[PATCH (try2)] winecoreaudio.drv: Use AUHAL API instead of AudioQueue
Andrew Eikum
aeikum at codeweavers.com
Tue Jul 1 14:53:45 CDT 2014
It turns out AudioQueue was a poor choice of API to base our driver
on. It did not have low-latency audio as a design goal:
http://lists.apple.com/archives/coreaudio-api/2008/Mar/msg00127.html
Specifically, the issue we were running into was capturing audio at
low sample rates (8 kHz, for example) would introduce a 0.5 second or
larger latency, which is unacceptable for voice communication.
So, this patch switches winecoreaudio to use the AUHAL API.
Thankfully, this also removes a bunch of hacks and workarounds we had
to do to get sane results from AudioQueue.
---
try2: Resample outside of the callback thread. Actually try to use the
given format in IsFormatSupported.
dlls/winecoreaudio.drv/mmdevdrv.c | 1151 +++++++++++++++++++------------------
1 file changed, 582 insertions(+), 569 deletions(-)
diff --git a/dlls/winecoreaudio.drv/mmdevdrv.c b/dlls/winecoreaudio.drv/mmdevdrv.c
index 7d8f4a0..0595051 100644
--- a/dlls/winecoreaudio.drv/mmdevdrv.c
+++ b/dlls/winecoreaudio.drv/mmdevdrv.c
@@ -54,8 +54,9 @@
#include <libkern/OSAtomic.h>
#include <CoreAudio/CoreAudio.h>
-#include <AudioToolbox/AudioQueue.h>
#include <AudioToolbox/AudioFormat.h>
+#include <AudioToolbox/AudioConverter.h>
+#include <AudioUnit/AudioUnit.h>
WINE_DEFAULT_DEBUG_CHANNEL(coreaudio);
@@ -64,19 +65,6 @@ WINE_DEFAULT_DEBUG_CHANNEL(coreaudio);
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 +116,29 @@ 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;
@@ -201,6 +181,28 @@ static struct list g_sessions = LIST_INIT(g_sessions);
static AudioSessionWrapper *AudioSessionWrapper_Create(ACImpl *client);
static HRESULT ca_setvol(ACImpl *This, UINT32 index);
+#ifndef HAVE_AUDIOUNIT_AUDIOCOMPONENT_H
+/* Define new AudioComponent Manager functions for OSX 10.5 */
+typedef Component AudioComponent;
+typedef ComponentDescription AudioComponentDescription;
+typedef ComponentInstance AudioComponentInstance;
+
+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
+
static inline ACImpl *impl_from_IAudioClient(IAudioClient *iface)
{
return CONTAINING_RECORD(iface, ACImpl, IAudioClient_iface);
@@ -593,6 +595,62 @@ 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 +697,19 @@ 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 +758,10 @@ 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 +769,10 @@ 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 +902,6 @@ 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 +984,299 @@ 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 +1340,13 @@ 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 +1358,88 @@ 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;
-
- This->capture_buf = HeapAlloc(GetProcessHeap(), 0, This->bufsize_frames * This->fmt->nBlockAlign);
-
- for(i = 0; i < nbuffs; ++i){
- AQBuffer *buf;
-
- 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;
- }
-
- 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);
- }
-
- 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);
+ AURenderCallbackStruct input;
+
+ memset(&input, 0, sizeof(input));
+ input.inputProc = &ca_capture_cb;
+ input.inputProcRefCon = This;
+
+ 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;
+
+ memset(&input, 0, sizeof(input));
+ input.inputProc = &ca_render_cb;
+ input.inputProcRefCon = This;
+
+ 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);
+ }
+ }
+
+ 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 +1458,8 @@ 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 +1472,8 @@ static HRESULT WINAPI AudioClient_Initialize(IAudioClient *iface,
ca_setvol(This, -1);
+ This->initted = TRUE;
+
OSSpinLockUnlock(&This->lock);
return S_OK;
@@ -1347,7 +1491,7 @@ 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 +1573,7 @@ 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 +1611,13 @@ 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 +1647,10 @@ 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 +1694,18 @@ 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 +1832,17 @@ 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 +1861,7 @@ 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 +1871,22 @@ 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 +1896,17 @@ 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 +1916,17 @@ 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 +1945,7 @@ 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 +1981,7 @@ 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 +2108,12 @@ 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 +2144,23 @@ 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(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;
}
+ *data = This->tmp_buffer;
+ This->getbuf_last = -frames;
+ }else{
+ *data = This->local_buffer + This->wri_offs_frames * This->fmt->nBlockAlign;
+ This->getbuf_last = frames;
}
- 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;
- }
- buf->used = FALSE;
- buf->buf = This->public_buffer;
- This->public_buffer->mUserData = buf;
- }
-
- This->getbuf_last = frames;
- *data = This->public_buffer->mAudioData;
silence_buffer(This, *data, frames);
OSSpinLockUnlock(&This->lock);
@@ -2116,9 +2172,7 @@ 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 +2180,6 @@ 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 +2189,31 @@ 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(This->getbuf_last >= 0)
+ buffer = This->local_buffer + This->wri_offs_frames * This->fmt->nBlockAlign;
+ else
+ buffer = This->tmp_buffer;
+
if(flags & AUDCLNT_BUFFERFLAGS_SILENT)
- silence_buffer(This, This->public_buffer->mAudioData, frames);
+ silence_buffer(This, buffer, frames);
- This->public_buffer->mAudioDataByteSize = frames * This->fmt->nBlockAlign;
+ 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);
- 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;
-
- bufinfo = HeapAlloc(GetProcessHeap(), 0, sizeof(*bufinfo));
- bufinfo->start_sampletime = start_time.mSampleTime;
- bufinfo->start_pos = This->written_frames;
- bufinfo->len_frames = frames;
-
- list_add_tail(&This->queued_bufinfos, &bufinfo->entry);
-
- 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 +2270,7 @@ 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 +2285,9 @@ 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 +2295,16 @@ 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 +2348,9 @@ 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 +2370,9 @@ 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 +2445,7 @@ 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 +2636,7 @@ 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 +2812,30 @@ 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 +3045,7 @@ 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 +3207,7 @@ 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);
--
2.0.1
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