From cee56e8623b18ddf37426ab9204b551925006083 Mon Sep 17 00:00:00 2001 From: wm4 Date: Tue, 4 Jun 2013 00:51:07 +0200 Subject: ao_dsound: uncrustify --- audio/out/ao_dsound.c | 714 ++++++++++++++++++++++++++------------------------ 1 file changed, 375 insertions(+), 339 deletions(-) (limited to 'audio') diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c index f57d7cd96e..95a45a1482 100644 --- a/audio/out/ao_dsound.c +++ b/audio/out/ao_dsound.c @@ -47,10 +47,10 @@ static const ao_info_t info = { - "Windows DirectSound audio output", - "dsound", - "Gabor Szecsi ", - "" + "Windows DirectSound audio output", + "dsound", + "Gabor Szecsi ", + "" }; LIBAO_EXTERN(dsound) @@ -62,7 +62,9 @@ LIBAO_EXTERN(dsound) #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE -static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}}; +static const GUID KSDATAFORMAT_SUBTYPE_PCM = { + 0x1, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71} +}; #if 0 #define DSSPEAKER_HEADPHONE 0x00000001 @@ -75,15 +77,15 @@ static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00 #ifndef _WAVEFORMATEXTENSIBLE_ typedef struct { - WAVEFORMATEX Format; + WAVEFORMATEX Format; union { WORD wValidBitsPerSample; /* bits of precision */ WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */ WORD wReserved; /* If neither applies, set to zero. */ } Samples; - DWORD dwChannelMask; /* which channels are */ + DWORD dwChannelMask; /* which channels are */ /* present in stream */ - GUID SubFormat; + GUID SubFormat; } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; #endif @@ -111,28 +113,28 @@ static int audio_volume; */ static char * dserr2str(int err) { - switch (err) { - case DS_OK: return "DS_OK"; - case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION"; - case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION"; - case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL"; - case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM"; - case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL"; - case DSERR_GENERIC: return "DSERR_GENERIC"; - case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED"; - case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY"; - case DSERR_BADFORMAT: return "DSERR_BADFORMAT"; - case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED"; - case DSERR_NODRIVER: return "DSERR_NODRIVER"; - case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED"; - case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION"; - case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST"; - case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO"; - case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED"; - case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE"; - case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED"; - default: return "unknown"; - } + switch (err) { + case DS_OK: return "DS_OK"; + case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION"; + case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION"; + case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL"; + case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM"; + case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL"; + case DSERR_GENERIC: return "DSERR_GENERIC"; + case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED"; + case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY"; + case DSERR_BADFORMAT: return "DSERR_BADFORMAT"; + case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED"; + case DSERR_NODRIVER: return "DSERR_NODRIVER"; + case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED"; + case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION"; + case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST"; + case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO"; + case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED"; + case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE"; + case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED"; + } + return "unknown"; } /** @@ -142,15 +144,15 @@ static void UninitDirectSound(void) { // finally release the DirectSound object if (hds) { - IDirectSound_Release(hds); - hds = NULL; + IDirectSound_Release(hds); + hds = NULL; } // free DSOUND.DLL if (hdsound_dll) { - FreeLibrary(hdsound_dll); - hdsound_dll = NULL; + FreeLibrary(hdsound_dll); + hdsound_dll = NULL; } - mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); } /** @@ -158,7 +160,7 @@ static void UninitDirectSound(void) */ static void print_help(void) { - mp_msg(MSGT_AO, MSGL_FATAL, + mp_msg(MSGT_AO, MSGL_FATAL, "\n-ao dsound commandline help:\n" "Example: mpv -ao dsound:device=1\n" " sets 1st device\n" @@ -172,17 +174,17 @@ static void print_help(void) \brief enumerate direct sound devices \return TRUE to continue with the enumeration */ -static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context) +static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module, + LPVOID context) { - int* device_index=context; - mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc); - if(device_num==*device_index){ - mp_msg(MSGT_AO, MSGL_V,"<--"); - if(guid){ - memcpy(&device,guid,sizeof(GUID)); - } + int *device_index = context; + mp_msg(MSGT_AO, MSGL_V, "%i %s ", *device_index, desc); + if (device_num == *device_index) { + mp_msg(MSGT_AO, MSGL_V, "<--"); + if (guid) + memcpy(&device, guid, sizeof(GUID)); } - mp_msg(MSGT_AO, MSGL_V,"\n"); + mp_msg(MSGT_AO, MSGL_V, "\n"); (*device_index)++; return TRUE; } @@ -194,73 +196,83 @@ static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOI */ static int InitDirectSound(void) { - DSCAPS dscaps; + DSCAPS dscaps; - // initialize directsound + // initialize directsound HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN); - HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID); - int device_index=0; - const opt_t subopts[] = { - {"device", OPT_ARG_INT, &device_num,NULL}, - {NULL} - }; - if (subopt_parse(ao_subdevice, subopts) != 0) { - print_help(); - return 0; - } - - hdsound_dll = LoadLibrary("DSOUND.DLL"); - if (hdsound_dll == NULL) { - mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n"); - return 0; - } - OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate"); - OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA"); - - if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) { - mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); - FreeLibrary(hdsound_dll); - return 0; - } - - // Enumerate all directsound devices - mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n"); - OurDirectSoundEnumerate(DirectSoundEnum,&device_index); - - // Create the direct sound object - if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) { - mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n"); - FreeLibrary(hdsound_dll); - return 0; - } - - /* Set DirectSound Cooperative level, ie what control we want over Windows - * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the - * settings of the primary buffer, but also that only the sound of our - * application will be hearable when it will have the focus. - * !!! (this is not really working as intended yet because to set the - * cooperative level you need the window handle of your application, and - * I don't know of any easy way to get it. Especially since we might play - * sound without any video, and so what window handle should we use ??? - * The hack for now is to use the Desktop window handle - it seems to be - * working */ - if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) { - mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n"); - IDirectSound_Release(hds); - FreeLibrary(hdsound_dll); - return 0; - } - mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n"); - - memset(&dscaps, 0, sizeof(DSCAPS)); - dscaps.dwSize = sizeof(DSCAPS); - if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) { - if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n"); - } else { - mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n"); - } - - return 1; + HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID); + int device_index = 0; + const opt_t subopts[] = { + {"device", OPT_ARG_INT, &device_num, NULL}, + {NULL} + }; + if (subopt_parse(ao_subdevice, subopts) != 0) { + print_help(); + return 0; + } + + hdsound_dll = LoadLibrary("DSOUND.DLL"); + if (hdsound_dll == NULL) { + mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n"); + return 0; + } + OurDirectSoundCreate = (void *)GetProcAddress(hdsound_dll, + "DirectSoundCreate"); + OurDirectSoundEnumerate = (void *)GetProcAddress(hdsound_dll, + "DirectSoundEnumerateA"); + + if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) { + mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); + FreeLibrary(hdsound_dll); + return 0; + } + + // Enumerate all directsound devices + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Output Devices:\n"); + OurDirectSoundEnumerate(DirectSoundEnum, &device_index); + + // Create the direct sound object + if (FAILED(OurDirectSoundCreate((device_num) ? &device : NULL, &hds, + NULL))) + { + mp_msg(MSGT_AO, MSGL_ERR, + "ao_dsound: cannot create a DirectSound device\n"); + FreeLibrary(hdsound_dll); + return 0; + } + + /* Set DirectSound Cooperative level, ie what control we want over Windows + * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the + * settings of the primary buffer, but also that only the sound of our + * application will be hearable when it will have the focus. + * !!! (this is not really working as intended yet because to set the + * cooperative level you need the window handle of your application, and + * I don't know of any easy way to get it. Especially since we might play + * sound without any video, and so what window handle should we use ??? + * The hack for now is to use the Desktop window handle - it seems to be + * working */ + if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), + DSSCL_EXCLUSIVE)) + { + mp_msg(MSGT_AO, MSGL_ERR, + "ao_dsound: cannot set direct sound cooperative level\n"); + IDirectSound_Release(hds); + FreeLibrary(hdsound_dll); + return 0; + } + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n"); + + memset(&dscaps, 0, sizeof(DSCAPS)); + dscaps.dwSize = sizeof(DSCAPS); + if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) { + if (dscaps.dwFlags & DSCAPS_EMULDRIVER) + mp_msg(MSGT_AO, MSGL_V, + "ao_dsound: DirectSound is emulated, waveOut may give better performance\n"); + } else { + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n"); + } + + return 1; } /** @@ -268,14 +280,14 @@ static int InitDirectSound(void) */ static void DestroyBuffer(void) { - if (hdsbuf) { - IDirectSoundBuffer_Release(hdsbuf); - hdsbuf = NULL; - } - if (hdspribuf) { - IDirectSoundBuffer_Release(hdspribuf); - hdspribuf = NULL; - } + if (hdsbuf) { + IDirectSoundBuffer_Release(hdsbuf); + hdsbuf = NULL; + } + if (hdspribuf) { + IDirectSoundBuffer_Release(hdspribuf); + hdspribuf = NULL; + } } /** @@ -286,57 +298,58 @@ static void DestroyBuffer(void) */ static int write_buffer(unsigned char *data, int len) { - HRESULT res; - LPVOID lpvPtr1; - DWORD dwBytes1; - LPVOID lpvPtr2; - DWORD dwBytes2; - - underrun_check = 0; - - // Lock the buffer - res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); - // If the buffer was lost, restore and retry lock. - if (DSERR_BUFFERLOST == res) - { - IDirectSoundBuffer_Restore(hdsbuf); - res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); - } - - - if (SUCCEEDED(res)) - { - if (!AF_FORMAT_IS_AC3(ao_data.format)) { + HRESULT res; + LPVOID lpvPtr1; + DWORD dwBytes1; + LPVOID lpvPtr2; + DWORD dwBytes2; + + underrun_check = 0; + + // Lock the buffer + res = IDirectSoundBuffer_Lock(hdsbuf, write_offset, len, &lpvPtr1, &dwBytes1, + &lpvPtr2, &dwBytes2, 0); + // If the buffer was lost, restore and retry lock. + if (DSERR_BUFFERLOST == res) { + IDirectSoundBuffer_Restore(hdsbuf); + res = IDirectSoundBuffer_Lock(hdsbuf, write_offset, len, &lpvPtr1, + &dwBytes1, &lpvPtr2, &dwBytes2, 0); + } + + + if (SUCCEEDED(res)) { + if (!AF_FORMAT_IS_AC3(ao_data.format)) { memcpy(lpvPtr1, data, dwBytes1); if (lpvPtr2 != NULL) memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2); - write_offset+=dwBytes1+dwBytes2; - if(write_offset>=buffer_size) - write_offset=dwBytes2; - } else { - // Write to pointers without reordering. - memcpy(lpvPtr1,data,dwBytes1); - if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2); - write_offset+=dwBytes1+dwBytes2; - if(write_offset>=buffer_size)write_offset=dwBytes2; - } - - // Release the data back to DirectSound. - res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2); - if (SUCCEEDED(res)) - { - // Success. - DWORD status; - IDirectSoundBuffer_GetStatus(hdsbuf, &status); - if (!(status & DSBSTATUS_PLAYING)){ - res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); - } - return dwBytes1+dwBytes2; + write_offset += dwBytes1 + dwBytes2; + if (write_offset >= buffer_size) + write_offset = dwBytes2; + } else { + // Write to pointers without reordering. + memcpy(lpvPtr1, data, dwBytes1); + if (NULL != lpvPtr2) + memcpy(lpvPtr2, data + dwBytes1, dwBytes2); + write_offset += dwBytes1 + dwBytes2; + if (write_offset >= buffer_size) + write_offset = dwBytes2; + } + + // Release the data back to DirectSound. + res = IDirectSoundBuffer_Unlock(hdsbuf, lpvPtr1, dwBytes1, lpvPtr2, + dwBytes2); + if (SUCCEEDED(res)) { + // Success. + DWORD status; + IDirectSoundBuffer_GetStatus(hdsbuf, &status); + if (!(status & DSBSTATUS_PLAYING)) + res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); + return dwBytes1 + dwBytes2; + } } - } - // Lock, Unlock, or Restore failed. - return 0; + // Lock, Unlock, or Restore failed. + return 0; } /***************************************************************************************/ @@ -349,24 +362,24 @@ static int write_buffer(unsigned char *data, int len) */ static int control(int cmd, void *arg) { - DWORD volume; - switch (cmd) { - case AOCONTROL_GET_VOLUME: { - ao_control_vol_t* vol = (ao_control_vol_t*)arg; - vol->left = vol->right = audio_volume; - return CONTROL_OK; - } - case AOCONTROL_SET_VOLUME: { - ao_control_vol_t* vol = (ao_control_vol_t*)arg; - volume = audio_volume = vol->right; - if (volume < 1) - volume = 1; - volume = (DWORD)(log10(volume) * 5000.0) - 10000; - IDirectSoundBuffer_SetVolume(hdsbuf, volume); - return CONTROL_OK; - } - } - return -1; + DWORD volume; + switch (cmd) { + case AOCONTROL_GET_VOLUME: { + ao_control_vol_t *vol = (ao_control_vol_t *)arg; + vol->left = vol->right = audio_volume; + return CONTROL_OK; + } + case AOCONTROL_SET_VOLUME: { + ao_control_vol_t *vol = (ao_control_vol_t *)arg; + volume = audio_volume = vol->right; + if (volume < 1) + volume = 1; + volume = (DWORD)(log10(volume) * 5000.0) - 10000; + IDirectSoundBuffer_SetVolume(hdsbuf, volume); + return CONTROL_OK; + } + } + return -1; } /** @@ -380,118 +393,138 @@ static int control(int cmd, void *arg) static int init(int rate, const struct mp_chmap *channels, int format, int flags) { int res; - if (!InitDirectSound()) return 0; + if (!InitDirectSound()) + return 0; + + global_ao->no_persistent_volume = true; + audio_volume = 100; + + // ok, now create the buffers + WAVEFORMATEXTENSIBLE wformat; + DSBUFFERDESC dsbpridesc; + DSBUFFERDESC dsbdesc; + + if (AF_FORMAT_IS_AC3(format)) + format = AF_FORMAT_AC3_NE; + else { + struct mp_chmap_sel sel = { + 0 + }; + mp_chmap_sel_add_waveext(&sel); + if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels)) + return 0; + } + switch (format) { + case AF_FORMAT_AC3_NE: + case AF_FORMAT_S24_LE: + case AF_FORMAT_S16_LE: + case AF_FORMAT_U8: + break; + default: + mp_msg(MSGT_AO, MSGL_V, + "ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n", + af_fmt2str_short(format)); + format = AF_FORMAT_S16_LE; + } + //fill global ao_data + ao_data.samplerate = rate; + ao_data.format = format; + ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format) >> 3); + if (ao_data.buffersize == -1) + ao_data.buffersize = ao_data.bps; // space for 1 sec + mp_msg(MSGT_AO, MSGL_V, + "ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, + ao_data.channels.num, af_fmt2str_short(format)); + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Buffersize:%d bytes (%d msec)\n", + ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000); + + //fill waveformatex + ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); + wformat.Format.cbSize = (ao_data.channels.num > 2) + ? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0; + wformat.Format.nChannels = ao_data.channels.num; + wformat.Format.nSamplesPerSec = rate; + if (AF_FORMAT_IS_AC3(format)) { + wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; + wformat.Format.wBitsPerSample = 16; + wformat.Format.nBlockAlign = 4; + } else { + wformat.Format.wFormatTag = (ao_data.channels.num > 2) + ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; + wformat.Format.wBitsPerSample = af_fmt2bits(format); + wformat.Format.nBlockAlign = wformat.Format.nChannels * + (wformat.Format.wBitsPerSample >> 3); + } - global_ao->no_persistent_volume = true; - audio_volume = 100; + // fill in primary sound buffer descriptor + memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC)); + dsbpridesc.dwSize = sizeof(DSBUFFERDESC); + dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER; + dsbpridesc.dwBufferBytes = 0; + dsbpridesc.lpwfxFormat = NULL; + + // fill in the secondary sound buffer (=stream buffer) descriptor + memset(&dsbdesc, 0, sizeof(DSBUFFERDESC)); + dsbdesc.dwSize = sizeof(DSBUFFERDESC); + dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */ + | DSBCAPS_GLOBALFOCUS /** Allows background playing */ + | DSBCAPS_CTRLVOLUME; /** volume control enabled */ + + if (ao_data.channels.num > 2) { + wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels); + wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; + wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; + // Needed for 5.1 on emu101k - shit soundblaster + dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE; + } + wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * + wformat.Format.nBlockAlign; + + dsbdesc.dwBufferBytes = ao_data.buffersize; + dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; + buffer_size = dsbdesc.dwBufferBytes; + write_offset = 0; + min_free_space = wformat.Format.nBlockAlign; + ao_data.outburst = wformat.Format.nBlockAlign * 512; + + // create primary buffer and set its format + + res = IDirectSound_CreateSoundBuffer(hds, &dsbpridesc, &hdspribuf, NULL); + if (res != DS_OK) { + UninitDirectSound(); + mp_msg(MSGT_AO, MSGL_ERR, + "ao_dsound: cannot create primary buffer (%s)\n", + dserr2str(res)); + return 0; + } + res = IDirectSoundBuffer_SetFormat(hdspribuf, (WAVEFORMATEX *)&wformat); + 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)); + } - // ok, now create the buffers - WAVEFORMATEXTENSIBLE wformat; - DSBUFFERDESC dsbpridesc; - DSBUFFERDESC dsbdesc; + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n"); - if (AF_FORMAT_IS_AC3(format)) { - format = AF_FORMAT_AC3_NE; - } else { - struct mp_chmap_sel sel = {0}; - mp_chmap_sel_add_waveext(&sel); - if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels)) - return 0; + // now create the stream buffer + + res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); + if (res != DS_OK) { + if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) { + // Try without DSBCAPS_LOCHARDWARE + dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE; + res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); } - switch(format){ - case AF_FORMAT_AC3_NE: - case AF_FORMAT_S24_LE: - case AF_FORMAT_S16_LE: - case AF_FORMAT_U8: - break; - default: - mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format)); - format=AF_FORMAT_S16_LE; - } - //fill global ao_data - ao_data.samplerate = rate; - ao_data.format = format; - ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format)>>3); - if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec - mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, ao_data.channels.num, af_fmt2str_short(format)); - mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000); - - //fill waveformatex - ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); - wformat.Format.cbSize = (ao_data.channels.num > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; - wformat.Format.nChannels = ao_data.channels.num; - wformat.Format.nSamplesPerSec = rate; - if (AF_FORMAT_IS_AC3(format)) { - wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; - wformat.Format.wBitsPerSample = 16; - wformat.Format.nBlockAlign = 4; - } else { - wformat.Format.wFormatTag = (ao_data.channels.num > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; - wformat.Format.wBitsPerSample = af_fmt2bits(format); - wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); - } - - // fill in primary sound buffer descriptor - memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC)); - dsbpridesc.dwSize = sizeof(DSBUFFERDESC); - dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER; - dsbpridesc.dwBufferBytes = 0; - dsbpridesc.lpwfxFormat = NULL; - - - // fill in the secondary sound buffer (=stream buffer) descriptor - memset(&dsbdesc, 0, sizeof(DSBUFFERDESC)); - dsbdesc.dwSize = sizeof(DSBUFFERDESC); - dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */ - | DSBCAPS_GLOBALFOCUS /** Allows background playing */ - | DSBCAPS_CTRLVOLUME; /** volume control enabled */ - - if (ao_data.channels.num > 2) { - wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels); - wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; - wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; - // Needed for 5.1 on emu101k - shit soundblaster - dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE; - } - wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; - - dsbdesc.dwBufferBytes = ao_data.buffersize; - dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; - buffer_size = dsbdesc.dwBufferBytes; - write_offset = 0; - min_free_space = wformat.Format.nBlockAlign; - ao_data.outburst = wformat.Format.nBlockAlign * 512; - - // create primary buffer and set its format - - res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL ); - if ( res != DS_OK ) { - UninitDirectSound(); - mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res)); - return 0; - } - res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat ); - 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)); - - mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n"); - - // now create the stream buffer - - res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); - if (res != DS_OK) { - if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) { - // Try without DSBCAPS_LOCHARDWARE - dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE; - res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); - } - if (res != DS_OK) { - UninitDirectSound(); - mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res)); - return 0; - } - } - mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); - return 1; + if (res != DS_OK) { + UninitDirectSound(); + mp_msg(MSGT_AO, MSGL_ERR, + "ao_dsound: cannot create secondary (stream)buffer (%s)\n", + dserr2str(res)); + return 0; + } + } + mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); + return 1; } @@ -501,11 +534,11 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags */ static void reset(void) { - IDirectSoundBuffer_Stop(hdsbuf); - // reset directsound buffer - IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0); - write_offset=0; - underrun_check=0; + IDirectSoundBuffer_Stop(hdsbuf); + // reset directsound buffer + IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0); + write_offset = 0; + underrun_check = 0; } /** @@ -513,7 +546,7 @@ static void reset(void) */ static void audio_pause(void) { - IDirectSoundBuffer_Stop(hdsbuf); + IDirectSoundBuffer_Stop(hdsbuf); } /** @@ -521,7 +554,7 @@ static void audio_pause(void) */ static void audio_resume(void) { - IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); + IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); } /** @@ -530,51 +563,53 @@ static void audio_resume(void) */ static void uninit(int immed) { - if (!immed) - mp_sleep_us(get_delay() * 1000000); - reset(); + if (!immed) + mp_sleep_us(get_delay() * 1000000); + reset(); - DestroyBuffer(); - UninitDirectSound(); + DestroyBuffer(); + UninitDirectSound(); } // return exact number of free (safe to write) bytes static int check_free_buffer_size(void) { - int space; - DWORD play_offset; - IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); - space=buffer_size-(write_offset-play_offset); - // | | <-- const --> | | | - // buffer start play_cursor write_cursor write_offset buffer end - // play_cursor is the actual postion of the play cursor - // write_cursor is the position after which it is assumed to be save to write data - // write_offset is the postion where we actually write the data to - if(space > buffer_size)space -= buffer_size; // write_offset < play_offset - // Check for buffer underruns. An underrun happens if DirectSound - // started to play old data beyond the current write_offset. Detect this - // by checking whether the free space shrinks, even though no data was - // written (i.e. no write_buffer). Doesn't always work, but the only - // reason we need this is to deal with the situation when playback ends, - // and the buffer is only half-filled. - if (space < underrun_check) { - // there's no useful data in the buffers - space = buffer_size; - reset(); - } - underrun_check = space; - return space; + int space; + DWORD play_offset; + IDirectSoundBuffer_GetCurrentPosition(hdsbuf, &play_offset, NULL); + space = buffer_size - (write_offset - play_offset); + // | | <-- const --> | | | + // buffer start play_cursor write_cursor write_offset buffer end + // play_cursor is the actual postion of the play cursor + // write_cursor is the position after which it is assumed to be save to write data + // write_offset is the postion where we actually write the data to + if (space > buffer_size) + space -= buffer_size; // write_offset < play_offset + // Check for buffer underruns. An underrun happens if DirectSound + // started to play old data beyond the current write_offset. Detect this + // by checking whether the free space shrinks, even though no data was + // written (i.e. no write_buffer). Doesn't always work, but the only + // reason we need this is to deal with the situation when playback ends, + // and the buffer is only half-filled. + if (space < underrun_check) { + // there's no useful data in the buffers + space = buffer_size; + reset(); + } + underrun_check = space; + return space; } /** -\brief find out how many bytes can be written into the audio buffer without -\return free space in bytes, has to return 0 if the buffer is almost full -*/ + \brief find out how many bytes can be written into the audio buffer without + \return free space in bytes, has to return 0 if the buffer is almost full + */ static int get_space(void) { - int space = check_free_buffer_size(); - if(space < min_free_space)return 0; - return space-min_free_space; + int space = check_free_buffer_size(); + if (space < min_free_space) + return 0; + return space - min_free_space; } /** @@ -584,14 +619,15 @@ static int get_space(void) \param flags currently unused \return number of played bytes */ -static int play(void* data, int len, int flags) +static int play(void *data, int len, int flags) { - int space = check_free_buffer_size(); - if(space < len) len = space; + int space = check_free_buffer_size(); + if (space < len) + len = space; - if (!(flags & AOPLAY_FINAL_CHUNK)) - len = (len / ao_data.outburst) * ao_data.outburst; - return write_buffer(data, len); + if (!(flags & AOPLAY_FINAL_CHUNK)) + len = (len / ao_data.outburst) * ao_data.outburst; + return write_buffer(data, len); } /** @@ -600,6 +636,6 @@ static int play(void* data, int len, int flags) */ static float get_delay(void) { - int space = check_free_buffer_size(); - return (float)(buffer_size - space) / (float)ao_data.bps; + int space = check_free_buffer_size(); + return (float)(buffer_size - space) / (float)ao_data.bps; } -- cgit v1.2.3