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Code Issues Releases Wiki Activity

ao_dsound: switch to new AO API

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wm4 2013-06-04 00:59:53 +02:00
parent cee56e8623
commit 8afcb84ee5
1 changed files with 70 additions and 61 deletions
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131
audio/out/ao_dsound.c
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@ -39,22 +39,10 @@
#include "audio/format.h" #include "audio/format.h"
#include "ao.h" #include "ao.h"
#include "audio/reorder_ch.h" #include "audio/reorder_ch.h"
#include "audio_out_internal.h"
#include "core/mp_msg.h" #include "core/mp_msg.h"
#include "osdep/timer.h" #include "osdep/timer.h"
#include "core/subopt-helper.h" #include "core/subopt-helper.h"
static const 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 \todo use the definitions from the win32 api headers when they define these
*/ */
@ -104,6 +92,8 @@ static int device_num = 0; ///wanted device number
static GUID device; ///guid of the device static GUID device; ///guid of the device
static int audio_volume; static int audio_volume;
static float get_delay(struct ao *ao);
/***************************************************************************************/ /***************************************************************************************/
/** /**
@ -296,7 +286,7 @@ static void DestroyBuffer(void)
\param len length of the data to copy in bytes \param len length of the data to copy in bytes
\return number of copyed bytes \return number of copyed bytes
*/ */
static int write_buffer(unsigned char *data, int len) static int write_buffer(struct ao *ao, unsigned char *data, int len)
{ {
HRESULT res; HRESULT res;
LPVOID lpvPtr1; LPVOID lpvPtr1;
@ -318,7 +308,7 @@ static int write_buffer(unsigned char *data, int len)
if (SUCCEEDED(res)) { if (SUCCEEDED(res)) {
if (!AF_FORMAT_IS_AC3(ao_data.format)) { if (!AF_FORMAT_IS_AC3(ao->format)) {
memcpy(lpvPtr1, data, dwBytes1); memcpy(lpvPtr1, data, dwBytes1);
if (lpvPtr2 != NULL) if (lpvPtr2 != NULL)
memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2); memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
@ -358,9 +348,9 @@ static int write_buffer(unsigned char *data, int len)
\brief handle control commands \brief handle control commands
\param cmd command \param cmd command
\param arg argument \param arg argument
\return CONTROL_OK or -1 in case the command can't be handled \return CONTROL_OK or CONTROL_UNKNOWN in case the command is not supported
*/ */
static int control(int cmd, void *arg) static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{ {
DWORD volume; DWORD volume;
switch (cmd) { switch (cmd) {
@ -379,7 +369,7 @@ static int control(int cmd, void *arg)
return CONTROL_OK; return CONTROL_OK;
} }
} }
return -1; return CONTROL_UNKNOWN;
} }
/** /**
@ -388,31 +378,31 @@ static int control(int cmd, void *arg)
\param channels number of channels \param channels number of channels
\param format format \param format format
\param flags unused \param flags unused
\return 1=success 0=fail \return 0=success -1=fail
*/ */
static int init(int rate, const struct mp_chmap *channels, int format, int flags) static int init(struct ao *ao, char *params)
{ {
int res; int res;
if (!InitDirectSound()) if (!InitDirectSound())
return 0; return -1;
global_ao->no_persistent_volume = true; ao->no_persistent_volume = true;
audio_volume = 100; audio_volume = 100;
// ok, now create the buffers // ok, now create the buffers
WAVEFORMATEXTENSIBLE wformat; WAVEFORMATEXTENSIBLE wformat;
DSBUFFERDESC dsbpridesc; DSBUFFERDESC dsbpridesc;
DSBUFFERDESC dsbdesc; DSBUFFERDESC dsbdesc;
int format = ao->format;
int rate = ao->samplerate;
if (AF_FORMAT_IS_AC3(format)) if (AF_FORMAT_IS_AC3(format))
format = AF_FORMAT_AC3_NE; format = AF_FORMAT_AC3_NE;
else { else {
struct mp_chmap_sel sel = { struct mp_chmap_sel sel = {0};
0
};
mp_chmap_sel_add_waveext(&sel); mp_chmap_sel_add_waveext(&sel);
if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels)) if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return 0; return -1;
} }
switch (format) { switch (format) {
case AF_FORMAT_AC3_NE: case AF_FORMAT_AC3_NE:
@ -426,30 +416,30 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
af_fmt2str_short(format)); af_fmt2str_short(format));
format = AF_FORMAT_S16_LE; format = AF_FORMAT_S16_LE;
} }
//fill global ao_data //set our audio parameters
ao_data.samplerate = rate; ao->samplerate = rate;
ao_data.format = format; ao->format = format;
ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format) >> 3); ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
if (ao_data.buffersize == -1) if (ao->buffersize == -1)
ao_data.buffersize = ao_data.bps; // space for 1 sec ao->buffersize = ao->bps; // space for 1 sec
mp_msg(MSGT_AO, MSGL_V, mp_msg(MSGT_AO, MSGL_V,
"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, "ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate,
ao_data.channels.num, af_fmt2str_short(format)); ao->channels.num, af_fmt2str_short(format));
mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Buffersize:%d bytes (%d msec)\n", mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Buffersize:%d bytes (%d msec)\n",
ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000); ao->buffersize, ao->buffersize / ao->bps * 1000);
//fill waveformatex //fill waveformatex
ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
wformat.Format.cbSize = (ao_data.channels.num > 2) wformat.Format.cbSize = (ao->channels.num > 2)
? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0; ? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
wformat.Format.nChannels = ao_data.channels.num; wformat.Format.nChannels = ao->channels.num;
wformat.Format.nSamplesPerSec = rate; wformat.Format.nSamplesPerSec = rate;
if (AF_FORMAT_IS_AC3(format)) { if (AF_FORMAT_IS_AC3(format)) {
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wformat.Format.wBitsPerSample = 16; wformat.Format.wBitsPerSample = 16;
wformat.Format.nBlockAlign = 4; wformat.Format.nBlockAlign = 4;
} else { } else {
wformat.Format.wFormatTag = (ao_data.channels.num > 2) wformat.Format.wFormatTag = (ao->channels.num > 2)
? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
wformat.Format.wBitsPerSample = af_fmt2bits(format); wformat.Format.wBitsPerSample = af_fmt2bits(format);
wformat.Format.nBlockAlign = wformat.Format.nChannels * wformat.Format.nBlockAlign = wformat.Format.nChannels *
@ -470,8 +460,8 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
| DSBCAPS_GLOBALFOCUS /** Allows background playing */ | DSBCAPS_GLOBALFOCUS /** Allows background playing */
| DSBCAPS_CTRLVOLUME; /** volume control enabled */ | DSBCAPS_CTRLVOLUME; /** volume control enabled */
if (ao_data.channels.num > 2) { if (ao->channels.num > 2) {
wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels); wformat.dwChannelMask = mp_chmap_to_waveext(&ao->channels);
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
// Needed for 5.1 on emu101k - shit soundblaster // Needed for 5.1 on emu101k - shit soundblaster
@ -480,12 +470,12 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec *
wformat.Format.nBlockAlign; wformat.Format.nBlockAlign;
dsbdesc.dwBufferBytes = ao_data.buffersize; dsbdesc.dwBufferBytes = ao->buffersize;
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
buffer_size = dsbdesc.dwBufferBytes; buffer_size = dsbdesc.dwBufferBytes;
write_offset = 0; write_offset = 0;
min_free_space = wformat.Format.nBlockAlign; min_free_space = wformat.Format.nBlockAlign;
ao_data.outburst = wformat.Format.nBlockAlign * 512; ao->outburst = wformat.Format.nBlockAlign * 512;
// create primary buffer and set its format // create primary buffer and set its format
@ -495,7 +485,7 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
mp_msg(MSGT_AO, MSGL_ERR, mp_msg(MSGT_AO, MSGL_ERR,
"ao_dsound: cannot create primary buffer (%s)\n", "ao_dsound: cannot create primary buffer (%s)\n",
dserr2str(res)); dserr2str(res));
return 0; return -1;
} }
res = IDirectSoundBuffer_SetFormat(hdspribuf, (WAVEFORMATEX *)&wformat); res = IDirectSoundBuffer_SetFormat(hdspribuf, (WAVEFORMATEX *)&wformat);
if (res != DS_OK) { if (res != DS_OK) {
@ -520,11 +510,11 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
mp_msg(MSGT_AO, MSGL_ERR, mp_msg(MSGT_AO, MSGL_ERR,
"ao_dsound: cannot create secondary (stream)buffer (%s)\n", "ao_dsound: cannot create secondary (stream)buffer (%s)\n",
dserr2str(res)); dserr2str(res));
return 0; return -1;
} }
} }
mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
return 1; return 0;
} }
@ -532,7 +522,7 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
/** /**
\brief stop playing and empty buffers (for seeking/pause) \brief stop playing and empty buffers (for seeking/pause)
*/ */
static void reset(void) static void reset(struct ao *ao)
{ {
IDirectSoundBuffer_Stop(hdsbuf); IDirectSoundBuffer_Stop(hdsbuf);
// reset directsound buffer // reset directsound buffer
@ -544,7 +534,7 @@ static void reset(void)
/** /**
\brief stop playing, keep buffers (for pause) \brief stop playing, keep buffers (for pause)
*/ */
static void audio_pause(void) static void audio_pause(struct ao *ao)
{ {
IDirectSoundBuffer_Stop(hdsbuf); IDirectSoundBuffer_Stop(hdsbuf);
} }
@ -552,7 +542,7 @@ static void audio_pause(void)
/** /**
\brief resume playing, after audio_pause() \brief resume playing, after audio_pause()
*/ */
static void audio_resume(void) static void audio_resume(struct ao *ao)
{ {
IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
} }
@ -561,18 +551,18 @@ static void audio_resume(void)
\brief close audio device \brief close audio device
\param immed stop playback immediately \param immed stop playback immediately
*/ */
static void uninit(int immed) static void uninit(struct ao *ao, bool immed)
{ {
if (!immed) if (!immed)
mp_sleep_us(get_delay() * 1000000); mp_sleep_us(get_delay(ao) * 1000000);
reset(); reset(ao);
DestroyBuffer(); DestroyBuffer();
UninitDirectSound(); UninitDirectSound();
} }
// return exact number of free (safe to write) bytes // return exact number of free (safe to write) bytes
static int check_free_buffer_size(void) static int check_free_buffer_size(struct ao *ao)
{ {
int space; int space;
DWORD play_offset; DWORD play_offset;
@ -594,7 +584,7 @@ static int check_free_buffer_size(void)
if (space < underrun_check) { if (space < underrun_check) {
// there's no useful data in the buffers // there's no useful data in the buffers
space = buffer_size; space = buffer_size;
reset(); reset(ao);
} }
underrun_check = space; underrun_check = space;
return space; return space;
@ -604,9 +594,9 @@ static int check_free_buffer_size(void)
\brief find out how many bytes can be written into the audio buffer without \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 \return free space in bytes, has to return 0 if the buffer is almost full
*/ */
static int get_space(void) static int get_space(struct ao *ao)
{ {
int space = check_free_buffer_size(); int space = check_free_buffer_size(ao);
if (space < min_free_space) if (space < min_free_space)
return 0; return 0;
return space - min_free_space; return space - min_free_space;
@ -616,26 +606,45 @@ static int get_space(void)
\brief play 'len' bytes of 'data' \brief play 'len' bytes of 'data'
\param data pointer to the data to play \param data pointer to the data to play
\param len size in bytes of the data buffer, gets rounded down to outburst*n \param len size in bytes of the data buffer, gets rounded down to outburst*n
NOTE: outburst stuff might be outdated/deprecated
\param flags currently unused \param flags currently unused
\return number of played bytes \return number of played bytes
*/ */
static int play(void *data, int len, int flags) static int play(struct ao *ao, void *data, int len, int flags)
{ {
int space = check_free_buffer_size(); int space = check_free_buffer_size(ao);
if (space < len) if (space < len)
len = space; len = space;
if (!(flags & AOPLAY_FINAL_CHUNK)) if (!(flags & AOPLAY_FINAL_CHUNK))
len = (len / ao_data.outburst) * ao_data.outburst; len = (len / ao->outburst) * ao->outburst;
return write_buffer(data, len); return write_buffer(ao, data, len);
} }
/** /**
\brief get the delay between the first and last sample in the buffer \brief get the delay between the first and last sample in the buffer
\return delay in seconds \return delay in seconds
*/ */
static float get_delay(void) static float get_delay(struct ao *ao)
{ {
int space = check_free_buffer_size(); int space = check_free_buffer_size(ao);
return (float)(buffer_size - space) / (float)ao_data.bps; return (float)(buffer_size - space) / (float)ao->bps;
} }
const struct ao_driver audio_out_dsound = {
.info = &(const struct ao_info) {
"Windows DirectSound audio output",
"dsound",
"Gabor Szecsi <deje@miki.hu>",
""
},
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
};