Mercurial > mplayer.hg
view libao2/ao_dsound.c @ 13929:bfaca342e3c4
MSS2 can decode also MSS1
xanim h261 codec does not work
author | rtognimp |
---|---|
date | Sat, 13 Nov 2004 23:39:20 +0000 |
parents | b7322244e53c |
children | a92101a7eb49 |
line wrap: on
line source
/****************************************************************************** * ao_dsound.c: Windows DirectSound interface for MPlayer * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA. * *****************************************************************************/ /** \todo verify/extend multichannel support */ #include <stdio.h> #include <stdlib.h> #include <windows.h> #define DIRECTSOUND_VERSION 0x0600 #include <dsound.h> #include "afmt.h" #include "audio_out.h" #include "audio_out_internal.h" #include "../mp_msg.h" #include "../libvo/fastmemcpy.h" #include "osdep/timer.h" static ao_info_t info = { "Windows DirectSound audio output", "dsound", "Gabor Szecsi <deje@miki.hu>", "" }; LIBAO_EXTERN(dsound) /** \todo use the definitions from the win32 api headers when they define these */ #if 1 #define WAVE_FORMAT_IEEE_FLOAT 0x0003 #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}}; #define SPEAKER_FRONT_LEFT 0x1 #define SPEAKER_FRONT_RIGHT 0x2 #define SPEAKER_FRONT_CENTER 0x4 #define SPEAKER_LOW_FREQUENCY 0x8 #define SPEAKER_BACK_LEFT 0x10 #define SPEAKER_BACK_RIGHT 0x20 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 #define SPEAKER_BACK_CENTER 0x100 #define SPEAKER_SIDE_LEFT 0x200 #define SPEAKER_SIDE_RIGHT 0x400 #define SPEAKER_TOP_CENTER 0x800 #define SPEAKER_TOP_FRONT_LEFT 0x1000 #define SPEAKER_TOP_FRONT_CENTER 0x2000 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 #define SPEAKER_TOP_BACK_LEFT 0x8000 #define SPEAKER_TOP_BACK_CENTER 0x10000 #define SPEAKER_TOP_BACK_RIGHT 0x20000 #define SPEAKER_RESERVED 0x80000000 #define DSSPEAKER_HEADPHONE 0x00000001 #define DSSPEAKER_MONO 0x00000002 #define DSSPEAKER_QUAD 0x00000003 #define DSSPEAKER_STEREO 0x00000004 #define DSSPEAKER_SURROUND 0x00000005 #define DSSPEAKER_5POINT1 0x00000006 #ifndef _WAVEFORMATEXTENSIBLE_ typedef struct { 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 */ /* present in stream */ GUID SubFormat; } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; #endif #endif static const int channel_mask[] = { SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT, SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY, SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY }; static HINSTANCE hdsound_dll = NULL; ///handle to the dll static LPDIRECTSOUND hds = NULL; ///direct sound object static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer) static int buffer_size = 0; ///size in bytes of the direct sound buffer static int write_offset = 0; ///offset of the write cursor in the direct sound buffer static int min_free_space = 4096; ///if the free space is below this value get_space() will return 0 /***************************************************************************************/ /** \brief output error message \param err error code \return string with the error message */ 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"; } } /** \brief uninitialize direct sound */ static void UninitDirectSound(void) { // finally release the DirectSound object if (hds) { IDirectSound_Release(hds); hds = NULL; } // free DSOUND.DLL if (hdsound_dll) { FreeLibrary(hdsound_dll); hdsound_dll = NULL; } mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); } /** \brief initilize direct sound \return 0 if error, 1 if ok */ static int InitDirectSound(void) { DSCAPS dscaps; // initialize directsound HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN); 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"); if (OurDirectSoundCreate == NULL) { mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); FreeLibrary(hdsound_dll); return 0; } // Create the direct sound object if FAILED(OurDirectSoundCreate(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; } /** \brief destroy the direct sound buffer */ static void DestroyBuffer(void) { if (hdsbuf) { IDirectSoundBuffer_Release(hdsbuf); hdsbuf = NULL; } if (hdspribuf) { IDirectSoundBuffer_Release(hdspribuf); hdspribuf = NULL; } } /** \brief fill sound buffer \param data pointer to the sound data to copy \param len length of the data to copy in bytes \return number of copyed bytes */ static int write_buffer(unsigned char *data, int len) { HRESULT res; LPVOID lpvPtr1; DWORD dwBytes1; LPVOID lpvPtr2; DWORD dwBytes2; // 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( (ao_data.channels == 6) && (ao_data.format!=AFMT_AC3) ) { // reorder channels while writing to pointers. // it's this easy because buffer size and len are always // aligned to multiples of channels*bytespersample // there's probably some room for speed improvements here const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix" int i, j; int numsamp,sampsize; sampsize = audio_out_format_bits(ao_data.format)>>3; // bytes per sample numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); } if (NULL != lpvPtr2 ) { numsamp = dwBytes2 / (ao_data.channels * sampsize); for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); } } 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; } /***************************************************************************************/ /** \brief handle control commands \param cmd command \param arg argument \return CONTROL_OK or -1 in case the command can't be handled */ 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; IDirectSoundBuffer_GetVolume(hdsbuf, &volume); vol->left = vol->right = (float)(volume+10000) / 100.0; //printf("ao_dsound: volume: %f\n",vol->left); return CONTROL_OK; } case AOCONTROL_SET_VOLUME: { ao_control_vol_t* vol = (ao_control_vol_t*)arg; volume = (vol->right * 100.0)-10000; IDirectSoundBuffer_SetVolume(hdsbuf, volume); //printf("ao_dsound: volume: %f\n",vol->left); return CONTROL_OK; } } return -1; } /** \brief setup sound device \param rate samplerate \param channels number of channels \param format format \param flags unused \return 1=success 0=fail */ static int init(int rate, int channels, int format, int flags) { int res; if (!InitDirectSound()) return 0; // ok, now create the buffers WAVEFORMATEXTENSIBLE wformat; DSBUFFERDESC dsbpridesc; DSBUFFERDESC dsbdesc; //fill global ao_data ao_data.channels = channels; ao_data.samplerate = rate; ao_data.format = format; ao_data.bps = channels * rate * (audio_out_format_bits(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, channels, audio_out_format_name(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 = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; wformat.Format.nChannels = channels; wformat.Format.nSamplesPerSec = rate; if (format == AFMT_AC3) { wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; wformat.Format.wBitsPerSample = 16; wformat.Format.nBlockAlign = 4; } else { wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; wformat.Format.wBitsPerSample = audio_out_format_bits(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 (channels > 2) { wformat.dwChannelMask = channel_mask[channels - 3]; 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; 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; } /** \brief stop playing and empty buffers (for seeking/pause) */ static void reset() { IDirectSoundBuffer_Stop(hdsbuf); // reset directsound buffer IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0); write_offset=0; } /** \brief stop playing, keep buffers (for pause) */ static void audio_pause() { IDirectSoundBuffer_Stop(hdsbuf); } /** \brief resume playing, after audio_pause() */ static void audio_resume() { IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); } /** \brief close audio device \param immed stop playback immediately, currently not supported */ static void uninit(int immed) { reset(); DestroyBuffer(); UninitDirectSound(); } /** \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() { 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 if(space < min_free_space)return 0; return space; } /** \brief play 'len' bytes of 'data' \param data pointer to the data to play \param len size in bytes of the data buffer, gets rounded down to outburst*n \param flags currently unused \return number of played bytes */ static int play(void* data, int len, int flags) { DWORD play_offset; int space; // make sure we have enough space to write data IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); space=buffer_size-(write_offset-play_offset); if(space > buffer_size)space -= buffer_size; // write_offset < play_offset if(space < len) len = space; len = (len / ao_data.outburst) * ao_data.outburst; return write_buffer(data, len); } /** \brief get the delay between the first and last sample in the buffer \return delay in seconds */ static float get_delay() { DWORD play_offset; int space; IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); space=play_offset-write_offset; if(space <= 0)space += buffer_size; return (float)(buffer_size - space) / (float)ao_data.bps; }