......@@ -1248,53 +1370,88 @@ static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
if(mode == AUDCLNT_SHAREMODE_EXCLUSIVE)
This->bufsize_frames -= This->bufsize_frames % This->period_frames;
hr = ca_setup_audiounit(This->dataflow, This->unit, This->fmt, &This->dev_desc, &This->converter);
if(FAILED(hr)){
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return hr;
}
if(This->dataflow == eCapture){
int i, nbuffs = (This->bufsize_frames / This->period_frames) + 1;
AURenderCallbackStruct input;
This->capture_buf = HeapAlloc(GetProcessHeap(), 0, This->bufsize_frames * This->fmt->nBlockAlign);
memset(&input, 0, sizeof(input));
input.inputProc = &ca_capture_cb;
input.inputProcRefCon = This;
for(i = 0; i < nbuffs; ++i){
AQBuffer *buf;
sc = AudioUnitSetProperty(This->unit, kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Output, 1, &input, sizeof(input));
if(sc != noErr){
WARN("Couldn't set callback: %lx\n", sc);
AudioConverterDispose(This->converter);
This->converter = NULL;
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return osstatus_to_hresult(sc);
}
}else{
AURenderCallbackStruct input;
buf = HeapAlloc(GetProcessHeap(), 0, sizeof(AQBuffer));
if(!buf){
HeapFree(GetProcessHeap(), 0, This->capture_buf);
AudioQueueDispose(This->aqueue, 1);
This->aqueue = NULL;
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return E_OUTOFMEMORY;
}
memset(&input, 0, sizeof(input));
input.inputProc = &ca_render_cb;
input.inputProcRefCon = This;
sc = AudioQueueAllocateBuffer(This->aqueue, This->period_frames * This->fmt->nBlockAlign, &buf->buf);
if(sc != noErr){
HeapFree(GetProcessHeap(), 0, This->capture_buf);
AudioQueueDispose(This->aqueue, 1);
This->aqueue = NULL;
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
WARN("Couldn't allocate buffer: %lx\n", sc);
return osstatus_to_hresult(sc);
}
sc = AudioUnitSetProperty(This->unit, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &input, sizeof(input));
if(sc != noErr){
WARN("Couldn't set callback: %lx\n", sc);
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return osstatus_to_hresult(sc);
}
}
buf->buf->mUserData = buf;
buf->used = TRUE;
sc = AudioQueueEnqueueBuffer(This->aqueue, buf->buf, 0, NULL);
if(sc != noErr){
ERR("Couldn't enqueue buffer: %lx\n", sc);
break;
}
list_add_tail(&This->queued_buffers, &buf->entry);
sc = AudioUnitInitialize(This->unit);
if(sc != noErr){
WARN("Couldn't initialize: %lx\n", sc);
if(This->converter){
AudioConverterDispose(This->converter);
This->converter = NULL;
}
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return osstatus_to_hresult(sc);
}
/* we play audio continuously because AudioOutputUnitStart sometimes takes
* a while to return */
sc = AudioOutputUnitStart(This->unit);
if(sc != noErr){
WARN("Unit failed to start: %lx\n", sc);
if(This->converter){
AudioConverterDispose(This->converter);
This->converter = NULL;
}
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
return osstatus_to_hresult(sc);
}
This->local_buffer = HeapAlloc(GetProcessHeap(), 0, This->bufsize_frames * fmt->nBlockAlign);
silence_buffer(This, This->local_buffer, This->bufsize_frames);
if(This->dataflow == eCapture){
This->cap_bufsize_frames = MulDiv(duration, This->dev_desc.mSampleRate, 10000000);
This->cap_buffer = HeapAlloc(GetProcessHeap(), 0, This->cap_bufsize_frames * This->fmt->nBlockAlign);
}
This->vols = HeapAlloc(GetProcessHeap(), 0, fmt->nChannels * sizeof(float));
if(!This->vols){
HeapFree(GetProcessHeap(), 0, This->capture_buf);
AudioQueueDispose(This->aqueue, 1);
This->aqueue = NULL;
CoTaskMemFree(This->fmt);
This->fmt = NULL;
OSSpinLockUnlock(&This->lock);
......@@ -1313,11 +1470,8 @@ static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
&This->session);
if(FAILED(hr)){
LeaveCriticalSection(&g_sessions_lock);
AudioQueueDispose(This->aqueue, 1);
This->aqueue = NULL;
CoTaskMemFree(This->fmt);
This->fmt = NULL;
HeapFree(GetProcessHeap(), 0, This->capture_buf);
HeapFree(GetProcessHeap(), 0, This->vols);
This->vols = NULL;
OSSpinLockUnlock(&This->lock);
......@@ -1330,6 +1484,8 @@ static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
ca_setvol(This, -1);
This->initted = TRUE;
OSSpinLockUnlock(&This->lock);
return S_OK;
......@@ -1347,7 +1503,7 @@ static HRESULT WINAPI AudioClient_GetBufferSize(IAudioClient *iface,
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
......@@ -1429,7 +1585,7 @@ static HRESULT WINAPI AudioClient_GetStreamLatency(IAudioClient *iface,
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
......@@ -1467,20 +1623,13 @@ static HRESULT WINAPI AudioClient_GetStreamLatency(IAudioClient *iface,
static HRESULT AudioClient_GetCurrentPadding_nolock(ACImpl *This,
UINT32 *numpad)
{
if(!This->aqueue)
if(!This->initted)
return AUDCLNT_E_NOT_INITIALIZED;
avail_update(This);
if(This->dataflow == eCapture)
capture_resample(This);
if(This->dataflow == eRender){
UINT64 bufpos;
bufpos = get_current_aqbuffer_position(This, BUFPOS_RELATIVE);
*numpad = This->inbuf_frames - bufpos;
}else{
*numpad = This->inbuf_frames;
if(*numpad < This->period_frames)
*numpad = 0;
}
*numpad = This->held_frames;
return S_OK;
}
......@@ -1510,8 +1659,10 @@ static HRESULT WINAPI AudioClient_IsFormatSupported(IAudioClient *iface,
WAVEFORMATEX **outpwfx)
{
ACImpl *This = impl_from_IAudioClient(iface);
AudioStreamBasicDescription dev_desc;
AudioConverterRef converter;
AudioComponentInstance unit;
WAVEFORMATEXTENSIBLE *fmtex = (WAVEFORMATEXTENSIBLE*)pwfx;
AudioQueueRef aqueue;
HRESULT hr;
TRACE("(%p)->(%x, %p, %p)\n", This, mode, pwfx, outpwfx);
......@@ -1555,20 +1706,18 @@ static HRESULT WINAPI AudioClient_IsFormatSupported(IAudioClient *iface,
if(pwfx->nChannels == 0)
return AUDCLNT_E_UNSUPPORTED_FORMAT;
OSSpinLockLock(&This->lock);
unit = get_audiounit(This->dataflow, This->adevid);
hr = ca_setup_aqueue(This->adevid, This->dataflow, pwfx, NULL, &aqueue);
if(SUCCEEDED(hr)){
AudioQueueDispose(aqueue, 1);
OSSpinLockUnlock(&This->lock);
TRACE("returning %08x\n", S_OK);
return S_OK;
}
OSSpinLockUnlock(&This->lock);
if(hr != AUDCLNT_E_UNSUPPORTED_FORMAT){
TRACE("returning %08x\n", hr);
return hr;
}
converter = NULL;
hr = ca_setup_audiounit(This->dataflow, unit, pwfx, &dev_desc, &converter);
AudioComponentInstanceDispose(unit);
if(FAILED(hr))
goto unsupported;
if(converter)
AudioConverterDispose(converter);
return S_OK;
unsupported:
if(outpwfx){
......@@ -1695,18 +1844,17 @@ void CALLBACK ca_period_cb(void *user, BOOLEAN timer)
static HRESULT WINAPI AudioClient_Start(IAudioClient *iface)
{
ACImpl *This = impl_from_IAudioClient(iface);
OSStatus sc;
TRACE("(%p)\n", This);
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
if(This->playing != StateStopped){
if(This->playing){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_STOPPED;
}
......@@ -1725,19 +1873,7 @@ static HRESULT WINAPI AudioClient_Start(IAudioClient *iface)
return E_OUTOFMEMORY;
}
/* enqueue buffers */
avail_update(This);
This->playing = StateInTransition;
sc = AudioQueueStart(This->aqueue, NULL);
if(sc != noErr){
OSSpinLockUnlock(&This->lock);
WARN("Unable to start audio queue: %lx\n", sc);
return osstatus_to_hresult(sc);
}
This->playing = StatePlaying;
This->playing = TRUE;
OSSpinLockUnlock(&This->lock);
......@@ -1747,49 +1883,22 @@ static HRESULT WINAPI AudioClient_Start(IAudioClient *iface)
static HRESULT WINAPI AudioClient_Stop(IAudioClient *iface)
{
ACImpl *This = impl_from_IAudioClient(iface);
AudioTimeStamp tstamp;
OSStatus sc;
TRACE("(%p)\n", This);
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
if(This->playing == StateStopped){
if(!This->playing){
OSSpinLockUnlock(&This->lock);
return S_FALSE;
}
if(This->playing == StateInTransition){
OSSpinLockUnlock(&This->lock);
return S_OK;
}
This->playing = StateInTransition;
sc = AudioQueueGetCurrentTime(This->aqueue, NULL, &tstamp, NULL);
if(sc == noErr){
if(tstamp.mFlags & kAudioTimeStampSampleTimeValid){
if(tstamp.mSampleTime > This->highest_sampletime)
This->highest_sampletime = tstamp.mSampleTime;
}else
WARN("Returned tstamp mSampleTime not valid: %lx\n", tstamp.mFlags);
}else
WARN("GetCurrentTime failed: %lx\n", sc);
/* Mac OS bug? Our capture callback is no more called past AQStop */
sc = AudioQueuePause(This->aqueue);
if(sc != noErr){
OSSpinLockUnlock(&This->lock);
WARN("Unable to pause audio queue: %lx\n", sc);
return osstatus_to_hresult(sc);
}
This->playing = StateStopped;
This->playing = FALSE;
OSSpinLockUnlock(&This->lock);
......@@ -1799,19 +1908,17 @@ static HRESULT WINAPI AudioClient_Stop(IAudioClient *iface)
static HRESULT WINAPI AudioClient_Reset(IAudioClient *iface)
{
ACImpl *This = impl_from_IAudioClient(iface);
OSStatus sc;
QueuedBufInfo *bufinfo, *bufinfo2;
TRACE("(%p)\n", This);
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
if(This->playing != StateStopped){
if(This->playing){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_STOPPED;
}
......@@ -1821,27 +1928,17 @@ static HRESULT WINAPI AudioClient_Reset(IAudioClient *iface)
return AUDCLNT_E_BUFFER_OPERATION_PENDING;
}
avail_update(This); /* going to skip over inbuf_frames */
LIST_FOR_EACH_ENTRY_SAFE(bufinfo, bufinfo2, &This->queued_bufinfos,
QueuedBufInfo, entry){
list_remove(&bufinfo->entry);
HeapFree(GetProcessHeap(), 0, bufinfo);
}
sc = AudioQueueReset(This->aqueue);
if(sc != noErr){
OSSpinLockUnlock(&This->lock);
WARN("Unable to reset audio queue: %lx\n", sc);
return osstatus_to_hresult(sc);
}
if(This->dataflow == eRender){
This->written_frames = 0;
}else{
This->written_frames += This->inbuf_frames;
This->written_frames += This->held_frames;
}
This->inbuf_frames = 0;
This->held_frames = 0;
This->lcl_offs_frames = 0;
This->wri_offs_frames = 0;
This->cap_offs_frames = 0;
This->cap_held_frames = 0;
OSSpinLockUnlock(&This->lock);
......@@ -1860,7 +1957,7 @@ static HRESULT WINAPI AudioClient_SetEventHandle(IAudioClient *iface,
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
......@@ -1896,7 +1993,7 @@ static HRESULT WINAPI AudioClient_GetService(IAudioClient *iface, REFIID riid,
OSSpinLockLock(&This->lock);
if(!This->aqueue){
if(!This->initted){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_NOT_INITIALIZED;
}
......@@ -2023,26 +2120,12 @@ static ULONG WINAPI AudioRenderClient_Release(IAudioRenderClient *iface)
return AudioClient_Release(&This->IAudioClient_iface);
}
static void silence_buffer(ACImpl *This, BYTE *buffer, UINT32 frames)
{
WAVEFORMATEXTENSIBLE *fmtex = (WAVEFORMATEXTENSIBLE*)This->fmt;
if((This->fmt->wFormatTag == WAVE_FORMAT_PCM ||
(This->fmt->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
IsEqualGUID(&fmtex->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))) &&
This->fmt->wBitsPerSample == 8)
memset(buffer, 128, frames * This->fmt->nBlockAlign);
else
memset(buffer, 0, frames * This->fmt->nBlockAlign);
}
static HRESULT WINAPI AudioRenderClient_GetBuffer(IAudioRenderClient *iface,
UINT32 frames, BYTE **data)
{
ACImpl *This = impl_from_IAudioRenderClient(iface);
AQBuffer *buf;
UINT32 pad, bytes;
UINT32 pad;
HRESULT hr;
OSStatus sc;
TRACE("(%p)->(%u, %p)\n", This, frames, data);
......@@ -2073,38 +2156,23 @@ static HRESULT WINAPI AudioRenderClient_GetBuffer(IAudioRenderClient *iface,
return AUDCLNT_E_BUFFER_TOO_LARGE;
}
bytes = frames * This->fmt->nBlockAlign;
LIST_FOR_EACH_ENTRY(buf, &This->avail_buffers, AQBuffer, entry){
if(buf->buf->mAudioDataBytesCapacity >= bytes){
This->public_buffer = buf->buf;
list_remove(&buf->entry);
break;
}
}
if(&buf->entry == &This->avail_buffers){
sc = AudioQueueAllocateBuffer(This->aqueue, bytes,
&This->public_buffer);
if(sc != noErr){
This->public_buffer = NULL;
OSSpinLockUnlock(&This->lock);
WARN("Unable to allocate buffer: %lx\n", sc);
return E_OUTOFMEMORY;
}
buf = HeapAlloc(GetProcessHeap(), 0, sizeof(AQBuffer));
if(!buf){
AudioQueueFreeBuffer(This->aqueue, This->public_buffer);
This->public_buffer = NULL;
OSSpinLockUnlock(&This->lock);
return E_OUTOFMEMORY;
if(This->wri_offs_frames + frames > This->bufsize_frames){
if(This->tmp_buffer_frames < frames){
HeapFree(GetProcessHeap(), 0, This->tmp_buffer);
This->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, frames * This->fmt->nBlockAlign);
if(!This->tmp_buffer){
OSSpinLockUnlock(&This->lock);
return E_OUTOFMEMORY;
}
This->tmp_buffer_frames = frames;
}
buf->used = FALSE;
buf->buf = This->public_buffer;
This->public_buffer->mUserData = buf;
*data = This->tmp_buffer;
This->getbuf_last = -frames;
}else{
*data = This->local_buffer + This->wri_offs_frames * This->fmt->nBlockAlign;
This->getbuf_last = frames;
}
This->getbuf_last = frames;
*data = This->public_buffer->mAudioData;
silence_buffer(This, *data, frames);
OSSpinLockUnlock(&This->lock);
......@@ -2116,9 +2184,7 @@ static HRESULT WINAPI AudioRenderClient_ReleaseBuffer(
IAudioRenderClient *iface, UINT32 frames, DWORD flags)
{
ACImpl *This = impl_from_IAudioRenderClient(iface);
AQBuffer *buf;
AudioTimeStamp start_time, req_time = {0}, *passed_time = NULL;
OSStatus sc;
BYTE *buffer;
TRACE("(%p)->(%u, %x)\n", This, frames, flags);
......@@ -2126,11 +2192,6 @@ static HRESULT WINAPI AudioRenderClient_ReleaseBuffer(
if(!frames){
This->getbuf_last = 0;
if(This->public_buffer){
buf = This->public_buffer->mUserData;
list_add_head(&This->avail_buffers, &buf->entry);
This->public_buffer = NULL;
}
OSSpinLockUnlock(&This->lock);
return S_OK;
}
......@@ -2140,62 +2201,31 @@ static HRESULT WINAPI AudioRenderClient_ReleaseBuffer(
return AUDCLNT_E_OUT_OF_ORDER;
}
if(frames > This->getbuf_last){
if(frames > (This->getbuf_last >= 0 ? This->getbuf_last : -This->getbuf_last)){
OSSpinLockUnlock(&This->lock);
return AUDCLNT_E_INVALID_SIZE;
}
if(flags & AUDCLNT_BUFFERFLAGS_SILENT)
silence_buffer(This, This->public_buffer->mAudioData, frames);
This->public_buffer->mAudioDataByteSize = frames * This->fmt->nBlockAlign;
buf = This->public_buffer->mUserData;
buf->used = TRUE;
if(list_empty(&This->queued_bufinfos)){
sc = AudioQueueGetCurrentTime(This->aqueue, NULL, &req_time, NULL);
if(sc == noErr)
passed_time = &req_time;
else
TRACE("AudioQueueGetCurrentTime failed: %lx\n", sc);
}else{
req_time.mSampleTime = This->next_sampletime;
req_time.mFlags = kAudioTimeStampSampleTimeValid;
passed_time = &req_time;
}
sc = AudioQueueEnqueueBufferWithParameters(This->aqueue,
This->public_buffer, 0, NULL, 0, 0, 0, NULL, passed_time,
&start_time);
if(sc != noErr){
OSSpinLockUnlock(&This->lock);
ERR("Unable to enqueue buffer: %lx\n", sc);
return AUDCLNT_E_DEVICE_INVALIDATED;
}
list_add_tail(&This->queued_buffers, &buf->entry);
if(start_time.mFlags & kAudioTimeStampSampleTimeValid){
QueuedBufInfo *bufinfo;
if(This->getbuf_last >= 0)
buffer = This->local_buffer + This->wri_offs_frames * This->fmt->nBlockAlign;
else
buffer = This->tmp_buffer;
bufinfo = HeapAlloc(GetProcessHeap(), 0, sizeof(*bufinfo));
bufinfo->start_sampletime = start_time.mSampleTime;
bufinfo->start_pos = This->written_frames;
bufinfo->len_frames = frames;
if(flags & AUDCLNT_BUFFERFLAGS_SILENT)
silence_buffer(This, buffer, frames);
list_add_tail(&This->queued_bufinfos, &bufinfo->entry);
if(This->getbuf_last < 0)
ca_wrap_buffer(This->local_buffer,
This->wri_offs_frames * This->fmt->nBlockAlign,
This->bufsize_frames * This->fmt->nBlockAlign,
buffer, frames * This->fmt->nBlockAlign);
This->next_sampletime = start_time.mSampleTime + bufinfo->len_frames;
}else
WARN("Start time didn't contain valid SampleTime member\n");
if(This->playing == StateStopped)
AudioQueuePrime(This->aqueue, 0, NULL);
This->public_buffer = NULL;
This->getbuf_last = 0;
This->wri_offs_frames += frames;
This->wri_offs_frames %= This->bufsize_frames;
This->held_frames += frames;
This->written_frames += frames;
This->inbuf_frames += frames;
This->getbuf_last = 0;
OSSpinLockUnlock(&This->lock);
......@@ -2252,7 +2282,7 @@ static HRESULT WINAPI AudioCaptureClient_GetBuffer(IAudioCaptureClient *iface,
UINT64 *qpcpos)
{
ACImpl *This = impl_from_IAudioCaptureClient(iface);
DWORD chunk_bytes;
UINT32 chunk_bytes, chunk_frames;
TRACE("(%p)->(%p, %p, %p, %p, %p)\n", This, data, frames, flags,
devpos, qpcpos);
......@@ -2267,9 +2297,9 @@ static HRESULT WINAPI AudioCaptureClient_GetBuffer(IAudioCaptureClient *iface,
return AUDCLNT_E_OUT_OF_ORDER;
}
avail_update(This);
capture_resample(This);
if(This->inbuf_frames < This->period_frames){
if(This->held_frames < This->period_frames){
*frames = 0;
OSSpinLockUnlock(&This->lock);
return AUDCLNT_S_BUFFER_EMPTY;
......@@ -2277,15 +2307,16 @@ static HRESULT WINAPI AudioCaptureClient_GetBuffer(IAudioCaptureClient *iface,
*flags = 0;
chunk_bytes = This->bufsize_frames * This->fmt->nBlockAlign - This->read_offs_bytes;
if(chunk_bytes < This->period_frames * This->fmt->nBlockAlign){
chunk_frames = This->bufsize_frames - This->lcl_offs_frames;
if(chunk_frames < This->period_frames){
chunk_bytes = chunk_frames * This->fmt->nBlockAlign;
if(!This->tmp_buffer)
This->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, This->period_frames * This->fmt->nBlockAlign);
*data = This->tmp_buffer;
memcpy(*data, This->capture_buf + This->read_offs_bytes, chunk_bytes);
memcpy((*data) + chunk_bytes, This->capture_buf, This->period_frames * This->fmt->nBlockAlign - chunk_bytes);
memcpy(*data, This->local_buffer + This->lcl_offs_frames * This->fmt->nBlockAlign, chunk_bytes);
memcpy((*data) + chunk_bytes, This->local_buffer, This->period_frames * This->fmt->nBlockAlign - chunk_bytes);
}else
*data = This->capture_buf + This->read_offs_bytes;
*data = This->local_buffer + This->lcl_offs_frames * This->fmt->nBlockAlign;
This->getbuf_last = *frames = This->period_frames;
......@@ -2329,9 +2360,9 @@ static HRESULT WINAPI AudioCaptureClient_ReleaseBuffer(
}
This->written_frames += done;
This->inbuf_frames -= done;
This->read_offs_bytes += done * This->fmt->nBlockAlign;
This->read_offs_bytes %= This->bufsize_frames * This->fmt->nBlockAlign;
This->held_frames -= done;
This->lcl_offs_frames += done;
This->lcl_offs_frames %= This->bufsize_frames;
This->getbuf_last = 0;
OSSpinLockUnlock(&This->lock);
......@@ -2351,9 +2382,9 @@ static HRESULT WINAPI AudioCaptureClient_GetNextPacketSize(
OSSpinLockLock(&This->lock);
avail_update(This);
capture_resample(This);
if(This->inbuf_frames >= This->period_frames)
if(This->held_frames >= This->period_frames)
*frames = This->period_frames;
else
*frames = 0;
......@@ -2426,12 +2457,7 @@ static HRESULT WINAPI AudioClock_GetFrequency(IAudioClock *iface, UINT64 *freq)
static HRESULT AudioClock_GetPosition_nolock(ACImpl *This,
UINT64 *pos, UINT64 *qpctime)
{
avail_update(This);
if(This->dataflow == eRender)
*pos = get_current_aqbuffer_position(This, BUFPOS_ABSOLUTE);
else
*pos = This->inbuf_frames + This->written_frames;
*pos = This->written_frames - This->held_frames;
if(This->share == AUDCLNT_SHAREMODE_SHARED)
*pos *= This->fmt->nBlockAlign;
......@@ -2622,8 +2648,7 @@ static HRESULT WINAPI AudioSessionControl_GetState(IAudioSessionControl2 *iface,
LIST_FOR_EACH_ENTRY(client, &This->session->clients, ACImpl, entry){
OSSpinLockLock(&client->lock);
if(client->playing == StatePlaying ||
client->playing == StateInTransition){
if(client->playing){
*state = AudioSessionStateActive;
OSSpinLockUnlock(&client->lock);
LeaveCriticalSection(&g_sessions_lock);
......@@ -2799,30 +2824,30 @@ static const IAudioSessionControl2Vtbl AudioSessionControl2_Vtbl =
/* index == -1 means set all channels, otherwise sets only the given channel */
static HRESULT ca_setvol(ACImpl *This, UINT32 index)
{
float level;
Float32 level;
OSStatus sc;
if(index == (UINT32)-1){
HRESULT ret = S_OK;
UINT32 i;
for(i = 0; i < This->fmt->nChannels; ++i){
HRESULT hr;
hr = ca_setvol(This, i);
if(FAILED(hr))
ret = hr;
}
return ret;
}
if(This->session->mute)
level = 0;
else
level = This->session->master_vol *
This->session->channel_vols[index] * This->vols[index];
level = 0.;
else{
if(index == (UINT32)-1){
UINT32 i;
level = 1.;
for(i = 0; i < This->fmt->nChannels; ++i){
Float32 tmp;
tmp = This->session->master_vol *
This->session->channel_vols[i] * This->vols[i];
level = tmp < level ? tmp : level;
}
}else
level = This->session->master_vol *
This->session->channel_vols[index] * This->vols[index];
}
sc = AudioQueueSetParameter(This->aqueue, kAudioQueueParam_Volume, level);
sc = AudioUnitSetParameter(This->unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, level, 0);
if(sc != noErr)
WARN("Setting _Volume property failed: %lx\n", sc);
WARN("Couldn't set volume: %lx\n", sc);
return S_OK;
}
......@@ -3032,7 +3057,7 @@ static HRESULT WINAPI AudioStreamVolume_SetChannelVolume(
This->vols[index] = level;
WARN("AudioQueue doesn't support per-channel volume control\n");
WARN("CoreAudio doesn't support per-channel volume control\n");
ret = ca_setvol(This, index);
OSSpinLockUnlock(&This->lock);
......@@ -3194,7 +3219,7 @@ static HRESULT WINAPI ChannelAudioVolume_SetChannelVolume(
session->channel_vols[index] = level;
WARN("AudioQueue doesn't support per-channel volume control\n");
WARN("CoreAudio doesn't support per-channel volume control\n");
ret = ca_session_setvol(session, index);
LeaveCriticalSection(&session->lock);
......
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