[Mplayer-cvslog] CVS: main/libao2 ao_dsound.c,1.1,1.2
Sascha Sommer CVS
syncmail at mplayerhq.hu
Thu Oct 14 15:03:06 CEST 2004
CVS change done by Sascha Sommer CVS
Update of /cvsroot/mplayer/main/libao2
In directory mail:/var2/tmp/cvs-serv17264/libao2
Modified Files:
ao_dsound.c
Log Message:
ao dsound improvements patch by Florian Dietrich <flodt8 at yahoo.de>
Index: ao_dsound.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_dsound.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -r1.1 -r1.2
--- ao_dsound.c 25 Sep 2004 15:34:42 -0000 1.1
+++ ao_dsound.c 14 Oct 2004 13:03:03 -0000 1.2
@@ -25,7 +25,7 @@
#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
-#include <mmsystem.h>
+#define DIRECTSOUND_VERSION 0x0600
#include <dsound.h>
#include "afmt.h"
@@ -108,13 +108,12 @@
static HINSTANCE hdsound_dll = NULL; ///handle to the dll
static LPDIRECTSOUND hds = NULL; ///direct sound object
-static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///direct sound buffer
+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
-#define BUFFERSIZE 32767 /// in samples - at 48khz 0.6 sec buffer, gets multiplied with nBlockAlign
-
/***************************************************************************************/
/**
@@ -234,6 +233,10 @@
IDirectSoundBuffer_Release(hdsbuf);
hdsbuf = NULL;
}
+ if (hdspribuf) {
+ IDirectSoundBuffer_Release(hdspribuf);
+ hdspribuf = NULL;
+ }
}
/**
@@ -250,6 +253,15 @@
LPVOID lpvPtr2;
DWORD dwBytes2;
+ 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;
+
// Lock the buffer
res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
// If the buffer was lost, restore and retry lock.
@@ -266,7 +278,7 @@
memcpy(lpvPtr1,data,dwBytes1);
if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
write_offset+=dwBytes1+dwBytes2;
- if(write_offset>=buffer_size)write_offset-=buffer_size;
+ if(write_offset>=buffer_size)write_offset=dwBytes2;
// Release the data back to DirectSound.
res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
@@ -328,8 +340,9 @@
int res;
if (!InitDirectSound()) return 0;
- // ok, now create the primary buffer
+ // ok, now create the buffers
WAVEFORMATEXTENSIBLE wformat;
+ DSBUFFERDESC dsbpridesc;
DSBUFFERDESC dsbdesc;
//fill global ao_data
@@ -337,18 +350,13 @@
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 = audio_out_format_bits(format) >> 3;
- ao_data.buffersize *= channels;
- ao_data.buffersize *= BUFFERSIZE;
- }
+ 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, BUFFERSIZE * 1000 / rate);
+ 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) : 0;
+ wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
wformat.Format.nChannels = channels;
wformat.Format.nSamplesPerSec = rate;
if (format == AFMT_AC3) {
@@ -361,7 +369,15 @@
wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
}
- // fill in the direct sound buffer descriptor
+ // 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 */
@@ -371,18 +387,29 @@
if (channels > 2) {
wformat.dwChannelMask = channel_mask[channels - 3];
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
- wformat.Samples.wValidBitsPerSample = audio_out_format_bits(format);
+ 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 = wformat.Format.nBlockAlign * BUFFERSIZE;
+ dsbdesc.dwBufferBytes = ao_data.buffersize;
dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
buffer_size = dsbdesc.dwBufferBytes;
ao_data.outburst = wformat.Format.nBlockAlign * 512;
- // now create the sound buffer
+ // 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));
+
+ // now create the stream buffer
res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
if (res != DS_OK) {
@@ -393,11 +420,11 @@
}
if (res != DS_OK) {
UninitDirectSound();
- mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary buffer (%s)\n", dserr2str(res));
+ 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 buffer created\n");
+ mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
return 1;
}
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