Merge svn changes up to r28862

25 files changed, 424 insertions, 99 deletions

diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 194890ff9a..9541f9c553 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -57,7 +57,7 @@
 #include "audio_out_internal.h"
 #include "libaf/af_format.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
     "ALSA-0.9.x-1.x audio output",
     "alsa",
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index abf2eff827..007a5f1b4f 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -32,7 +32,7 @@
 #include "mp_msg.h"
 #include "help_mp.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
     "ALSA-0.5.x audio output",
     "alsa5",
diff --git a/libao2/ao_arts.c b/libao2/ao_arts.c
index 30db48fb22..df44db4f02 100644
--- a/libao2/ao_arts.c
+++ b/libao2/ao_arts.c
@@ -38,7 +38,7 @@
 
 static arts_stream_t stream;
 
-static ao_info_t info =
+static const ao_info_t info =
 {
     "aRts audio output",
     "arts",
diff --git a/libao2/ao_dart.c b/libao2/ao_dart.c
new file mode 100644
index 0000000000..e9bff8df1b
--- /dev/null
+++ b/libao2/ao_dart.c
@@ -0,0 +1,334 @@
+/*
+ * OS/2 DART audio output driver
+ *
+ * Copyright (c) 2007-2009 by KO Myung-Hun (komh@chollian.net)
+ *
+ * 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.
+ */
+
+#define INCL_DOS
+#define INCL_DOSERRORS
+#include <os2.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <float.h>
+
+#include <dart.h>
+
+#include "config.h"
+#include "libaf/af_format.h"
+#include "audio_out.h"
+#include "audio_out_internal.h"
+#include "mp_msg.h"
+#include "libvo/fastmemcpy.h"
+#include "subopt-helper.h"
+
+static const ao_info_t info = {
+    "DART audio output",
+    "dart",
+    "KO Myung-Hun <komh@chollian.net>",
+    ""
+};
+
+LIBAO_EXTERN(dart)
+
+#define OUTBURST_SAMPLES        512
+#define DEFAULT_DART_SAMPLES    (OUTBURST_SAMPLES << 2)
+
+#define CHUNK_SIZE  ao_data.outburst
+
+static uint8_t *m_audioBuf = NULL;
+
+static int m_nBufSize = 0;
+
+static volatile int m_fQuit = FALSE;
+// may only be modified by DART's playback thread or while it is stopped
+static volatile int m_iBufReadPos = 0;
+// may only be modified by MPlayer's thread
+static volatile int m_iBufWritePos = 0;
+
+// may only be called by MPlayer's thread
+// return value may change between immediately following two calls,
+// and the real number of free bytes might be larger!
+static int buf_free(void)
+{
+    int nFree = m_iBufReadPos - m_iBufWritePos - CHUNK_SIZE;
+
+    if (nFree < 0)
+        nFree += m_nBufSize;
+
+    return nFree;
+}
+
+// may only be called by DART's playback thread
+// return value may change between immediately following two calls,
+// and the real number of buffered bytes might be larger!
+static int buf_used(void)
+{
+    int nUsed = m_iBufWritePos - m_iBufReadPos;
+
+    if (nUsed < 0)
+        nUsed += m_nBufSize;
+
+    return nUsed;
+}
+
+static int write_buffer(unsigned char *data, int len)
+{
+    int nFirstLen = m_nBufSize - m_iBufWritePos;
+    int nFree = buf_free();
+
+    if (len > nFree)
+        len = nFree;
+
+    if (nFirstLen > len)
+        nFirstLen = len;
+
+    // till end of buffer
+    fast_memcpy(m_audioBuf + m_iBufWritePos, data, nFirstLen);
+    if (len > nFirstLen) { // we have to wrap around
+        // remaining part from beginning of buffer
+        fast_memcpy(m_audioBuf, data + nFirstLen, len - nFirstLen);
+    }
+
+    m_iBufWritePos = (m_iBufWritePos + len) % m_nBufSize;
+
+    return len;
+}
+
+static int read_buffer(unsigned char *data, int len)
+{
+    int nFirstLen = m_nBufSize - m_iBufReadPos;
+    int nBuffered = buf_used();
+
+    if (len > nBuffered)
+        len = nBuffered;
+
+    if (nFirstLen > len)
+        nFirstLen = len;
+
+    // till end of buffer
+    fast_memcpy(data, m_audioBuf + m_iBufReadPos, nFirstLen);
+    if (len > nFirstLen) { // we have to wrap around
+        // remaining part from beginning of buffer
+        fast_memcpy(data + nFirstLen, m_audioBuf, len - nFirstLen);
+    }
+
+    m_iBufReadPos = (m_iBufReadPos + len) % m_nBufSize;
+
+    return len;
+}
+
+// end ring buffer stuff
+
+static ULONG APIENTRY dart_audio_callback(PVOID pCBData, PVOID pBuffer,
+                                          ULONG ulSize)
+{
+    int nReadLen;
+
+    nReadLen = read_buffer(pBuffer, ulSize);
+    if (nReadLen < ulSize && !m_fQuit) {
+        memset((uint8_t *)pBuffer + nReadLen, DART.bSilence, ulSize - nReadLen);
+        nReadLen = ulSize;
+    }
+
+    return nReadLen;
+}
+
+// to set/get/query special features/parameters
+static int control(int cmd, void *arg)
+{
+    switch (cmd) {
+    case AOCONTROL_GET_VOLUME:
+        {
+        ao_control_vol_t *vol = arg;
+
+        vol->left = vol->right = LOUSHORT(dartGetVolume());
+
+        return CONTROL_OK;
+        }
+
+    case AOCONTROL_SET_VOLUME:
+        {
+        int mid;
+        ao_control_vol_t *vol = arg;
+
+        mid = (vol->left + vol->right) / 2;
+        dartSetVolume(MCI_SET_AUDIO_ALL, mid);
+
+        return CONTROL_OK;
+        }
+    }
+
+    return CONTROL_UNKNOWN;
+}
+
+static void print_help(void)
+{
+    mp_msg(MSGT_AO, MSGL_FATAL,
+           "\n-ao dart commandline help:\n"
+           "Example: mplayer -ao dart:noshare\n"
+           "    open DART in exclusive mode\n"
+           "\nOptions:\n"
+           "    (no)share\n"
+           "        Open DART in shareable or exclusive mode\n"
+           "    bufsize=<size>\n"
+           "        Set buffer size to <size> in samples(default: 2048)\n");
+}
+
+// open & set up audio device
+// return: 1=success 0=fail
+static int init(int rate, int channels, int format, int flags)
+{
+    int fShare = 1;
+    int nDartSamples = DEFAULT_DART_SAMPLES;
+    int nBytesPerSample;
+
+    opt_t subopts[] = {
+        {"share", OPT_ARG_BOOL, &fShare, NULL},
+        {"bufsize", OPT_ARG_INT, &nDartSamples, (opt_test_f)int_non_neg},
+        {NULL}
+    };
+
+    if (subopt_parse(ao_subdevice, subopts) != 0) {
+        print_help();
+        return 0;
+    }
+
+    if (!nDartSamples)
+        nDartSamples = DEFAULT_DART_SAMPLES;
+
+    mp_msg(MSGT_AO, MSGL_V, "DART: opened in %s mode, buffer size = %d sample(s)\n",
+           fShare ? "shareable" : "exclusive", nDartSamples);
+
+    switch (format) {
+    case AF_FORMAT_S16_LE:
+    case AF_FORMAT_S8:
+        break;
+
+    default:
+        format = AF_FORMAT_S16_LE;
+        mp_msg(MSGT_AO, MSGL_V, "DART: format %s not supported defaulting to Signed 16-bit Little-Endian\n",
+               af_fmt2str_short(format));
+        break;
+    }
+
+    nBytesPerSample = (af_fmt2bits(format) >> 3) * channels;
+
+    if (dartInit(0, af_fmt2bits(format), rate, MCI_WAVE_FORMAT_PCM, channels,
+                 2, nBytesPerSample * nDartSamples, fShare,
+                 dart_audio_callback, NULL))
+        return 0;
+
+    mp_msg(MSGT_AO, MSGL_V, "DART: obtained buffer size = %lu bytes\n",
+           DART.ulBufferSize);
+
+    m_fQuit = FALSE;
+
+    ao_data.channels    = channels;
+    ao_data.samplerate  = rate;
+    ao_data.format      = format;
+    ao_data.bps         = nBytesPerSample * rate;
+    ao_data.outburst    = nBytesPerSample * OUTBURST_SAMPLES;
+    ao_data.buffersize  = DART.ulBufferSize;
+
+    // multiple of CHUNK_SIZE
+    m_nBufSize = ((DART.ulBufferSize << 2) / CHUNK_SIZE) * CHUNK_SIZE;
+    // and one more chunk plus round up
+    m_nBufSize += 2 * CHUNK_SIZE;
+
+    m_audioBuf = malloc(m_nBufSize);
+
+    m_iBufReadPos  = 0;
+    m_iBufWritePos = 0;
+
+    dartPlay();
+
+    // might cause PM DLLs to be loaded which incorrectly enable SIG_FPE,
+    // which AAC decoding might trigger.
+    // so, mask off all floating-point exceptions.
+    _control87(MCW_EM, MCW_EM);
+
+    return 1;
+}
+
+// close audio device
+static void uninit(int immed)
+{
+    m_fQuit = TRUE;
+
+    if (!immed) {
+        while (DART.fPlaying)
+            DosSleep(1);
+    }
+
+    dartClose();
+
+    free(m_audioBuf);
+}
+
+// stop playing and empty buffers (for seeking/pause)
+static void reset(void)
+{
+    dartPause();
+
+    // Reset ring-buffer state
+    m_iBufReadPos  = 0;
+    m_iBufWritePos = 0;
+
+    dartResume();
+}
+
+// stop playing, keep buffers (for pause)
+static void audio_pause(void)
+{
+    dartPause();
+}
+
+// resume playing, after audio_pause()
+static void audio_resume(void)
+{
+    dartResume();
+}
+
+// return: how many bytes can be played without blocking
+static int get_space(void)
+{
+    return buf_free();
+}
+
+// plays 'len' bytes of 'data'
+// it should round it down to outburst*n
+// return: number of bytes played
+static int play(void *data, int len, int flags)
+{
+
+    if (!(flags & AOPLAY_FINAL_CHUNK))
+        len = (len / ao_data.outburst) * ao_data.outburst;
+
+    return write_buffer(data, len);
+}
+
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(void)
+{
+    int nBuffered = m_nBufSize - CHUNK_SIZE - buf_free(); // could be less
+
+    return (float)nBuffered / (float)ao_data.bps;
+}
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index acd59983dd..729084b145 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -42,7 +42,7 @@
 #include "subopt-helper.h"
 
 
-static ao_info_t info =
+static const ao_info_t info =
 {
 	"Windows DirectSound audio output",
 	"dsound",
@@ -431,7 +431,7 @@ static int init(int rate, int channels, int format, int flags)
 		case AF_FORMAT_AC3:
 		case AF_FORMAT_S24_LE:
 		case AF_FORMAT_S16_LE:
-		case AF_FORMAT_S8:
+		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));
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index cd92ea325c..7f9594fb4d 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -36,7 +36,7 @@
 #include "libmpdemux/mpeg_packetizer.h"
 
 
-static ao_info_t info =
+static const ao_info_t info =
 {
 	"DXR2 audio output",
 	"dxr2",
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
index a5602ff894..d54f843c65 100644
--- a/libao2/ao_esd.c
+++ b/libao2/ao_esd.c
@@ -67,7 +67,7 @@
 #define	ESD_CLIENT_NAME	"MPlayer"
 #define	ESD_MAX_DELAY	(1.0f)	/* max amount of data buffered in esd (#sec) */
 
-static ao_info_t info =
+static const ao_info_t info =
 {
     "EsounD audio output",
     "esd",
diff --git a/libao2/ao_ivtv.c b/libao2/ao_ivtv.c
index 71b58c0637..2b182ff191 100644
--- a/libao2/ao_ivtv.c
+++ b/libao2/ao_ivtv.c
@@ -41,7 +41,7 @@
 
 static int freq = 0;
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
   "IVTV MPEG Audio Decoder output",
   "ivtv",
diff --git a/libao2/ao_jack.c b/libao2/ao_jack.c
index d7f2092a43..e861d08aed 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -40,7 +40,7 @@
 
 #include <jack/jack.h>
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
   "JACK audio output",
   "jack",
diff --git a/libao2/ao_macosx.c b/libao2/ao_macosx.c
index fcbda6f1a1..fb5883b3fb 100644
--- a/libao2/ao_macosx.c
+++ b/libao2/ao_macosx.c
@@ -53,7 +53,7 @@
 #include "libaf/af_format.h"
 #include "osdep/timer.h"
 
-static ao_info_t info =
+static const ao_info_t info =
   {
     "Darwin/Mac OS X native audio output",
     "macosx",
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index 5af124fd42..88ab903ed1 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -63,7 +63,7 @@ int vo_mpegpes_fd2 = -1;
 
 #include <errno.h>
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 #ifdef CONFIG_DVB
 	"DVB audio output",
diff --git a/libao2/ao_nas.c b/libao2/ao_nas.c
index 9394649cc0..e5e675be3f 100644
--- a/libao2/ao_nas.c
+++ b/libao2/ao_nas.c
@@ -119,7 +119,7 @@ static const char* nas_state(unsigned int state) {
 	return nas_states[state];
 }
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"NAS audio output",
 	"nas",
diff --git a/libao2/ao_null.c b/libao2/ao_null.c
index 2e28b8f1e8..20b715fb96 100644
--- a/libao2/ao_null.c
+++ b/libao2/ao_null.c
@@ -27,7 +27,7 @@
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"Null audio output",
 	"null",
diff --git a/libao2/ao_openal.c b/libao2/ao_openal.c
index c91694dc05..c80c49b27a 100644
--- a/libao2/ao_openal.c
+++ b/libao2/ao_openal.c
@@ -42,7 +42,7 @@
 #include "osdep/timer.h"
 #include "subopt-helper.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
   "OpenAL audio output",
   "openal",
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index a98d5d1922..e4688d1723 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -48,7 +48,7 @@
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"OSS/ioctl audio output",
 	"oss",
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index bd66ed0dab..bf3f224e81 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -35,7 +35,7 @@
 #include "help_mp.h"
 
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"RAW PCM/WAVE file writer audio output",
 	"pcm",
diff --git a/libao2/ao_pulse.c b/libao2/ao_pulse.c
index d6974f85e0..2d27c85540 100644
--- a/libao2/ao_pulse.c
+++ b/libao2/ao_pulse.c
@@ -33,7 +33,7 @@
 #define PULSE_CLIENT_NAME "MPlayer"
 
 /** General driver info */
-static ao_info_t info = {
+static const ao_info_t info = {
     "PulseAudio audio output",
     "pulse",
     "Lennart Poettering",
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index 645c9e9685..31693be229 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -38,7 +38,7 @@
 
 #include "libvo/fastmemcpy.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"SDLlib audio output",
 	"sdl",
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index db7c486c83..46b79c1e2e 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -32,7 +32,7 @@
 #include "help_mp.h"
 #include "libaf/af_format.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
 	"sgi audio output",
 	"sgi",
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c
index cae2287f6c..725a49e324 100644
--- a/libao2/ao_sun.c
+++ b/libao2/ao_sun.c
@@ -49,7 +49,7 @@
 #include "mp_msg.h"
 #include "help_mp.h"
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
     "Sun audio output",
     "sun",
diff --git a/libao2/ao_v4l2.c b/libao2/ao_v4l2.c
index eca46f01ac..a57cea91c3 100644
--- a/libao2/ao_v4l2.c
+++ b/libao2/ao_v4l2.c
@@ -38,7 +38,7 @@
 
 static int freq = 0;
 
-static ao_info_t info = 
+static const ao_info_t info = 
 {
   "V4L2 MPEG Audio Decoder output",
   "v4l2",
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index 40830eb233..c29e05935f 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -87,12 +87,10 @@ static const int channel_mask[] = {
 static WAVEHDR*     waveBlocks;         //pointer to our ringbuffer memory
 static HWAVEOUT     hWaveOut;           //handle to the waveout device
 static unsigned int buf_write=0;
-static unsigned int buf_write_pos=0;
-static int          full_buffers=0;
-static int          buffered_bytes=0;
+static volatile int buf_read=0;
 
 
-static ao_info_t info = 
+static const ao_info_t info =
 {
 	"Windows waveOut audio output",
 	"win32",
@@ -102,17 +100,12 @@ static ao_info_t info =
 
 LIBAO_EXTERN(win32)
 
-static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,  
+static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
     DWORD dwParam1,DWORD dwParam2)
 {
 	if(uMsg != WOM_DONE)
         return;
-	if (full_buffers) {
-		buffered_bytes-=BUFFER_SIZE;
-		--full_buffers;
-	} else {
-		buffered_bytes=0;
-	}
+	buf_read = (buf_read + 1) % BUFFER_COUNT;
 }
 
 // to set/get/query special features/parameters
@@ -145,34 +138,34 @@ static int control(int cmd,void *arg)
 // return: 1=success 0=fail
 static int init(int rate,int channels,int format,int flags)
 {
-	WAVEFORMATEXTENSIBLE wformat;      
-	DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
+	WAVEFORMATEXTENSIBLE wformat;
 	MMRESULT result;
 	unsigned char* buffer;
 	int i;
-   
+
 	switch(format){
 		case AF_FORMAT_AC3:
 		case AF_FORMAT_S24_LE:
 		case AF_FORMAT_S16_LE:
-		case AF_FORMAT_S8:
+		case AF_FORMAT_U8:
 			break;
 		default:
 			mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
 			format=AF_FORMAT_S16_LE;
-	}   
+	}
 
 	// FIXME multichannel mode is buggy
 	if(channels > 2)
 		channels = 2;
-   
-	//fill global ao_data 
+
+	//fill global ao_data
 	ao_data.channels=channels;
 	ao_data.samplerate=rate;
 	ao_data.format=format;
 	ao_data.bps=channels*rate;
 	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 	  ao_data.bps*=2;
+	ao_data.outburst = BUFFER_SIZE;
 	if(ao_data.buffersize==-1)
 	{
 		ao_data.buffersize=af_fmt2bits(format)/8;
@@ -181,22 +174,22 @@ static int init(int rate,int channels,int format,int flags)
 	}
 	mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
     mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
-	
+
 	//fill waveformatex
     ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
     wformat.Format.cbSize          = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
-    wformat.Format.nChannels       = channels;                
-    wformat.Format.nSamplesPerSec  = rate;            
+    wformat.Format.nChannels       = channels;
+    wformat.Format.nSamplesPerSec  = rate;
     if(format == AF_FORMAT_AC3)
     {
         wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
         wformat.Format.wBitsPerSample  = 16;
         wformat.Format.nBlockAlign     = 4;
     }
-    else 
+    else
     {
         wformat.Format.wFormatTag      = (channels>2)?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.wBitsPerSample >> 3);
     }
 	if(channels>2)
@@ -205,9 +198,9 @@ static int init(int rate,int channels,int format,int flags)
         wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
 	    wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
     }
-  
+
     wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
- 	
+ 
     //open sound device
     //WAVE_MAPPER always points to the default wave device on the system
     result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
@@ -231,9 +224,8 @@ static int init(int rate,int channels,int format,int flags)
 		return 0;
     }
 	//allocate buffer memory as one big block
-	buffer = malloc(totalBufferSize);
-	memset(buffer,0x0,totalBufferSize);
-    //and setup pointers to each buffer 
+	buffer = calloc(BUFFER_COUNT, BUFFER_SIZE + sizeof(WAVEHDR));
+    //and setup pointers to each buffer
     waveBlocks = (WAVEHDR*)buffer;
     buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
     for(i = 0; i < BUFFER_COUNT; i++) {
@@ -241,9 +233,7 @@ static int init(int rate,int channels,int format,int flags)
         buffer += BUFFER_SIZE;
     }
     buf_write=0;
-    buf_write_pos=0;
-    full_buffers=0;
-    buffered_bytes=0;
+    buf_read=0;
 
     return 1;
 }
@@ -251,8 +241,8 @@ static int init(int rate,int channels,int format,int flags)
 // close audio device
 static void uninit(int immed)
 {
-    if(!immed)while(buffered_bytes > 0)usec_sleep(50000);
-    else buffered_bytes=0;
+    if(!immed)
+	usec_sleep(get_delay() * 1000 * 1000);
 	waveOutReset(hWaveOut);
 	waveOutClose(hWaveOut);
 	mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
@@ -265,9 +255,7 @@ static void reset(void)
 {
    	waveOutReset(hWaveOut);
 	buf_write=0;
-    buf_write_pos=0;
-	full_buffers=0;
-	buffered_bytes=0;
+	buf_read=0;
 }
 
 // stop playing, keep buffers (for pause)
@@ -285,7 +273,9 @@ static void audio_resume(void)
 // return: how many bytes can be played without blocking
 static int get_space(void)
 {
-    return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes;
+    int free = buf_read - buf_write - 1;
+    if (free < 0) free += BUFFER_COUNT;
+    return free * BUFFER_SIZE;
 }
 
 //writes data into buffer, based on ringbuffer code in ao_sdl.c
@@ -293,28 +283,23 @@ static int write_waveOutBuffer(unsigned char* data,int len){
   WAVEHDR* current;
   int len2=0;
   int x;
-  while(len>0){                       
+  while(len>0){
+    int buf_next = (buf_write + 1) % BUFFER_COUNT;
     current = &waveBlocks[buf_write];
-    if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break;
+    if(buf_next == buf_read) break;
     //unprepare the header if it is prepared
-	if(current->dwFlags & WHDR_PREPARED) 
+	if(current->dwFlags & WHDR_PREPARED)
            waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
-	x=BUFFER_SIZE-buf_write_pos;          
-    if(x>len) x=len;                   
-    fast_memcpy(current->lpData+buf_write_pos,data+len2,x); 
-    if(buf_write_pos==0)full_buffers++;
-    len2+=x; len-=x;                 
-	buffered_bytes+=x; buf_write_pos+=x; 
+	x=BUFFER_SIZE;
+    if(x>len) x=len;
+    fast_memcpy(current->lpData,data+len2,x);
+    len2+=x; len-=x;
 	//prepare header and write data to device
-	current->dwBufferLength = buf_write_pos;
+	current->dwBufferLength = x;
 	waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
 	waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
-    
-	if(buf_write_pos>=BUFFER_SIZE){        //buffer is full find next
-       // block is full, find next!
-       buf_write=(buf_write+1)%BUFFER_COUNT;  
-	   buf_write_pos=0;                 
-    }                                 
+
+       buf_write = buf_next;
   }
   return len2;
 }
@@ -332,5 +317,7 @@ static int play(void* data,int len,int flags)
 // return: delay in seconds between first and last sample in buffer
 static float get_delay(void)
 {
-	return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
+	int used = buf_write - buf_read;
+	if (used < 0) used += BUFFER_COUNT;
+	return (float)(used * BUFFER_SIZE + ao_data.buffersize)/(float)ao_data.bps;
 }
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index ee62ebc016..8c64abbb95 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -30,28 +30,29 @@
 ao_data_t ao_data={0,0,0,0,OUTBURST,-1,0};
 char *ao_subdevice = NULL;
 
-extern ao_functions_t audio_out_oss;
-extern ao_functions_t audio_out_macosx;
-extern ao_functions_t audio_out_arts;
-extern ao_functions_t audio_out_esd;
-extern ao_functions_t audio_out_pulse;
-extern ao_functions_t audio_out_jack;
-extern ao_functions_t audio_out_openal;
-extern ao_functions_t audio_out_null;
-extern ao_functions_t audio_out_alsa5;
-extern ao_functions_t audio_out_alsa;
-extern ao_functions_t audio_out_nas;
-extern ao_functions_t audio_out_sdl;
-extern ao_functions_t audio_out_sun;
-extern ao_functions_t audio_out_sgi;