winecoreaudio: Add a temporary capture_resample syscall.

Eventually everything that calls this will reside in the unixlib.

Signed-off-by: Huw Davies <huw@codeweavers.com>
Signed-off-by: Andrew Eikum <aeikum@codeweavers.com>
Signed-off-by: Alexandre Julliard <julliard@winehq.org>
This commit is contained in:
Huw Davies 2021-11-23 07:54:59 +00:00 committed by Alexandre Julliard
parent 75b57ea13d
commit de39001552
3 changed files with 108 additions and 101 deletions

View File

@ -693,6 +693,7 @@ static NTSTATUS release_stream( void *args )
}
if(stream->converter) AudioConverterDispose(stream->converter);
free(stream->resamp_buffer);
free(stream->wrap_buffer);
free(stream->cap_buffer);
if(stream->local_buffer)
@ -1012,6 +1013,108 @@ unsupported:
return STATUS_SUCCESS;
}
static UINT buf_ptr_diff(UINT left, UINT right, UINT bufsize)
{
if(left <= right)
return right - left;
return bufsize - (left - right);
}
/* place data from cap_buffer into provided AudioBufferList */
static OSStatus feed_cb(AudioConverterRef converter, UInt32 *nframes, AudioBufferList *data,
AudioStreamPacketDescription **packets, void *user)
{
struct coreaudio_stream *stream = user;
*nframes = min(*nframes, stream->cap_held_frames);
if(!*nframes){
data->mBuffers[0].mData = NULL;
data->mBuffers[0].mDataByteSize = 0;
data->mBuffers[0].mNumberChannels = stream->fmt->nChannels;
return noErr;
}
data->mBuffers[0].mDataByteSize = *nframes * stream->fmt->nBlockAlign;
data->mBuffers[0].mNumberChannels = stream->fmt->nChannels;
if(stream->cap_offs_frames + *nframes > stream->cap_bufsize_frames){
UINT32 chunk_frames = stream->cap_bufsize_frames - stream->cap_offs_frames;
if(stream->wrap_bufsize_frames < *nframes){
free(stream->wrap_buffer);
stream->wrap_buffer = malloc(data->mBuffers[0].mDataByteSize);
stream->wrap_bufsize_frames = *nframes;
}
memcpy(stream->wrap_buffer, stream->cap_buffer + stream->cap_offs_frames * stream->fmt->nBlockAlign,
chunk_frames * stream->fmt->nBlockAlign);
memcpy(stream->wrap_buffer + chunk_frames * stream->fmt->nBlockAlign, stream->cap_buffer,
(*nframes - chunk_frames) * stream->fmt->nBlockAlign);
data->mBuffers[0].mData = stream->wrap_buffer;
}else
data->mBuffers[0].mData = stream->cap_buffer + stream->cap_offs_frames * stream->fmt->nBlockAlign;
stream->cap_offs_frames += *nframes;
stream->cap_offs_frames %= stream->cap_bufsize_frames;
stream->cap_held_frames -= *nframes;
if(packets)
*packets = NULL;
return noErr;
}
static NTSTATUS capture_resample(void *args)
{
struct coreaudio_stream *stream = args;
UINT32 resamp_period_frames = muldiv(stream->period_frames, stream->dev_desc.mSampleRate,
stream->fmt->nSamplesPerSec);
OSStatus sc;
/* the resampling process often needs more source frames than we'd
* guess from a straight conversion using the sample rate ratio. so
* only convert if we have extra source data. */
while(stream->cap_held_frames > resamp_period_frames * 2){
AudioBufferList converted_list;
UInt32 wanted_frames = stream->period_frames;
converted_list.mNumberBuffers = 1;
converted_list.mBuffers[0].mNumberChannels = stream->fmt->nChannels;
converted_list.mBuffers[0].mDataByteSize = wanted_frames * stream->fmt->nBlockAlign;
if(stream->resamp_bufsize_frames < wanted_frames){
free(stream->resamp_buffer);
stream->resamp_buffer = malloc(converted_list.mBuffers[0].mDataByteSize);
stream->resamp_bufsize_frames = wanted_frames;
}
converted_list.mBuffers[0].mData = stream->resamp_buffer;
sc = AudioConverterFillComplexBuffer(stream->converter, feed_cb,
stream, &wanted_frames, &converted_list, NULL);
if(sc != noErr){
WARN("AudioConverterFillComplexBuffer failed: %x\n", (int)sc);
break;
}
ca_wrap_buffer(stream->local_buffer,
stream->wri_offs_frames * stream->fmt->nBlockAlign,
stream->bufsize_frames * stream->fmt->nBlockAlign,
stream->resamp_buffer, wanted_frames * stream->fmt->nBlockAlign);
stream->wri_offs_frames += wanted_frames;
stream->wri_offs_frames %= stream->bufsize_frames;
if(stream->held_frames + wanted_frames > stream->bufsize_frames){
stream->lcl_offs_frames += buf_ptr_diff(stream->lcl_offs_frames, stream->wri_offs_frames,
stream->bufsize_frames);
stream->held_frames = stream->bufsize_frames;
}else
stream->held_frames += wanted_frames;
}
return STATUS_SUCCESS;
}
unixlib_entry_t __wine_unix_call_funcs[] =
{
get_endpoint_ids,
@ -1019,4 +1122,6 @@ unixlib_entry_t __wine_unix_call_funcs[] =
release_stream,
get_mix_format,
is_format_supported,
capture_resample /* temporary */
};

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@ -138,10 +138,6 @@ struct ACImpl {
struct coreaudio_stream *stream;
struct list entry;
/* Temporary */
BYTE *feed_wrap_buffer;
UINT32 feed_wrap_bufsize_frames;
};
static const IAudioClient3Vtbl AudioClient3_Vtbl;
@ -581,7 +577,6 @@ static ULONG WINAPI AudioClient_Release(IAudioClient3 *iface)
if(This->stream->tmp_buffer)
NtFreeVirtualMemory(GetCurrentProcess(), (void **)&This->stream->tmp_buffer,
&This->stream->tmp_buffer_size, MEM_RELEASE);
HeapFree(GetProcessHeap(), 0, This->stream->resamp_buffer);
params.stream = This->stream;
UNIX_CALL(release_stream, &params);
}
@ -591,7 +586,6 @@ static ULONG WINAPI AudioClient_Release(IAudioClient3 *iface)
LeaveCriticalSection(&g_sessions_lock);
}
HeapFree(GetProcessHeap(), 0, This->vols);
free(This->feed_wrap_buffer);
IMMDevice_Release(This->parent);
IUnknown_Release(This->pUnkFTMarshal);
HeapFree(GetProcessHeap(), 0, This);
@ -736,103 +730,9 @@ static void silence_buffer(struct coreaudio_stream *stream, BYTE *buffer, UINT32
memset(buffer, 0, frames * stream->fmt->nBlockAlign);
}
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->stream->cap_held_frames);
if(!*nframes){
data->mBuffers[0].mData = NULL;
data->mBuffers[0].mDataByteSize = 0;
data->mBuffers[0].mNumberChannels = This->stream->fmt->nChannels;
return noErr;
}
data->mBuffers[0].mDataByteSize = *nframes * This->stream->fmt->nBlockAlign;
data->mBuffers[0].mNumberChannels = This->stream->fmt->nChannels;
if(This->stream->cap_offs_frames + *nframes > This->stream->cap_bufsize_frames){
UINT32 chunk_frames = This->stream->cap_bufsize_frames - This->stream->cap_offs_frames;
if(This->feed_wrap_bufsize_frames < *nframes){
free(This->feed_wrap_buffer);
This->feed_wrap_buffer = malloc(data->mBuffers[0].mDataByteSize);
This->feed_wrap_bufsize_frames = *nframes;
}
memcpy(This->feed_wrap_buffer, This->stream->cap_buffer + This->stream->cap_offs_frames * This->stream->fmt->nBlockAlign,
chunk_frames * This->stream->fmt->nBlockAlign);
memcpy(This->feed_wrap_buffer + chunk_frames * This->stream->fmt->nBlockAlign, This->stream->cap_buffer,
(*nframes - chunk_frames) * This->stream->fmt->nBlockAlign);
data->mBuffers[0].mData = This->feed_wrap_buffer;
}else
data->mBuffers[0].mData = This->stream->cap_buffer + This->stream->cap_offs_frames * This->stream->fmt->nBlockAlign;
This->stream->cap_offs_frames += *nframes;
This->stream->cap_offs_frames %= This->stream->cap_bufsize_frames;
This->stream->cap_held_frames -= *nframes;
if(packets)
*packets = NULL;
return noErr;
}
static void capture_resample(ACImpl *This)
{
UINT32 resamp_period_frames = MulDiv(This->stream->period_frames, This->stream->dev_desc.mSampleRate, This->stream->fmt->nSamplesPerSec);
OSStatus sc;
/* the resampling process often needs more source frames than we'd
* guess from a straight conversion using the sample rate ratio. so
* only convert if we have extra source data. */
while(This->stream->cap_held_frames > resamp_period_frames * 2){
AudioBufferList converted_list;
UInt32 wanted_frames = This->stream->period_frames;
converted_list.mNumberBuffers = 1;
converted_list.mBuffers[0].mNumberChannels = This->stream->fmt->nChannels;
converted_list.mBuffers[0].mDataByteSize = wanted_frames * This->stream->fmt->nBlockAlign;
if(This->stream->resamp_bufsize_frames < wanted_frames){
HeapFree(GetProcessHeap(), 0, This->stream->resamp_buffer);
This->stream->resamp_buffer = HeapAlloc(GetProcessHeap(), 0, converted_list.mBuffers[0].mDataByteSize);
This->stream->resamp_bufsize_frames = wanted_frames;
}
converted_list.mBuffers[0].mData = This->stream->resamp_buffer;
sc = AudioConverterFillComplexBuffer(This->stream->converter, feed_cb,
This, &wanted_frames, &converted_list, NULL);
if(sc != noErr){
WARN("AudioConverterFillComplexBuffer failed: %x\n", (int)sc);
break;
}
ca_wrap_buffer(This->stream->local_buffer,
This->stream->wri_offs_frames * This->stream->fmt->nBlockAlign,
This->stream->bufsize_frames * This->stream->fmt->nBlockAlign,
This->stream->resamp_buffer, wanted_frames * This->stream->fmt->nBlockAlign);
This->stream->wri_offs_frames += wanted_frames;
This->stream->wri_offs_frames %= This->stream->bufsize_frames;
if(This->stream->held_frames + wanted_frames > This->stream->bufsize_frames){
This->stream->lcl_offs_frames += buf_ptr_diff(This->stream->lcl_offs_frames,
This->stream->wri_offs_frames, This->stream->bufsize_frames);
This->stream->held_frames = This->stream->bufsize_frames;
}else
This->stream->held_frames += wanted_frames;
}
UNIX_CALL(capture_resample, This->stream);
}
static HRESULT WINAPI AudioClient_Initialize(IAudioClient3 *iface,

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@ -100,6 +100,8 @@ enum unix_funcs
unix_release_stream,
unix_get_mix_format,
unix_is_format_supported,
unix_capture_resample /* temporary */
};
extern unixlib_handle_t coreaudio_handle;