Use the converters to enable mmdevapi capture
This commit is contained in:
Chris Robinson
parent
bcdd1cee10
commit
78d5492d2c
@ -44,6 +44,7 @@
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#include "threads.h"
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#include "compat.h"
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#include "alstring.h"
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#include "converter.h"
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#include "backends/base.h"
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@ -1206,6 +1207,8 @@ typedef struct ALCmmdevCapture {
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HANDLE MsgEvent;
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ChannelConverter *ChannelConv;
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SampleConverter *SampleConv;
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ll_ringbuffer_t *Ring;
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volatile int killNow;
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@ -1253,6 +1256,8 @@ static void ALCmmdevCapture_Construct(ALCmmdevCapture *self, ALCdevice *device)
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self->MsgEvent = NULL;
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self->ChannelConv = NULL;
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self->SampleConv = NULL;
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self->Ring = NULL;
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self->killNow = 0;
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@ -1263,6 +1268,9 @@ static void ALCmmdevCapture_Destruct(ALCmmdevCapture *self)
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ll_ringbuffer_free(self->Ring);
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self->Ring = NULL;
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DestroySampleConverter(&self->SampleConv);
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DestroyChannelConverter(&self->ChannelConv);
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if(self->NotifyEvent != NULL)
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CloseHandle(self->NotifyEvent);
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self->NotifyEvent = NULL;
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@ -1282,6 +1290,8 @@ FORCE_ALIGN int ALCmmdevCapture_recordProc(void *arg)
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{
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ALCmmdevCapture *self = arg;
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ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
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ALfloat *samples = NULL;
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size_t samplesmax = 0;
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HRESULT hr;
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hr = CoInitialize(NULL);
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@ -1304,33 +1314,74 @@ FORCE_ALIGN int ALCmmdevCapture_recordProc(void *arg)
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hr = IAudioCaptureClient_GetNextPacketSize(self->capture, &avail);
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if(FAILED(hr))
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ERR("Failed to get next packet size: 0x%08lx\n", hr);
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else while(avail > 0 && SUCCEEDED(hr))
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else if(avail > 0)
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{
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UINT32 numsamples;
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DWORD flags;
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BYTE *data;
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BYTE *rdata;
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hr = IAudioCaptureClient_GetBuffer(self->capture,
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&data, &numsamples, &flags, NULL, NULL
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&rdata, &numsamples, &flags, NULL, NULL
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);
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if(FAILED(hr))
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{
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ERR("Failed to get capture buffer: 0x%08lx\n", hr);
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break;
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}
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ll_ringbuffer_write(self->Ring, (char*)data, numsamples);
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hr = IAudioCaptureClient_ReleaseBuffer(self->capture, numsamples);
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if(FAILED(hr))
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else
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{
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ERR("Failed to release capture buffer: 0x%08lx\n", hr);
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break;
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}
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ll_ringbuffer_data_t data[2];
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size_t dstframes = 0;
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hr = IAudioCaptureClient_GetNextPacketSize(self->capture, &avail);
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if(FAILED(hr))
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ERR("Failed to get next packet size: 0x%08lx\n", hr);
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if(self->ChannelConv)
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{
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if(samplesmax < numsamples)
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{
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size_t newmax = RoundUp(numsamples, 4096);
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ALfloat *tmp = al_calloc(DEF_ALIGN, newmax*2*sizeof(ALfloat));
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al_free(samples);
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samples = tmp;
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samplesmax = newmax;
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}
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ChannelConverterInput(self->ChannelConv, rdata, samples, numsamples);
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rdata = (BYTE*)samples;
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}
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ll_ringbuffer_get_write_vector(self->Ring, data);
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if(self->SampleConv)
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{
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const ALvoid *srcdata = rdata;
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ALsizei srcframes = numsamples;
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dstframes = SampleConverterInput(self->SampleConv,
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&srcdata, &srcframes, data[0].buf, data[0].len
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);
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if(srcframes > 0 && dstframes == data[0].len && data[1].len > 0)
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{
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/* If some source samples remain, all of the first dest
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* block was filled, and there's space in the second
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* dest block, do another run for the second block.
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*/
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dstframes += SampleConverterInput(self->SampleConv,
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&srcdata, &srcframes, data[1].buf, data[1].len
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);
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}
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}
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else
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{
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size_t framesize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
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ALuint len1 = minu(data[0].len, numsamples);
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ALuint len2 = minu(data[1].len, numsamples-len1);
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memcpy(data[0].buf, rdata, len1*framesize);
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if(len2 > 0)
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memcpy(data[1].buf, rdata+len1*framesize, len2*framesize);
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dstframes = len1 + len2;
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}
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ll_ringbuffer_write_advance(self->Ring, dstframes);
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hr = IAudioCaptureClient_ReleaseBuffer(self->capture, numsamples);
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if(FAILED(hr)) ERR("Failed to release capture buffer: 0x%08lx\n", hr);
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}
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}
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if(FAILED(hr))
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@ -1346,6 +1397,10 @@ FORCE_ALIGN int ALCmmdevCapture_recordProc(void *arg)
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ERR("WaitForSingleObjectEx error: 0x%lx\n", res);
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}
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al_free(samples);
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samples = NULL;
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samplesmax = 0;
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CoUninitialize();
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return 0;
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}
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@ -1528,6 +1583,7 @@ static HRESULT ALCmmdevCapture_resetProxy(ALCmmdevCapture *self)
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ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
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WAVEFORMATEXTENSIBLE OutputType;
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WAVEFORMATEX *wfx = NULL;
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enum DevFmtType srcType;
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REFERENCE_TIME buf_time;
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UINT32 buffer_len;
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void *ptr = NULL;
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@ -1587,33 +1643,28 @@ static HRESULT ALCmmdevCapture_resetProxy(ALCmmdevCapture *self)
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}
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switch(device->FmtType)
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{
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/* NOTE: Signedness doesn't matter, the converter will handle it. */
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case DevFmtByte:
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case DevFmtUByte:
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OutputType.Format.wBitsPerSample = 8;
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OutputType.Samples.wValidBitsPerSample = 8;
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OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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break;
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case DevFmtShort:
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case DevFmtUShort:
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OutputType.Format.wBitsPerSample = 16;
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OutputType.Samples.wValidBitsPerSample = 16;
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OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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break;
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case DevFmtInt:
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case DevFmtUInt:
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OutputType.Format.wBitsPerSample = 32;
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OutputType.Samples.wValidBitsPerSample = 32;
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OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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break;
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case DevFmtFloat:
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OutputType.Format.wBitsPerSample = 32;
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OutputType.Samples.wValidBitsPerSample = 32;
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OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
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break;
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case DevFmtByte:
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case DevFmtUShort:
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case DevFmtUInt:
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WARN("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
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return E_FAIL;
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}
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OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
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OutputType.Format.nSamplesPerSec = device->Frequency;
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OutputType.Format.nBlockAlign = OutputType.Format.nChannels *
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@ -1631,27 +1682,103 @@ static HRESULT ALCmmdevCapture_resetProxy(ALCmmdevCapture *self)
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return hr;
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}
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/* FIXME: We should do conversion/resampling if we didn't get a matching format. */
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if(wfx->nSamplesPerSec != OutputType.Format.nSamplesPerSec ||
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wfx->wBitsPerSample != OutputType.Format.wBitsPerSample ||
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wfx->nChannels != OutputType.Format.nChannels ||
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wfx->nBlockAlign != OutputType.Format.nBlockAlign)
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DestroySampleConverter(&self->SampleConv);
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DestroyChannelConverter(&self->ChannelConv);
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if(wfx != NULL)
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{
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ERR("Failed to get matching format, wanted: %s %s %uhz, got: %d channel%s %d-bit %luhz\n",
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DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType),
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device->Frequency, wfx->nChannels, (wfx->nChannels==1)?"":"s", wfx->wBitsPerSample,
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wfx->nSamplesPerSec);
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if(!(wfx->nChannels == OutputType.Format.nChannels ||
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(wfx->nChannels == 1 && OutputType.Format.nChannels == 2) ||
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(wfx->nChannels == 2 && OutputType.Format.nChannels == 1)))
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{
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ERR("Failed to get matching format, wanted: %s %s %uhz, got: %d channel%s %d-bit %luhz\n",
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DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType),
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device->Frequency, wfx->nChannels, (wfx->nChannels==1)?"":"s", wfx->wBitsPerSample,
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wfx->nSamplesPerSec);
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CoTaskMemFree(wfx);
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return E_FAIL;
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}
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if(!MakeExtensible(&OutputType, wfx))
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{
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CoTaskMemFree(wfx);
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return E_FAIL;
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}
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CoTaskMemFree(wfx);
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wfx = NULL;
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}
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if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))
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{
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if(OutputType.Format.wBitsPerSample == 8)
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srcType = DevFmtUByte;
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else if(OutputType.Format.wBitsPerSample == 16)
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srcType = DevFmtShort;
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else if(OutputType.Format.wBitsPerSample == 32)
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srcType = DevFmtInt;
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else
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{
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ERR("Unhandled integer bit depth: %d\n", OutputType.Format.wBitsPerSample);
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return E_FAIL;
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}
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}
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else if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))
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{
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if(OutputType.Format.wBitsPerSample == 32)
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srcType = DevFmtFloat;
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else
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{
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ERR("Unhandled float bit depth: %d\n", OutputType.Format.wBitsPerSample);
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return E_FAIL;
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}
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}
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else
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{
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ERR("Unhandled format sub-type\n");
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return E_FAIL;
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}
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if(!MakeExtensible(&OutputType, wfx))
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if(device->FmtChans == DevFmtMono && OutputType.Format.nChannels == 2)
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{
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CoTaskMemFree(wfx);
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return E_FAIL;
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self->ChannelConv = CreateChannelConverter(srcType, DevFmtStereo,
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device->FmtChans);
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if(!self->ChannelConv)
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{
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ERR("Failed to create stereo-to-mono converter\n");
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return E_FAIL;
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}
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/* The channel converter always outputs float, so change the input type
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* for the resampler/type-converter.
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*/
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srcType = DevFmtFloat;
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}
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else if(device->FmtChans == DevFmtStereo && OutputType.Format.nChannels == 1)
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{
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self->ChannelConv = CreateChannelConverter(srcType, DevFmtMono,
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device->FmtChans);
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if(!self->ChannelConv)
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{
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ERR("Failed to create mono-to-stereo converter\n");
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return E_FAIL;
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}
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srcType = DevFmtFloat;
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}
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if(device->Frequency != OutputType.Format.nSamplesPerSec || device->FmtType != srcType)
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{
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self->SampleConv = CreateSampleConverter(
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srcType, device->FmtType, ChannelsFromDevFmt(device->FmtChans),
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OutputType.Format.nSamplesPerSec, device->Frequency
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);
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if(!self->SampleConv)
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{
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ERR("Failed to create converter for format, dst: %s %s %uhz, src: %d-bit %luhz\n",
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DevFmtChannelsString(device->FmtChans), DevFmtTypeString(device->FmtType),
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device->Frequency, OutputType.Format.wBitsPerSample,
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OutputType.Format.nSamplesPerSec);
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return E_FAIL;
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}
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}
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CoTaskMemFree(wfx);
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wfx = NULL;
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hr = IAudioClient_Initialize(self->client,
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AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
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@ -1672,7 +1799,9 @@ static HRESULT ALCmmdevCapture_resetProxy(ALCmmdevCapture *self)
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buffer_len = maxu(device->UpdateSize*device->NumUpdates + 1, buffer_len);
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ll_ringbuffer_free(self->Ring);
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self->Ring = ll_ringbuffer_create(buffer_len, OutputType.Format.nBlockAlign);
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self->Ring = ll_ringbuffer_create(buffer_len,
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FrameSizeFromDevFmt(device->FmtChans, device->FmtType)
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);
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if(!self->Ring)
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{
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ERR("Failed to allocate capture ring buffer\n");
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@ -1851,7 +1980,7 @@ static ALCboolean ALCmmdevBackendFactory_querySupport(ALCmmdevBackendFactory* UN
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* stereo input, for example, and the app asks for 22050hz mono,
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* initialization will fail.
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*/
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if(type == ALCbackend_Playback /*|| type == ALCbackend_Capture*/)
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if(type == ALCbackend_Playback || type == ALCbackend_Capture)
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return ALC_TRUE;
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return ALC_FALSE;
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}
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