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wine-winehq
Commit 0e52c42a authored by Ken Thomases's avatar Ken Thomases Committed by Alexandre Julliard
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winecoreaudio: Report actual supported input wave formats.

Also, added some output formats that Core Audio ought to be able to handle.
parent fa844516
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Showing with 73 additions and 1 deletion
dlls/winmm/winecoreaudio/audio.c
View file @ 0e52c42a
......@@ -224,6 +224,8 @@ extern OSStatus AudioUnitUninitialize(AudioUnit au);
extern int AudioUnit_SetVolume(AudioUnit au, float left, float right);
extern int AudioUnit_GetVolume(AudioUnit au, float *left, float *right);
extern int AudioUnit_GetInputDeviceSampleRate(void);
extern int AudioUnit_CreateInputUnit(void* wwi, AudioUnit* out_au,
WORD nChannels, DWORD nSamplesPerSec, WORD wBitsPerSample,
UInt32* outFrameCount);
......@@ -500,6 +502,7 @@ LONG CoreAudio_WaveInit(void)
HANDLE hThread;
CFStringRef messageThreadPortName;
CFMessagePortRef port_ReceiveInMessageThread;
int inputSampleRate;
TRACE("()\n");
......@@ -550,6 +553,14 @@ LONG CoreAudio_WaveInit(void)
WOutDev[i].caps.wChannels = 2;
/* WOutDev[i].caps.dwSupport |= WAVECAPS_LRVOLUME; */ /* FIXME */
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_96M08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_96S08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_96M16;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_96S16;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_48M08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_48S08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_48M16;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_48S16;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_4M08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_4S08;
WOutDev[i].caps.dwFormats |= WAVE_FORMAT_4S16;
......@@ -566,6 +577,9 @@ LONG CoreAudio_WaveInit(void)
WOutDev[i].lock = 0; /* initialize the mutex */
}
/* FIXME: implement sample rate conversion on input */
inputSampleRate = AudioUnit_GetInputDeviceSampleRate();
for (i = 0; i < MAX_WAVEINDRV; ++i)
{
memset(&WInDev[i], 0, sizeof(WInDev[i]));
......@@ -584,18 +598,41 @@ LONG CoreAudio_WaveInit(void)
MultiByteToWideChar(CP_ACP, 0, szPname, -1, WInDev[i].caps.szPname, sizeof(WInDev[i].caps.szPname)/sizeof(WCHAR));
snprintf(WInDev[i].interface_name, sizeof(WInDev[i].interface_name), "winecoreaudio in: %d", i);
if (inputSampleRate == 96000)
{
WInDev[i].caps.dwFormats |= WAVE_FORMAT_96M08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_96S08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_96M16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_96S16;
}
if (inputSampleRate == 48000)
{
WInDev[i].caps.dwFormats |= WAVE_FORMAT_48M08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_48S08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_48M16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_48S16;
}
if (inputSampleRate == 44100)
{
WInDev[i].caps.dwFormats |= WAVE_FORMAT_4M08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_4S08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_4S16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_4M16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_4S16;
}
if (inputSampleRate == 22050)
{
WInDev[i].caps.dwFormats |= WAVE_FORMAT_2M08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_2S08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_2M16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_2S16;
}
if (inputSampleRate == 11025)
{
WInDev[i].caps.dwFormats |= WAVE_FORMAT_1M08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_1S08;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_1M16;
WInDev[i].caps.dwFormats |= WAVE_FORMAT_1S16;
}
WInDev[i].caps.wChannels = 2;
}
......@@ -2140,6 +2177,9 @@ OSStatus CoreAudio_wiAudioUnitIOProc(void *inRefCon,
*ioActionFlags, inTimeStamp->mSampleTime, inTimeStamp->mHostTime, inTimeStamp->mRateScalar, inTimeStamp->mWordClockTime, inTimeStamp->mFlags, inBusNumber, inNumberFrames);
/* Render into audio buffer */
/* FIXME: implement sample rate conversion on input. This will require
* a different render strategy. We'll need to buffer the sound data
* received here and pass it off to an AUConverter in another thread. */
err = AudioUnitRender(wwi->audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, wwi->bufferList);
if (err)
{
......
dlls/winmm/winecoreaudio/audiounit.c
View file @ 0e52c42a
......@@ -130,6 +130,34 @@ int AudioUnit_GetVolume(AudioUnit au, float *left, float *right)
}
/* FIXME: implement sample rate conversion on input */
int AudioUnit_GetInputDeviceSampleRate(void)
{
AudioDeviceID defaultInputDevice;
UInt32 param;
Float64 sampleRate;
OSStatus err;
param = sizeof(defaultInputDevice);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &param, &defaultInputDevice);
if (err != noErr || defaultInputDevice == kAudioDeviceUnknown)
{
ERR("Couldn't get the default audio input device ID: %08lx\n", err);
return 0;
}
param = sizeof(sampleRate);
err = AudioDeviceGetProperty(defaultInputDevice, 0, 1, kAudioDevicePropertyNominalSampleRate, &param, &sampleRate);
if (err != noErr)
{
ERR("Couldn't get the device sample rate: %08lx\n", err);
return 0;
}
return sampleRate;
}
int AudioUnit_CreateInputUnit(void* wwi, AudioUnit* out_au,
WORD nChannels, DWORD nSamplesPerSec, WORD wBitsPerSample,
UInt32* outFrameCount)
......@@ -232,6 +260,10 @@ int AudioUnit_CreateInputUnit(void* wwi, AudioUnit* out_au,
}
/* Setup the desired data format. */
/* FIXME: implement sample rate conversion on input. We shouldn't set
* the mSampleRate of this to the desired sample rate. We need to query
* the input device and use that. If they don't match, we need to set up
* an AUConverter to do the sample rate conversion on a separate thread. */
desiredFormat.mFormatID = kAudioFormatLinearPCM;
desiredFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked;
if (wBitsPerSample != 8)
......
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