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mpv/audio/out/ao_dsound.c
wm4 380fc765e4 audio/out: prepare for non-interleaved audio 
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00

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/*
* Windows DirectSound interface
*
* Copyright (c) 2004 Gabor Szecsi <deje@miki.hu>
*
* This file is part of MPlayer.
*
* MPlayer 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.
*
* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
/**
\todo verify/extend multichannel support
*/
#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
#define DIRECTSOUND_VERSION 0x0600
#include <dsound.h>
#include <math.h>
#include <libavutil/avutil.h>
#include <libavutil/common.h>
#include "config.h"
#include "audio/format.h"
#include "ao.h"
#include "audio/reorder_ch.h"
#include "mpvcore/mp_msg.h"
#include "osdep/timer.h"
#include "mpvcore/m_option.h"
/**
\todo use the definitions from the win32 api headers when they define these
*/
#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}
};
#if 0
#define DSSPEAKER_HEADPHONE 0x00000001
#define DSSPEAKER_MONO 0x00000002
#define DSSPEAKER_QUAD 0x00000003
#define DSSPEAKER_STEREO 0x00000004
#define DSSPEAKER_SURROUND 0x00000005
#define DSSPEAKER_5POINT1 0x00000006
#endif
#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
struct priv {
HINSTANCE hdsound_dll; ///handle to the dll
LPDIRECTSOUND hds; ///direct sound object
LPDIRECTSOUNDBUFFER hdspribuf; ///primary direct sound buffer
LPDIRECTSOUNDBUFFER hdsbuf; ///secondary direct sound buffer (stream buffer)
int buffer_size; ///size in bytes of the direct sound buffer
int write_offset; ///offset of the write cursor in the direct sound buffer
int min_free_space; ///if the free space is below this value get_space() will return 0
///there will always be at least this amout of free space to prevent
///get_space() from returning wrong values when buffer is 100% full.
///will be replaced with nBlockAlign in init()
int underrun_check; ///0 or last reported free space (underrun detection)
int device_num; ///wanted device number
GUID device; ///guid of the device
int audio_volume;
int device_index;
int outburst; ///play in multiple of chunks of this size
int cfg_device;
};
static float get_delay(struct ao *ao);
/***************************************************************************************/
/**
\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";
}
return "unknown";
}
/**
\brief uninitialize direct sound
*/
static void UninitDirectSound(struct ao *ao)
{
struct priv *p = ao->priv;
// finally release the DirectSound object
if (p->hds) {
IDirectSound_Release(p->hds);
p->hds = NULL;
}
// free DSOUND.DLL
if (p->hdsound_dll) {
FreeLibrary(p->hdsound_dll);
p->hdsound_dll = NULL;
}
MP_VERBOSE(ao, "DirectSound uninitialized\n");
}
/**
\brief enumerate direct sound devices
\return TRUE to continue with the enumeration
*/
static BOOL CALLBACK DirectSoundEnum(LPGUID guid, LPCSTR desc, LPCSTR module,
LPVOID context)
{
struct ao *ao = context;
struct priv *p = ao->priv;
MP_VERBOSE(ao, "%i %s ", p->device_index, desc);
if (p->device_num == p->device_index) {
MP_VERBOSE(ao, "<--");
if (guid)
memcpy(&p->device, guid, sizeof(GUID));
}
MP_VERBOSE(ao, "\n");
p->device_index++;
return TRUE;
}
/**
\brief initilize direct sound
\return 0 if error, 1 if ok
*/
static int InitDirectSound(struct ao *ao)
{
struct priv *p = ao->priv;
DSCAPS dscaps;
// initialize directsound
HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
p->device_index = 0;
p->device_num = p->cfg_device;
p->hdsound_dll = LoadLibrary("DSOUND.DLL");
if (p->hdsound_dll == NULL) {
MP_ERR(ao, "cannot load DSOUND.DLL\n");
return 0;
}
OurDirectSoundCreate = (void *)GetProcAddress(p->hdsound_dll,
"DirectSoundCreate");
OurDirectSoundEnumerate = (void *)GetProcAddress(p->hdsound_dll,
"DirectSoundEnumerateA");
if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) {
MP_ERR(ao, "GetProcAddress FAILED\n");
FreeLibrary(p->hdsound_dll);
return 0;
}
// Enumerate all directsound p->devices
MP_VERBOSE(ao, "Output Devices:\n");
OurDirectSoundEnumerate(DirectSoundEnum, ao);
// Create the direct sound object
if (FAILED(OurDirectSoundCreate((p->device_num) ? &p->device : NULL,
&p->hds, NULL)))
{
MP_ERR(ao, "cannot create a DirectSound device\n");
FreeLibrary(p->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(p->hds, GetDesktopWindow(),
DSSCL_EXCLUSIVE))
{
MP_ERR(ao, "cannot set direct sound cooperative level\n");
IDirectSound_Release(p->hds);
FreeLibrary(p->hdsound_dll);
return 0;
}
MP_VERBOSE(ao, "DirectSound initialized\n");
memset(&dscaps, 0, sizeof(DSCAPS));
dscaps.dwSize = sizeof(DSCAPS);
if (DS_OK == IDirectSound_GetCaps(p->hds, &dscaps)) {
if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
MP_VERBOSE(ao, "DirectSound is emulated\n");
} else {
MP_VERBOSE(ao, "cannot get device capabilities\n");
}
return 1;
}
/**
\brief destroy the direct sound buffer
*/
static void DestroyBuffer(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->hdsbuf) {
IDirectSoundBuffer_Release(p->hdsbuf);
p->hdsbuf = NULL;
}
if (p->hdspribuf) {
IDirectSoundBuffer_Release(p->hdspribuf);
p->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(struct ao *ao, unsigned char *data, int len)
{
struct priv *p = ao->priv;
HRESULT res;
LPVOID lpvPtr1;
DWORD dwBytes1;
LPVOID lpvPtr2;
DWORD dwBytes2;
p->underrun_check = 0;
// Lock the buffer
res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
&dwBytes1, &lpvPtr2, &dwBytes2, 0);
// If the buffer was lost, restore and retry lock.
if (DSERR_BUFFERLOST == res) {
IDirectSoundBuffer_Restore(p->hdsbuf);
res = IDirectSoundBuffer_Lock(p->hdsbuf, p->write_offset, len, &lpvPtr1,
&dwBytes1, &lpvPtr2, &dwBytes2, 0);
}
if (SUCCEEDED(res)) {
if (!AF_FORMAT_IS_AC3(ao->format)) {
memcpy(lpvPtr1, data, dwBytes1);
if (lpvPtr2 != NULL)
memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
p->write_offset += dwBytes1 + dwBytes2;
if (p->write_offset >= p->buffer_size)
p->write_offset = dwBytes2;
} else {
// Write to pointers without reordering.
memcpy(lpvPtr1, data, dwBytes1);
if (NULL != lpvPtr2)
memcpy(lpvPtr2, data + dwBytes1, dwBytes2);
p->write_offset += dwBytes1 + dwBytes2;
if (p->write_offset >= p->buffer_size)
p->write_offset = dwBytes2;
}
// Release the data back to DirectSound.
res = IDirectSoundBuffer_Unlock(p->hdsbuf, lpvPtr1, dwBytes1, lpvPtr2,
dwBytes2);
if (SUCCEEDED(res)) {
// Success.
DWORD status;
IDirectSoundBuffer_GetStatus(p->hdsbuf, &status);
if (!(status & DSBSTATUS_PLAYING))
res = IDirectSoundBuffer_Play(p->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 CONTROL_UNKNOWN in case the command is not supported
*/
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
DWORD volume;
switch (cmd) {
case AOCONTROL_GET_VOLUME: {
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
vol->left = vol->right = p->audio_volume;
return CONTROL_OK;
}
case AOCONTROL_SET_VOLUME: {
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
volume = p->audio_volume = vol->right;
if (volume < 1)
volume = 1;
volume = (DWORD)(log10(volume) * 5000.0) - 10000;
IDirectSoundBuffer_SetVolume(p->hdsbuf, volume);
return CONTROL_OK;
}
}
return CONTROL_UNKNOWN;
}
/**
\brief setup sound device
\param rate samplerate
\param channels number of channels
\param format format
\param flags unused
\return 0=success -1=fail
*/
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
int res;
if (!InitDirectSound(ao))
return -1;
ao->no_persistent_volume = true;
p->audio_volume = 100;
// ok, now create the buffers
WAVEFORMATEXTENSIBLE wformat;
DSBUFFERDESC dsbpridesc;
DSBUFFERDESC dsbdesc;
int format = af_fmt_from_planar(ao->format);
int rate = ao->samplerate;
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, &sel, &ao->channels))
return -1;
}
switch (format) {
case AF_FORMAT_AC3_NE:
case AF_FORMAT_S24_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_U8:
break;
default:
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
af_fmt_to_str(format));
format = AF_FORMAT_S16_LE;
}
//set our audio parameters
ao->samplerate = rate;
ao->format = format;
ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
int buffersize = ao->bps; // space for 1 sec
MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
ao->channels.num, af_fmt_to_str(format));
MP_VERBOSE(ao, "Buffersize:%d bytes (%d msec)\n",
buffersize, buffersize / ao->bps * 1000);
//fill waveformatex
ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
wformat.Format.cbSize = (ao->channels.num > 2)
? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
wformat.Format.nChannels = ao->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->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->channels.num > 2) {
wformat.dwChannelMask = mp_chmap_to_waveext(&ao->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 = buffersize;
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
p->buffer_size = dsbdesc.dwBufferBytes;
p->write_offset = 0;
p->min_free_space = wformat.Format.nBlockAlign;
p->outburst = wformat.Format.nBlockAlign * 512;
// create primary buffer and set its format
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbpridesc, &p->hdspribuf, NULL);
if (res != DS_OK) {
UninitDirectSound(ao);
MP_ERR(ao, "cannot create primary buffer (%s)\n", dserr2str(res));
return -1;
}
res = IDirectSoundBuffer_SetFormat(p->hdspribuf, (WAVEFORMATEX *)&wformat);
if (res != DS_OK) {
MP_WARN(ao, "cannot set primary buffer format (%s), using "
"standard setting (bad quality)", dserr2str(res));
}
MP_VERBOSE(ao, "primary buffer created\n");
// now create the stream buffer
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
if (res != DS_OK) {
if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) {
// Try without DSBCAPS_LOCHARDWARE
dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE;
res = IDirectSound_CreateSoundBuffer(p->hds, &dsbdesc, &p->hdsbuf, NULL);
}
if (res != DS_OK) {
UninitDirectSound(ao);
MP_ERR(ao, "cannot create secondary (stream)buffer (%s)\n",
dserr2str(res));
return -1;
}
}
MP_VERBOSE(ao, "secondary (stream)buffer created\n");
return 0;
}
/**
\brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
IDirectSoundBuffer_Stop(p->hdsbuf);
// reset directsound buffer
IDirectSoundBuffer_SetCurrentPosition(p->hdsbuf, 0);
p->write_offset = 0;
p->underrun_check = 0;
}
/**
\brief stop playing, keep buffers (for pause)
*/
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
IDirectSoundBuffer_Stop(p->hdsbuf);
}
/**
\brief resume playing, after audio_pause()
*/
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
IDirectSoundBuffer_Play(p->hdsbuf, 0, 0, DSBPLAY_LOOPING);
}
/**
\brief close audio device
\param immed stop playback immediately
*/
static void uninit(struct ao *ao, bool immed)
{
if (!immed)
mp_sleep_us(get_delay(ao) * 1000000);
reset(ao);
DestroyBuffer(ao);
UninitDirectSound(ao);
}
// return exact number of free (safe to write) bytes
static int check_free_buffer_size(struct ao *ao)
{
struct priv *p = ao->priv;
int space;
DWORD play_offset;
IDirectSoundBuffer_GetCurrentPosition(p->hdsbuf, &play_offset, NULL);
space = p->buffer_size - (p->write_offset - play_offset);
// | | <-- const --> | | |
// buffer start play_cursor write_cursor p->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
// p->write_offset is the postion where we actually write the data to
if (space > p->buffer_size)
space -= p->buffer_size; // p->write_offset < play_offset
// Check for buffer underruns. An underrun happens if DirectSound
// started to play old data beyond the current p->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 < p->underrun_check) {
// there's no useful data in the buffers
space = p->buffer_size;
reset(ao);
}
p->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
*/
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
int space = check_free_buffer_size(ao);
if (space < p->min_free_space)
return 0;
return (space - p->min_free_space) / ao->sstride;
}
/**
\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(struct ao *ao, void **data, int samples, int flags)
{
struct priv *p = ao->priv;
int len = samples * ao->sstride;
int space = check_free_buffer_size(ao);
if (space < len)
len = space;
if (!(flags & AOPLAY_FINAL_CHUNK))
len = (len / p->outburst) * p->outburst;
return write_buffer(ao, data[0], len) / ao->sstride;
}
/**
\brief get the delay between the first and last sample in the buffer
\return delay in seconds
*/
static float get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
int space = check_free_buffer_size(ao);
return (float)(p->buffer_size - space) / (float)ao->bps;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_dsound = {
.description = "Windows DirectSound audio output",
.name = "dsound",
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
.priv_size = sizeof(struct priv),
.options = (const struct m_option[]) {
OPT_INT("device", cfg_device, 0),
{0}
},
};
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