mirror of https://github.com/odrling/Aegisub
500 lines
13 KiB
C++
500 lines
13 KiB
C++
// Copyright (c) 2006, Rodrigo Braz Monteiro
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// All rights reserved.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// -----------------------------------------------------------------------------
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//
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// AEGISUB
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//
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// Website: http://aegisub.cellosoft.com
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// Contact: mailto:zeratul@cellosoft.com
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//
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///////////
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// Headers
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#ifdef WITH_DIRECTSOUND
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#include <wx/wxprec.h>
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#include "audio_player.h"
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#include "audio_provider.h"
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#include "utils.h"
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#include "main.h"
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#include "frame_main.h"
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#include "audio_player.h"
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#include <mmsystem.h>
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#include <dsound.h>
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//////////////
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// Prototypes
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class DirectSoundPlayer;
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//////////
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// Thread
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class DirectSoundPlayerThread : public wxThread {
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private:
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DirectSoundPlayer *parent;
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HANDLE stopnotify;
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public:
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void Stop(); // Notify thread to stop audio playback. Thread safe.
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DirectSoundPlayerThread(DirectSoundPlayer *parent);
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~DirectSoundPlayerThread();
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wxThread::ExitCode Entry();
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};
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/*
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TODO: Rewrite playback thread to manage all of the buffer, and properly marshal the IDirectSound8
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object into the thread for creating the buffer there.
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The thread should own the buffer and manage all of the playback.
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It must be created with start and duration set, and begins playback at the given position.
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New functions:
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* Seek(pos) : Restart playback from the given position
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* SetEnd(pos) : Set new end point
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* GetPosition() : Get the current sample number being played
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* Stop() : Stop playback immediately
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Instead of using a stop event, use a playback parameters changed event. When that one's fired,
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detect which were actually changed and act accordingly.
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All but GetPosition() set appropriate fields and then raise the parameters changed event.
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*/
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////////////////////
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// Portaudio player
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class DirectSoundPlayer : public AudioPlayer {
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friend class DirectSoundPlayerThread;
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private:
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volatile bool playing;
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float volume;
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int offset;
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DWORD bufSize;
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volatile int64_t playPos;
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int64_t startPos;
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volatile int64_t endPos;
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DWORD startTime;
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IDirectSound8 *directSound;
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IDirectSoundBuffer8 *buffer;
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bool FillBuffer(bool fill);
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DirectSoundPlayerThread *thread;
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public:
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DirectSoundPlayer();
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~DirectSoundPlayer();
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void OpenStream();
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void CloseStream();
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void Play(int64_t start,int64_t count);
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void Stop(bool timerToo=true);
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bool IsPlaying() { return playing; }
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int64_t GetStartPosition() { return startPos; }
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int64_t GetEndPosition() { return endPos; }
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int64_t GetCurrentPosition();
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void SetEndPosition(int64_t pos);
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void SetCurrentPosition(int64_t pos);
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void SetVolume(double vol) { volume = vol; }
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double GetVolume() { return volume; }
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//wxMutex *GetMutex() { return &DSMutex; }
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};
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///////////
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// Factory
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class DirectSoundPlayerFactory : public AudioPlayerFactory {
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public:
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AudioPlayer *CreatePlayer() { return new DirectSoundPlayer(); }
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DirectSoundPlayerFactory() : AudioPlayerFactory(_T("dsound")) {}
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} registerDirectSoundPlayer;
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///////////////
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// Constructor
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DirectSoundPlayer::DirectSoundPlayer() {
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playing = false;
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volume = 1.0f;
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playPos = 0;
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startPos = 0;
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endPos = 0;
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offset = 0;
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buffer = NULL;
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directSound = NULL;
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thread = NULL;
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}
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//////////////
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// Destructor
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DirectSoundPlayer::~DirectSoundPlayer() {
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CloseStream();
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}
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///////////////
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// Open stream
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void DirectSoundPlayer::OpenStream() {
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// Get provider
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AudioProvider *provider = GetProvider();
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// Initialize the DirectSound object
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HRESULT res;
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res = DirectSoundCreate8(&DSDEVID_DefaultPlayback,&directSound,NULL); // TODO: support selecting audio device
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if (FAILED(res)) throw _T("Failed initializing DirectSound");
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// Set DirectSound parameters
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AegisubApp *app = (AegisubApp*) wxTheApp;
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directSound->SetCooperativeLevel((HWND)app->frame->GetHandle(),DSSCL_PRIORITY);
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// Create the wave format structure
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WAVEFORMATEX waveFormat;
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waveFormat.wFormatTag = WAVE_FORMAT_PCM;
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waveFormat.nSamplesPerSec = provider->GetSampleRate();
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waveFormat.nChannels = provider->GetChannels();
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waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
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waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
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waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
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waveFormat.cbSize = sizeof(waveFormat);
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// Create the buffer initializer
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int aim = waveFormat.nAvgBytesPerSec * 15/100; // 150 ms buffer
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int min = DSBSIZE_MIN;
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int max = DSBSIZE_MAX;
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bufSize = MIN(MAX(min,aim),max);
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DSBUFFERDESC desc;
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desc.dwSize = sizeof(DSBUFFERDESC);
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desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
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desc.dwBufferBytes = bufSize;
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desc.dwReserved = 0;
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desc.lpwfxFormat = &waveFormat;
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desc.guid3DAlgorithm = GUID_NULL;
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// Create the buffer
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IDirectSoundBuffer *buf;
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res = directSound->CreateSoundBuffer(&desc,&buf,NULL);
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if (res != DS_OK) throw _T("Failed creating DirectSound buffer");
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// Copy interface to buffer
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res = buf->QueryInterface(IID_IDirectSoundBuffer8,(LPVOID*) &buffer);
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if (res != S_OK) throw _T("Failed casting interface to IDirectSoundBuffer8");
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// Set data
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offset = 0;
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}
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////////////////
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// Close stream
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void DirectSoundPlayer::CloseStream() {
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// Stop it
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Stop();
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// Unref the DirectSound buffer
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if (buffer) {
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buffer->Release();
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buffer = NULL;
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}
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// Unref the DirectSound object
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if (directSound) {
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directSound->Release();
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directSound = NULL;
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}
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}
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///////////////
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// Fill buffer
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bool DirectSoundPlayer::FillBuffer(bool fill) {
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if (playPos >= endPos) return false;
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// Variables
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HRESULT res;
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void *ptr1, *ptr2;
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unsigned long int size1, size2;
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AudioProvider *provider = GetProvider();
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int bytesps = provider->GetBytesPerSample();
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// To write length
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int toWrite = 0;
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if (fill) {
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toWrite = bufSize;
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}
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else {
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DWORD bufplay;
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res = buffer->GetCurrentPosition(&bufplay, NULL);
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if (FAILED(res)) return false;
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toWrite = (int)bufplay - (int)offset;
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if (toWrite < 0) toWrite += bufSize;
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}
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if (toWrite == 0) return true;
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// Make sure we only get as many samples as are available
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if (playPos + toWrite/bytesps > endPos) {
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toWrite = (endPos - playPos) * bytesps;
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}
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// If we're going to fill the entire buffer (ie. at start of playback) start by zeroing it out
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// If it's not zeroed out we might have a playback selection shorter than the buffer
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// and then everything after the playback selection will be junk, which we don't want played.
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if (fill) {
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RetryClear:
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res = buffer->Lock(0, bufSize, &ptr1, &size1, &ptr2, &size2, 0);
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if (res == DSERR_BUFFERLOST) {
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buffer->Restore();
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goto RetryClear;
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}
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memset(ptr1, 0, size1);
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memset(ptr2, 0, size2);
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buffer->Unlock(ptr1, size1, ptr2, size2);
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}
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// Lock buffer
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RetryLock:
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if (fill) {
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res = buffer->Lock(offset, toWrite, &ptr1, &size1, &ptr2, &size2, 0);
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}
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else {
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res = buffer->Lock(offset, toWrite, &ptr1, &size1, &ptr2, &size2, 0);//DSBLOCK_FROMWRITECURSOR);
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}
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// Buffer lost?
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if (res == DSERR_BUFFERLOST) {
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wxLogDebug(_T("Lost DSound buffer"));
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buffer->Restore();
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goto RetryLock;
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}
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// Error
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if (FAILED(res)) return false;
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// Convert size to number of samples
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unsigned long int count1 = size1 / bytesps;
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unsigned long int count2 = size2 / bytesps;
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if (count1) wxLogDebug(_T("DS fill: %05lu -> %05lu"), (unsigned long)playPos, (unsigned long)playPos+count1);
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if (count2) wxLogDebug(_T("DS fill: %05lu => %05lu"), (unsigned long)playPos+count1, (unsigned long)playPos+count1+count2);
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if (!count1 && !count2) wxLogDebug(_T("DS fill: nothing"));
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// Get source wave
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if (count1) provider->GetAudioWithVolume(ptr1, playPos, count1, volume);
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if (count2) provider->GetAudioWithVolume(ptr2, playPos+count1, count2, volume);
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playPos += count1+count2;
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// Unlock
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buffer->Unlock(ptr1,count1*bytesps,ptr2,count2*bytesps);
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// Update offset
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offset = (offset + count1*bytesps + count2*bytesps) % bufSize;
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return playPos < endPos;
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}
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////////
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// Play
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void DirectSoundPlayer::Play(int64_t start,int64_t count) {
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// Make sure that it's stopped
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Stop();
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// The thread is now guaranteed dead
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HRESULT res;
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// We sure better have a buffer
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assert(buffer);
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// Set variables
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startPos = start;
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endPos = start+count;
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playPos = start;
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offset = 0;
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// Fill whole buffer
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FillBuffer(true);
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DWORD play_flag = 0;
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if (count*provider->GetBytesPerSample() > bufSize) {
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// Start thread
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thread = new DirectSoundPlayerThread(this);
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thread->Create();
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thread->Run();
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play_flag = DSBPLAY_LOOPING;
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}
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// Play
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buffer->SetCurrentPosition(0);
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res = buffer->Play(0,0,play_flag);
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if (SUCCEEDED(res)) playing = true;
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startTime = GetTickCount();
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// Update timer
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if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15);
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}
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////////
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// Stop
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void DirectSoundPlayer::Stop(bool timerToo) {
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// Stop the thread
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if (thread) {
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if (thread->IsAlive()) {
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thread->Stop();
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thread->Wait();
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}
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thread = NULL;
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}
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// The thread is now guaranteed dead and there are no concurrency problems to worry about
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// Stop
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if (buffer) buffer->Stop(); // the thread should have done this already
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// Reset variables
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playing = false;
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playPos = 0;
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startPos = 0;
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endPos = 0;
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offset = 0;
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// Stop timer
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if (timerToo && displayTimer) {
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displayTimer->Stop();
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}
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}
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///////////
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// Set end
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void DirectSoundPlayer::SetEndPosition(int64_t pos) {
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if (playing) endPos = pos;
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}
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////////////////////////
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// Set current position
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void DirectSoundPlayer::SetCurrentPosition(int64_t pos) {
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startPos = playPos = pos;
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startTime = GetTickCount();
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}
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////////////////////////
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// Get current position
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int64_t DirectSoundPlayer::GetCurrentPosition() {
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// Check if buffer is loaded
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if (!buffer || !playing) return 0;
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// FIXME: this should be based on not duration played but actual sample being heard
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// (during vidoeo playback, cur_frame might get changed to resync)
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DWORD curtime = GetTickCount();
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int64_t tdiff = curtime - startTime;
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return startPos + tdiff * provider->GetSampleRate() / 1000;
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}
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//////////////////////
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// Thread constructor
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DirectSoundPlayerThread::DirectSoundPlayerThread(DirectSoundPlayer *par) : wxThread(wxTHREAD_JOINABLE) {
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parent = par;
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stopnotify = CreateEvent(NULL, true, false, NULL);
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}
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/////////////////////
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// Thread destructor
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DirectSoundPlayerThread::~DirectSoundPlayerThread() {
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CloseHandle(stopnotify);
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}
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//////////////////////
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// Thread entry point
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wxThread::ExitCode DirectSoundPlayerThread::Entry() {
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CoInitialize(0);
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// Wake up thread every half second to fill buffer as needed
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// This more or less assumes the buffer is at least one second long
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while (WaitForSingleObject(stopnotify, 50) == WAIT_TIMEOUT) {
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if (!parent->FillBuffer(false)) {
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// FillBuffer returns false when end of stream is reached
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wxLogDebug(_T("DS thread hit end of stream"));
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break;
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}
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}
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// Now fill buffer with silence
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DWORD bytesFilled = 0;
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while (WaitForSingleObject(stopnotify, 50) == WAIT_TIMEOUT) {
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void *buf1, *buf2;
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DWORD size1, size2;
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DWORD playpos;
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HRESULT res;
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res = parent->buffer->GetCurrentPosition(&playpos, NULL);
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if (FAILED(res)) break;
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int toWrite = playpos - parent->offset;
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while (toWrite < 0) toWrite += parent->bufSize;
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res = parent->buffer->Lock(parent->offset, toWrite, &buf1, &size1, &buf2, &size2, 0);
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if (FAILED(res)) break;
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if (size1) memset(buf1, 0, size1);
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if (size2) memset(buf2, 0, size2);
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if (size1) wxLogDebug(_T("DS blnk: %05ld -> %05ld"), (unsigned long)parent->playPos+bytesFilled, (unsigned long)parent->playPos+bytesFilled+size1);
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if (size2) wxLogDebug(_T("DS blnk: %05ld => %05ld"), (unsigned long)parent->playPos+bytesFilled+size1, (unsigned long)parent->playPos+bytesFilled+size1+size2);
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bytesFilled += size1 + size2;
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parent->buffer->Unlock(buf1, size1, buf2, size2);
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if (bytesFilled > parent->bufSize) break;
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parent->offset = (parent->offset + size1 + size2) % parent->bufSize;
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}
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WaitForSingleObject(stopnotify, 150);
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wxLogDebug(_T("DS thread dead"));
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parent->playing = false;
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parent->buffer->Stop();
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CoUninitialize();
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return 0;
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}
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////////////////////////
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// Stop playback thread
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void DirectSoundPlayerThread::Stop() {
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// Increase the stopnotify by one, causing a wait for it to succeed
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SetEvent(stopnotify);
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}
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#endif // WITH_DIRECTSOUND
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