mirror of https://github.com/mpv-player/mpv
ao_dsound: uncrustify
This commit is contained in:
wm4
parent
92ae48db0f
commit
cee56e8623
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@ -47,10 +47,10 @@
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static const ao_info_t info =
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{
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"Windows DirectSound audio output",
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"dsound",
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"Gabor Szecsi <deje@miki.hu>",
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""
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"Windows DirectSound audio output",
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"dsound",
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"Gabor Szecsi <deje@miki.hu>",
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""
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};
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LIBAO_EXTERN(dsound)
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@ -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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@ -75,15 +77,15 @@ static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00
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#ifndef _WAVEFORMATEXTENSIBLE_
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typedef struct {
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WAVEFORMATEX Format;
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WAVEFORMATEX Format;
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union {
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WORD wValidBitsPerSample; /* bits of precision */
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WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */
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WORD wReserved; /* If neither applies, set to zero. */
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} Samples;
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DWORD dwChannelMask; /* which channels are */
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DWORD dwChannelMask; /* which channels are */
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/* present in stream */
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GUID SubFormat;
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GUID SubFormat;
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} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
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#endif
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@ -111,28 +113,28 @@ static int audio_volume;
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*/
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static char * dserr2str(int err)
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{
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switch (err) {
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case DS_OK: return "DS_OK";
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case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
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case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
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case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
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case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
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case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
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case DSERR_GENERIC: return "DSERR_GENERIC";
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case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
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case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
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case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
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case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
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case DSERR_NODRIVER: return "DSERR_NODRIVER";
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case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
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case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
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case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
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case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
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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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switch (err) {
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case DS_OK: return "DS_OK";
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case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION";
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case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION";
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case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL";
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case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM";
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case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL";
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case DSERR_GENERIC: return "DSERR_GENERIC";
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case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED";
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case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY";
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case DSERR_BADFORMAT: return "DSERR_BADFORMAT";
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case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED";
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case DSERR_NODRIVER: return "DSERR_NODRIVER";
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case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED";
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case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION";
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case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST";
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case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO";
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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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}
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return "unknown";
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}
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/**
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@ -142,15 +144,15 @@ static void UninitDirectSound(void)
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{
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// finally release the DirectSound object
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if (hds) {
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IDirectSound_Release(hds);
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hds = NULL;
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IDirectSound_Release(hds);
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hds = NULL;
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}
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// free DSOUND.DLL
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if (hdsound_dll) {
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FreeLibrary(hdsound_dll);
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hdsound_dll = NULL;
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FreeLibrary(hdsound_dll);
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hdsound_dll = NULL;
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}
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n");
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n");
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}
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/**
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@ -158,7 +160,7 @@ static void UninitDirectSound(void)
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*/
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static void print_help(void)
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{
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mp_msg(MSGT_AO, MSGL_FATAL,
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mp_msg(MSGT_AO, MSGL_FATAL,
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"\n-ao dsound commandline help:\n"
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"Example: mpv -ao dsound:device=1\n"
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" sets 1st device\n"
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@ -172,17 +174,17 @@ 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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memcpy(&device,guid,sizeof(GUID));
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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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memcpy(&device, guid, sizeof(GUID));
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}
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mp_msg(MSGT_AO, MSGL_V,"\n");
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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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}
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@ -194,73 +196,83 @@ static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOI
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*/
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static int InitDirectSound(void)
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{
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DSCAPS dscaps;
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DSCAPS dscaps;
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// initialize directsound
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// initialize directsound
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HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
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HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
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int device_index=0;
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const opt_t subopts[] = {
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{"device", OPT_ARG_INT, &device_num,NULL},
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{NULL}
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};
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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print_help();
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return 0;
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}
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HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
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int device_index = 0;
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const opt_t subopts[] = {
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{"device", OPT_ARG_INT, &device_num, NULL},
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{NULL}
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};
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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print_help();
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return 0;
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}
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hdsound_dll = LoadLibrary("DSOUND.DLL");
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if (hdsound_dll == NULL) {
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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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hdsound_dll = LoadLibrary("DSOUND.DLL");
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if (hdsound_dll == NULL) {
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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,
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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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FreeLibrary(hdsound_dll);
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return 0;
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}
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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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FreeLibrary(hdsound_dll);
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return 0;
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}
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// Enumerate all directsound devices
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mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
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OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
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// Enumerate all directsound devices
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Output Devices:\n");
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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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FreeLibrary(hdsound_dll);
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return 0;
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}
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// Create the direct sound object
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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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/* Set DirectSound Cooperative level, ie what control we want over Windows
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* sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
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* settings of the primary buffer, but also that only the sound of our
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* application will be hearable when it will have the focus.
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* !!! (this is not really working as intended yet because to set the
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* cooperative level you need the window handle of your application, and
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* I don't know of any easy way to get it. Especially since we might play
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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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IDirectSound_Release(hds);
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FreeLibrary(hdsound_dll);
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return 0;
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}
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n");
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/* Set DirectSound Cooperative level, ie what control we want over Windows
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* sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
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* settings of the primary buffer, but also that only the sound of our
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* application will be hearable when it will have the focus.
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* !!! (this is not really working as intended yet because to set the
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* cooperative level you need the window handle of your application, and
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* I don't know of any easy way to get it. Especially since we might play
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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(),
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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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}
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n");
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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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} 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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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)
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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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return 1;
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return 1;
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}
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/**
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@ -268,14 +280,14 @@ static int InitDirectSound(void)
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*/
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static void DestroyBuffer(void)
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{
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if (hdsbuf) {
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IDirectSoundBuffer_Release(hdsbuf);
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hdsbuf = NULL;
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}
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if (hdspribuf) {
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IDirectSoundBuffer_Release(hdspribuf);
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hdspribuf = NULL;
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}
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if (hdsbuf) {
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IDirectSoundBuffer_Release(hdsbuf);
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hdsbuf = NULL;
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}
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if (hdspribuf) {
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IDirectSoundBuffer_Release(hdspribuf);
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hdspribuf = NULL;
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}
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}
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/**
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@ -286,57 +298,58 @@ static void DestroyBuffer(void)
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*/
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static int write_buffer(unsigned char *data, int len)
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{
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HRESULT res;
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LPVOID lpvPtr1;
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DWORD dwBytes1;
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LPVOID lpvPtr2;
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DWORD dwBytes2;
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HRESULT res;
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LPVOID lpvPtr1;
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DWORD dwBytes1;
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LPVOID lpvPtr2;
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DWORD dwBytes2;
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underrun_check = 0;
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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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// 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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IDirectSoundBuffer_Restore(hdsbuf);
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res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
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}
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// Lock the buffer
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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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IDirectSoundBuffer_Restore(hdsbuf);
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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 (!AF_FORMAT_IS_AC3(ao_data.format)) {
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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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memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
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write_offset+=dwBytes1+dwBytes2;
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if(write_offset>=buffer_size)
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write_offset=dwBytes2;
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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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write_offset+=dwBytes1+dwBytes2;
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if(write_offset>=buffer_size)write_offset=dwBytes2;
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}
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write_offset += dwBytes1 + dwBytes2;
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if (write_offset >= buffer_size)
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write_offset = dwBytes2;
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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)
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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)
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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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// 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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res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
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}
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return dwBytes1+dwBytes2;
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// Release the data back to DirectSound.
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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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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;
|
||||
global_ao->no_persistent_volume = true;
|
||||
audio_volume = 100;
|
||||
|
||||
// ok, now create the buffers
|
||||
WAVEFORMATEXTENSIBLE wformat;
|
||||
DSBUFFERDESC dsbpridesc;
|
||||
DSBUFFERDESC dsbdesc;
|
||||
// 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;
|
||||
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);
|
||||
}
|
||||
|
||||
// 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);
|
||||
}
|
||||
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;
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue