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authorfaust3 <faust3@b3059339-0415-0410-9bf9-f77b7e298cf2>2002-10-25 16:18:33 +0000
committerfaust3 <faust3@b3059339-0415-0410-9bf9-f77b7e298cf2>2002-10-25 16:18:33 +0000
commitf1ae8e8e5dc01b295b15c572471eff26285eb3a9 (patch)
tree56275e0b3918847836e3611cc800cf9ad9a9f42e /libao2
parentf8ce5ab876fcc40b01785c873688dff14f994848 (diff)
downloadmpv-f1ae8e8e5dc01b295b15c572471eff26285eb3a9.tar.bz2
mpv-f1ae8e8e5dc01b295b15c572471eff26285eb3a9.tar.xz
new windows waveout audio driver
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7914 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'libao2')
-rw-r--r--libao2/ao_win32.c249
1 files changed, 249 insertions, 0 deletions
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
new file mode 100644
index 0000000000..597e2cc993
--- /dev/null
+++ b/libao2/ao_win32.c
@@ -0,0 +1,249 @@
+/******************************************************************************
+ * ao_win32.c: Windows waveOut interface for MPlayer
+ * Copyright (c) 2002 Sascha Sommer <saschasommer@freenet.de>.
+ *
+ * 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.
+ *
+ *****************************************************************************/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <windows.h>
+#include <mmsystem.h>
+
+#include "afmt.h"
+#include "audio_out.h"
+#include "audio_out_internal.h"
+#include "../mp_msg.h"
+#include "../libvo/fastmemcpy.h"
+
+#define SAMPLESIZE 1024
+#define BUFFER_SIZE 4096
+#define BUFFER_COUNT 16
+
+
+static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory
+static HWAVEOUT hWaveOut; //handle to the waveout device
+static DWORD restoredvolume; //saves the volume to restore after playing
+static unsigned int buf_write=0;
+static unsigned int buf_write_pos=0;
+static int full_buffers=0;
+static int buffered_bytes=0;
+
+
+static ao_info_t info =
+{
+ "Windows waveOut audio output",
+ "win32",
+ "Sascha Sommer <saschasommer@freenet.de>",
+ ""
+};
+
+LIBAO_EXTERN(win32)
+
+static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
+ DWORD dwParam1,DWORD dwParam2)
+{
+ if(uMsg != WOM_DONE)
+ return;
+ if(full_buffers==0) return; //no more data buffered!
+ buffered_bytes=0; //buffer is empty
+ --full_buffers;
+}
+
+// to set/get/query special features/parameters
+static int control(int cmd,int arg)
+{
+ DWORD volume;
+ switch (cmd)
+ {
+ case AOCONTROL_GET_VOLUME:
+ {
+ ao_control_vol_t* vol = (ao_control_vol_t*)arg;
+ waveOutGetVolume(hWaveOut,&volume);
+ vol->left = (float)(LOWORD(volume)/655.35);
+ vol->right = (float)(HIWORD(volume)/655.35);
+ mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
+ return CONTROL_OK;
+ }
+ case AOCONTROL_SET_VOLUME:
+ {
+ ao_control_vol_t* vol = (ao_control_vol_t*)arg;
+ volume = MAKELONG(vol->left*655.35,vol->right*655.35);
+ waveOutSetVolume(hWaveOut,volume);
+ return CONTROL_OK;
+ }
+ }
+ return -1;
+}
+
+// open & setup audio device
+// return: 1=success 0=fail
+static int init(int rate,int channels,int format,int flags)
+{
+ WAVEFORMATEX wformat;
+ DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
+ MMRESULT result;
+ unsigned char* buffer;
+ int i;
+
+ //fill global ao_data
+ ao_data.channels=channels;
+ ao_data.samplerate=rate;
+ ao_data.format=format;
+ ao_data.bps=channels*rate;
+ if(format != AFMT_U8 && format != AFMT_S8)
+ ao_data.bps*=2;
+ if(ao_data.buffersize==-1)
+ {
+ ao_data.buffersize=audio_out_format_bits(format)/8;
+ ao_data.buffersize*= channels;
+ ao_data.buffersize*= SAMPLESIZE;
+ }
+ mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
+
+ //fill waveformatex
+ ZeroMemory( &wformat, sizeof(WAVEFORMATEX));
+ wformat.cbSize = 0; /* size of _extra_ info */
+ wformat.wFormatTag = WAVE_FORMAT_PCM;
+ wformat.nChannels = channels;
+ wformat.nSamplesPerSec = rate;
+ wformat.wBitsPerSample = audio_out_format_bits(format);
+ wformat.nBlockAlign = wformat.nChannels * (wformat.wBitsPerSample >> 3);
+ wformat.nAvgBytesPerSec = wformat.nSamplesPerSec * wformat.nBlockAlign;
+
+ //open sound device
+ //WAVE_MAPPER always points to the default wave device on the system
+ result = waveOutOpen(&hWaveOut,WAVE_MAPPER,&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
+ if(result == WAVERR_BADFORMAT)
+ {
+ mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
+ ao_data.channels = wformat.nChannels = 2;
+ ao_data.samplerate = wformat.nSamplesPerSec = 44100;
+ ao_data.format = AFMT_S16_LE;
+ ao_data.bps=ao_data.channels * ao_data.samplerate;
+ ao_data.buffersize=wformat.wBitsPerSample=16;
+ wformat.nBlockAlign = wformat.nChannels * (wformat.wBitsPerSample >> 3);
+ wformat.nAvgBytesPerSec = wformat.nSamplesPerSec * wformat.nBlockAlign;
+ ao_data.buffersize/=8;
+ ao_data.buffersize*= ao_data.channels;
+ ao_data.buffersize*= SAMPLESIZE;
+ result = waveOutOpen(&hWaveOut,WAVE_MAPPER,&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
+ }
+ if(result != MMSYSERR_NOERROR)
+ {
+ mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device\n");
+ return 0;
+ }
+ //save volume
+ waveOutGetVolume(hWaveOut,&restoredvolume);
+ //allocate buffer memory as one big block
+ buffer = malloc(totalBufferSize);
+ //and setup pointers to each buffer
+ waveBlocks = (WAVEHDR*)buffer;
+ buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
+ for(i = 0; i < BUFFER_COUNT; i++) {
+ waveBlocks[i].dwBufferLength = BUFFER_SIZE;
+ waveBlocks[i].lpData = buffer;
+ buffer += BUFFER_SIZE;
+ }
+
+ return 1;
+}
+
+// close audio device
+static void uninit()
+{
+ waveOutSetVolume(hWaveOut,restoredvolume); //restore volume
+ waveOutReset(hWaveOut);
+ waveOutClose(hWaveOut);
+ mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
+ full_buffers=0;
+ free(waveBlocks);
+ mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
+}
+
+// stop playing and empty buffers (for seeking/pause)
+static void reset()
+{
+ waveOutReset(hWaveOut);
+ buf_write=0;
+ buf_write_pos=0;
+ full_buffers=0;
+ buffered_bytes=0;
+}
+
+// stop playing, keep buffers (for pause)
+static void audio_pause()
+{
+ waveOutPause(hWaveOut);
+}
+
+// resume playing, after audio_pause()
+static void audio_resume()
+{
+ waveOutRestart(hWaveOut);
+}
+
+// return: how many bytes can be played without blocking
+static int get_space()
+{
+ return (BUFFER_COUNT-full_buffers)*BUFFER_SIZE - buf_write_pos;
+}
+
+//writes data into buffer, based on ringbuffer code in ao_sdl.c
+static int write_waveOutBuffer(unsigned char* data,int len){
+ WAVEHDR* current;
+ int len2=0;
+ int x;
+ while(len>0){
+ current = &waveBlocks[buf_write];
+ if(full_buffers==BUFFER_COUNT) break;
+ //unprepare the header if it is prepared
+ if(current->dwFlags & WHDR_PREPARED)
+ waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
+ x=BUFFER_SIZE-buf_write_pos;
+ if(x>len) x=len;
+ memcpy(current->lpData+buf_write_pos,data+len2,x);
+ len2+=x; len-=x;
+ buffered_bytes+=x; buf_write_pos+=x;
+ //prepare header and write data to device
+ 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;
+ ++full_buffers;
+ buf_write_pos=0;
+ }
+ }
+ return len2;
+}
+
+// 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)
+{
+ return write_waveOutBuffer(data,len);
+}
+int previous = 0;
+// return: delay in seconds between first and last sample in buffer
+static float get_delay()
+{
+ return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
+}