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https://github.com/mpv-player/mpv
synced 2025-04-11 04:01:31 +00:00
ao_dsound: uncrustify
This commit is contained in:
wm4
parent
92ae48db0f
commit
cee56e8623
@ -62,7 +62,9 @@ LIBAO_EXTERN(dsound)
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#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
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#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
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static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
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static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
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0x1, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}
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};
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#if 0
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#define DSSPEAKER_HEADPHONE 0x00000001
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@ -131,8 +133,8 @@ static char * dserr2str(int err)
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case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED";
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case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE";
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case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED";
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default: return "unknown";
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}
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return "unknown";
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}
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/**
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@ -172,16 +174,16 @@ static void print_help(void)
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\brief enumerate direct sound devices
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\return TRUE to continue with the enumeration
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*/
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static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context)
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static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module,
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LPVOID context)
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{
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int *device_index = context;
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mp_msg(MSGT_AO, MSGL_V, "%i %s ", *device_index, desc);
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if (device_num == *device_index) {
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mp_msg(MSGT_AO, MSGL_V, "<--");
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if(guid){
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if (guid)
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memcpy(&device, guid, sizeof(GUID));
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}
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}
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mp_msg(MSGT_AO, MSGL_V, "\n");
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(*device_index)++;
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return TRUE;
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@ -214,8 +216,10 @@ static int InitDirectSound(void)
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mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
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return 0;
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}
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OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate");
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OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA");
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OurDirectSoundCreate = (void *)GetProcAddress(hdsound_dll,
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"DirectSoundCreate");
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OurDirectSoundEnumerate = (void *)GetProcAddress(hdsound_dll,
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"DirectSoundEnumerateA");
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if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
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mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n");
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@ -228,8 +232,11 @@ static int InitDirectSound(void)
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OurDirectSoundEnumerate(DirectSoundEnum, &device_index);
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// Create the direct sound object
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if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) {
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mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n");
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if (FAILED(OurDirectSoundCreate((device_num) ? &device : NULL, &hds,
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NULL)))
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{
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot create a DirectSound device\n");
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FreeLibrary(hdsound_dll);
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return 0;
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}
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@ -244,8 +251,11 @@ static int InitDirectSound(void)
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* sound without any video, and so what window handle should we use ???
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* The hack for now is to use the Desktop window handle - it seems to be
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* working */
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if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) {
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mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n");
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if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(),
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DSSCL_EXCLUSIVE))
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{
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot set direct sound cooperative level\n");
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IDirectSound_Release(hds);
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FreeLibrary(hdsound_dll);
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return 0;
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@ -255,7 +265,9 @@ static int InitDirectSound(void)
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memset(&dscaps, 0, sizeof(DSCAPS));
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dscaps.dwSize = sizeof(DSCAPS);
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if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) {
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if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n");
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if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
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mp_msg(MSGT_AO, MSGL_V,
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"ao_dsound: DirectSound is emulated, waveOut may give better performance\n");
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} else {
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n");
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}
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@ -295,17 +307,17 @@ static int write_buffer(unsigned char *data, int len)
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underrun_check = 0;
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// Lock the buffer
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res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
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res = IDirectSoundBuffer_Lock(hdsbuf, write_offset, len, &lpvPtr1, &dwBytes1,
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&lpvPtr2, &dwBytes2, 0);
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// If the buffer was lost, restore and retry lock.
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if (DSERR_BUFFERLOST == res)
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{
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if (DSERR_BUFFERLOST == res) {
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IDirectSoundBuffer_Restore(hdsbuf);
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res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
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res = IDirectSoundBuffer_Lock(hdsbuf, write_offset, len, &lpvPtr1,
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&dwBytes1, &lpvPtr2, &dwBytes2, 0);
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}
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if (SUCCEEDED(res))
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{
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if (SUCCEEDED(res)) {
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if (!AF_FORMAT_IS_AC3(ao_data.format)) {
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memcpy(lpvPtr1, data, dwBytes1);
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if (lpvPtr2 != NULL)
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@ -317,21 +329,22 @@ static int write_buffer(unsigned char *data, int len)
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} else {
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// Write to pointers without reordering.
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memcpy(lpvPtr1, data, dwBytes1);
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if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
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if (NULL != lpvPtr2)
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memcpy(lpvPtr2, data + dwBytes1, dwBytes2);
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write_offset += dwBytes1 + dwBytes2;
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if(write_offset>=buffer_size)write_offset=dwBytes2;
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if (write_offset >= buffer_size)
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write_offset = dwBytes2;
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}
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// Release the data back to DirectSound.
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res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
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if (SUCCEEDED(res))
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{
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res = IDirectSoundBuffer_Unlock(hdsbuf, lpvPtr1, dwBytes1, lpvPtr2,
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dwBytes2);
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if (SUCCEEDED(res)) {
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// Success.
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DWORD status;
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IDirectSoundBuffer_GetStatus(hdsbuf, &status);
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if (!(status & DSBSTATUS_PLAYING)){
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if (!(status & DSBSTATUS_PLAYING))
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res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
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}
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return dwBytes1 + dwBytes2;
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}
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}
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@ -380,7 +393,8 @@ static int control(int cmd, void *arg)
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static int init(int rate, const struct mp_chmap *channels, int format, int flags)
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{
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int res;
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if (!InitDirectSound()) return 0;
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if (!InitDirectSound())
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return 0;
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global_ao->no_persistent_volume = true;
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audio_volume = 100;
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@ -390,10 +404,12 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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DSBUFFERDESC dsbpridesc;
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DSBUFFERDESC dsbdesc;
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if (AF_FORMAT_IS_AC3(format)) {
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if (AF_FORMAT_IS_AC3(format))
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format = AF_FORMAT_AC3_NE;
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} else {
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struct mp_chmap_sel sel = {0};
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else {
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struct mp_chmap_sel sel = {
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0
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};
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mp_chmap_sel_add_waveext(&sel);
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if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
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return 0;
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@ -405,20 +421,27 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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case AF_FORMAT_U8:
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break;
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default:
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mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V,
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"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",
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af_fmt2str_short(format));
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format = AF_FORMAT_S16_LE;
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}
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//fill global ao_data
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ao_data.samplerate = rate;
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ao_data.format = format;
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ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format) >> 3);
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if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
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mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, ao_data.channels.num, af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
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if (ao_data.buffersize == -1)
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ao_data.buffersize = ao_data.bps; // space for 1 sec
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mp_msg(MSGT_AO, MSGL_V,
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"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate,
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ao_data.channels.num, af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Buffersize:%d bytes (%d msec)\n",
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ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
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//fill waveformatex
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ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
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wformat.Format.cbSize = (ao_data.channels.num > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
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wformat.Format.cbSize = (ao_data.channels.num > 2)
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? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
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wformat.Format.nChannels = ao_data.channels.num;
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wformat.Format.nSamplesPerSec = rate;
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if (AF_FORMAT_IS_AC3(format)) {
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@ -426,9 +449,11 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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wformat.Format.wBitsPerSample = 16;
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wformat.Format.nBlockAlign = 4;
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} else {
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wformat.Format.wFormatTag = (ao_data.channels.num > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
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wformat.Format.wFormatTag = (ao_data.channels.num > 2)
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? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
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wformat.Format.wBitsPerSample = af_fmt2bits(format);
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wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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wformat.Format.nBlockAlign = wformat.Format.nChannels *
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(wformat.Format.wBitsPerSample >> 3);
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}
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// fill in primary sound buffer descriptor
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@ -438,7 +463,6 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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dsbpridesc.dwBufferBytes = 0;
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dsbpridesc.lpwfxFormat = NULL;
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// fill in the secondary sound buffer (=stream buffer) descriptor
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memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
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dsbdesc.dwSize = sizeof(DSBUFFERDESC);
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@ -453,7 +477,8 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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// Needed for 5.1 on emu101k - shit soundblaster
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dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE;
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}
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wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec *
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wformat.Format.nBlockAlign;
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dsbdesc.dwBufferBytes = ao_data.buffersize;
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dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
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@ -467,11 +492,17 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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res = IDirectSound_CreateSoundBuffer(hds, &dsbpridesc, &hdspribuf, NULL);
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if (res != DS_OK) {
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UninitDirectSound();
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mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res));
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot create primary buffer (%s)\n",
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dserr2str(res));
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return 0;
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}
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res = IDirectSoundBuffer_SetFormat(hdspribuf, (WAVEFORMATEX *)&wformat);
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if ( res != DS_OK ) mp_msg(MSGT_AO, MSGL_WARN,"ao_dsound: cannot set primary buffer format (%s), using standard setting (bad quality)", dserr2str(res));
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if (res != DS_OK) {
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mp_msg(MSGT_AO, MSGL_WARN,
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"ao_dsound: cannot set primary buffer format (%s), using "
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"standard setting (bad quality)", dserr2str(res));
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}
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n");
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@ -486,7 +517,9 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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}
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if (res != DS_OK) {
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UninitDirectSound();
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mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res));
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot create secondary (stream)buffer (%s)\n",
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dserr2str(res));
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return 0;
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}
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}
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@ -550,7 +583,8 @@ static int check_free_buffer_size(void)
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// play_cursor is the actual postion of the play cursor
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// write_cursor is the position after which it is assumed to be save to write data
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// write_offset is the postion where we actually write the data to
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if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
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if (space > buffer_size)
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space -= buffer_size; // write_offset < play_offset
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// Check for buffer underruns. An underrun happens if DirectSound
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// started to play old data beyond the current write_offset. Detect this
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// by checking whether the free space shrinks, even though no data was
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@ -573,7 +607,8 @@ static int check_free_buffer_size(void)
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static int get_space(void)
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{
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int space = check_free_buffer_size();
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if(space < min_free_space)return 0;
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if (space < min_free_space)
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return 0;
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return space - min_free_space;
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}
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@ -587,7 +622,8 @@ static int get_space(void)
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static int play(void *data, int len, int flags)
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{
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int space = check_free_buffer_size();
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if(space < len) len = space;
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if (space < len)
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len = space;
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if (!(flags & AOPLAY_FINAL_CHUNK))
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len = (len / ao_data.outburst) * ao_data.outburst;
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