Aegisub/aegisub/audio_player_dsound.cpp
Niels Martin Hansen 0718cf5a2c Hack: Use a single-play (non-looping) buffer for very short samples that fit entirely in the buffer for DSound playback. This avoids unwanted repeating of very short samples, but also makes it impossible to extend these selections to continue playback.
Fixing the real problem could prove to be very hard or even impossible, as it seems to be a problem of timing and possibly even related to the time slices allocated by the OS. As a buffer size in the DSound player is fixed at 150 ms the side-effect of this hack hopefully won't be a real problem.

Originally committed to SVN as r1460.
2007-07-29 21:00:57 +00:00

483 lines
12 KiB
C++

// Copyright (c) 2006, Rodrigo Braz Monteiro
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// -----------------------------------------------------------------------------
//
// AEGISUB
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:zeratul@cellosoft.com
//
#pragma once
///////////
// Headers
#include <wx/wxprec.h>
#include "audio_player.h"
#include "audio_provider.h"
#include "utils.h"
#include "main.h"
#include "frame_main.h"
#include "audio_player.h"
#include <mmsystem.h>
#include <dsound.h>
/////////////
// Libraries
#pragma comment(lib, "dsound.lib")
#pragma comment(lib, "dxguid.lib")
//////////////
// Prototypes
class DirectSoundPlayer;
//////////
// Thread
class DirectSoundPlayerThread : public wxThread {
private:
DirectSoundPlayer *parent;
HANDLE stopnotify;
public:
void Stop(); // Notify thread to stop audio playback. Thread safe.
DirectSoundPlayerThread(DirectSoundPlayer *parent);
~DirectSoundPlayerThread();
wxThread::ExitCode Entry();
};
////////////////////
// Portaudio player
class DirectSoundPlayer : public AudioPlayer {
friend class DirectSoundPlayerThread;
private:
volatile bool playing;
float volume;
int offset;
DWORD bufSize;
volatile __int64 playPos;
__int64 startPos;
volatile __int64 endPos;
DWORD startTime;
IDirectSound8 *directSound;
IDirectSoundBuffer8 *buffer;
bool FillBuffer(bool fill);
DirectSoundPlayerThread *thread;
public:
DirectSoundPlayer();
~DirectSoundPlayer();
void OpenStream();
void CloseStream();
void Play(__int64 start,__int64 count);
void Stop(bool timerToo=true);
bool IsPlaying() { return playing; }
__int64 GetStartPosition() { return startPos; }
__int64 GetEndPosition() { return endPos; }
__int64 GetCurrentPosition();
void SetEndPosition(__int64 pos);
void SetCurrentPosition(__int64 pos);
void SetVolume(double vol) { volume = vol; }
double GetVolume() { return volume; }
//wxMutex *GetMutex() { return &DSMutex; }
};
///////////
// Factory
class DirectSoundPlayerFactory : public AudioPlayerFactory {
public:
AudioPlayer *CreatePlayer() { return new DirectSoundPlayer(); }
DirectSoundPlayerFactory() : AudioPlayerFactory(_T("dsound")) {}
} registerDirectSoundPlayer;
///////////////
// Constructor
DirectSoundPlayer::DirectSoundPlayer() {
playing = false;
volume = 1.0f;
playPos = 0;
startPos = 0;
endPos = 0;
offset = 0;
buffer = NULL;
directSound = NULL;
thread = NULL;
}
//////////////
// Destructor
DirectSoundPlayer::~DirectSoundPlayer() {
CloseStream();
}
///////////////
// Open stream
void DirectSoundPlayer::OpenStream() {
// Get provider
AudioProvider *provider = GetProvider();
// Initialize the DirectSound object
HRESULT res;
res = DirectSoundCreate8(&DSDEVID_DefaultPlayback,&directSound,NULL); // TODO: support selecting audio device
if (FAILED(res)) throw _T("Failed initializing DirectSound");
// Set DirectSound parameters
AegisubApp *app = (AegisubApp*) wxTheApp;
directSound->SetCooperativeLevel((HWND)app->frame->GetHandle(),DSSCL_PRIORITY);
// Create the wave format structure
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
// Create the buffer initializer
int aim = waveFormat.nAvgBytesPerSec * 15/100; // 150 ms buffer
int min = DSBSIZE_MIN;
int max = DSBSIZE_MAX;
bufSize = MIN(MAX(min,aim),max);
DSBUFFERDESC desc;
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLPOSITIONNOTIFY;
desc.dwBufferBytes = bufSize;
desc.dwReserved = 0;
desc.lpwfxFormat = &waveFormat;
desc.guid3DAlgorithm = GUID_NULL;
// Create the buffer
IDirectSoundBuffer *buf;
res = directSound->CreateSoundBuffer(&desc,&buf,NULL);
if (res != DS_OK) throw _T("Failed creating DirectSound buffer");
// Copy interface to buffer
res = buf->QueryInterface(IID_IDirectSoundBuffer8,(LPVOID*) &buffer);
if (res != S_OK) throw _T("Failed casting interface to IDirectSoundBuffer8");
// Set data
offset = 0;
}
////////////////
// Close stream
void DirectSoundPlayer::CloseStream() {
// Stop it
Stop();
// Unref the DirectSound buffer
if (buffer) {
buffer->Release();
buffer = NULL;
}
// Unref the DirectSound object
if (directSound) {
directSound->Release();
directSound = NULL;
}
}
///////////////
// Fill buffer
bool DirectSoundPlayer::FillBuffer(bool fill) {
if (playPos >= endPos) return false;
// Variables
HRESULT res;
void *ptr1, *ptr2;
unsigned long int size1, size2;
AudioProvider *provider = GetProvider();
int bytesps = provider->GetBytesPerSample();
// To write length
int toWrite = 0;
if (fill) {
toWrite = bufSize;
}
else {
DWORD bufplay;
res = buffer->GetCurrentPosition(&bufplay, NULL);
if (FAILED(res)) return false;
toWrite = (int)bufplay - (int)offset;
if (toWrite < 0) toWrite += bufSize;
}
if (toWrite == 0) return true;
// Make sure we only get as many samples as are available
if (playPos + toWrite/bytesps > endPos) {
toWrite = (endPos - playPos) * bytesps;
}
// If we're going to fill the entire buffer (ie. at start of playback) start by zeroing it out
// If it's not zeroed out we might have a playback selection shorter than the buffer
// and then everything after the playback selection will be junk, which we don't want played.
if (fill) {
RetryClear:
res = buffer->Lock(0, bufSize, &ptr1, &size1, &ptr2, &size2, 0);
if (res == DSERR_BUFFERLOST) {
buffer->Restore();
goto RetryClear;
}
memset(ptr1, 0, size1);
memset(ptr2, 0, size2);
buffer->Unlock(ptr1, size1, ptr2, size2);
}
// Lock buffer
RetryLock:
if (fill) {
res = buffer->Lock(offset, toWrite, &ptr1, &size1, &ptr2, &size2, 0);
}
else {
res = buffer->Lock(offset, toWrite, &ptr1, &size1, &ptr2, &size2, 0);//DSBLOCK_FROMWRITECURSOR);
}
// Buffer lost?
if (res == DSERR_BUFFERLOST) {
wxLogDebug(_T("Lost DSound buffer"));
buffer->Restore();
goto RetryLock;
}
// Error
if (FAILED(res)) return false;
// Convert size to number of samples
unsigned long int count1 = size1 / bytesps;
unsigned long int count2 = size2 / bytesps;
if (count1) wxLogDebug(_T("DS fill: %05lu -> %05lu"), (unsigned long)playPos, (unsigned long)playPos+count1);
if (count2) wxLogDebug(_T("DS fill: %05lu => %05lu"), (unsigned long)playPos+count1, (unsigned long)playPos+count1+count2);
if (!count1 && !count2) wxLogDebug(_T("DS fill: nothing"));
// Get source wave
if (count1) provider->GetAudioWithVolume(ptr1, playPos, count1, volume);
if (count2) provider->GetAudioWithVolume(ptr2, playPos+count1, count2, volume);
playPos += count1+count2;
// Unlock
buffer->Unlock(ptr1,count1*bytesps,ptr2,count2*bytesps);
// Update offset
offset = (offset + count1*bytesps + count2*bytesps) % bufSize;
return playPos < endPos;
}
////////
// Play
void DirectSoundPlayer::Play(__int64 start,__int64 count) {
// Make sure that it's stopped
Stop();
// The thread is now guaranteed dead
HRESULT res;
// We sure better have a buffer
assert(buffer);
// Set variables
startPos = start;
endPos = start+count;
playPos = start;
offset = 0;
// Fill whole buffer
FillBuffer(true);
DWORD play_flag = 0;
if (count > bufSize) {
// Start thread
thread = new DirectSoundPlayerThread(this);
thread->Create();
thread->Run();
play_flag = DSBPLAY_LOOPING;
}
// Play
buffer->SetCurrentPosition(0);
res = buffer->Play(0,0,play_flag);
if (SUCCEEDED(res)) playing = true;
startTime = GetTickCount();
// Update timer
if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15);
}
////////
// Stop
void DirectSoundPlayer::Stop(bool timerToo) {
// Stop the thread
if (thread) {
thread->Stop();
thread->Wait();
thread = NULL;
}
// The thread is now guaranteed dead and there are no concurrency problems to worry about
// Stop
if (buffer) buffer->Stop(); // the thread should have done this already
// Reset variables
playing = false;
playPos = 0;
startPos = 0;
endPos = 0;
offset = 0;
// Stop timer
if (timerToo && displayTimer) {
displayTimer->Stop();
}
}
///////////
// Set end
void DirectSoundPlayer::SetEndPosition(__int64 pos) {
if (playing) endPos = pos;
}
////////////////////////
// Set current position
void DirectSoundPlayer::SetCurrentPosition(__int64 pos) {
startPos = playPos = pos;
startTime = GetTickCount();
}
////////////////////////
// Get current position
__int64 DirectSoundPlayer::GetCurrentPosition() {
// Check if buffer is loaded
if (!buffer || !playing) return 0;
// FIXME: this should be based on not duration played but actual sample being heard
// (during vidoeo playback, cur_frame might get changed to resync)
DWORD curtime = GetTickCount();
__int64 tdiff = curtime - startTime;
return startPos + tdiff * provider->GetSampleRate() / 1000;
}
//////////////////////
// Thread constructor
DirectSoundPlayerThread::DirectSoundPlayerThread(DirectSoundPlayer *par) : wxThread(wxTHREAD_JOINABLE) {
parent = par;
stopnotify = CreateEvent(NULL, true, false, NULL);
}
/////////////////////
// Thread destructor
DirectSoundPlayerThread::~DirectSoundPlayerThread() {
CloseHandle(stopnotify);
}
//////////////////////
// Thread entry point
wxThread::ExitCode DirectSoundPlayerThread::Entry() {
// Wake up thread every half second to fill buffer as needed
// This more or less assumes the buffer is at least one second long
while (WaitForSingleObject(stopnotify, 50) == WAIT_TIMEOUT) {
if (!parent->FillBuffer(false)) {
// FillBuffer returns false when end of stream is reached
wxLogDebug(_T("DS thread hit end of stream"));
break;
}
}
// Now fill buffer with silence
DWORD bytesFilled = 0;
while (WaitForSingleObject(stopnotify, 50) == WAIT_TIMEOUT) {
void *buf1, *buf2;
DWORD size1, size2;
DWORD playpos;
HRESULT res;
res = parent->buffer->GetCurrentPosition(&playpos, NULL);
if (FAILED(res)) break;
int toWrite = playpos - parent->offset;
while (toWrite < 0) toWrite += parent->bufSize;
res = parent->buffer->Lock(parent->offset, toWrite, &buf1, &size1, &buf2, &size2, 0);
if (FAILED(res)) break;
if (size1) memset(buf1, 0, size1);
if (size2) memset(buf2, 0, size2);
if (size1) wxLogDebug(_T("DS blnk: %05ld -> %05ld"), (unsigned long)parent->playPos+bytesFilled, (unsigned long)parent->playPos+bytesFilled+size1);
if (size2) wxLogDebug(_T("DS blnk: %05ld => %05ld"), (unsigned long)parent->playPos+bytesFilled+size1, (unsigned long)parent->playPos+bytesFilled+size1+size2);
bytesFilled += size1 + size2;
parent->buffer->Unlock(buf1, size1, buf2, size2);
if (bytesFilled > parent->bufSize) break;
parent->offset = (parent->offset + size1 + size2) % parent->bufSize;
}
WaitForSingleObject(stopnotify, 150);
wxLogDebug(_T("DS thread dead"));
parent->playing = false;
parent->buffer->Stop();
return 0;
}
////////////////////////
// Stop playback thread
void DirectSoundPlayerThread::Stop() {
// Increase the stopnotify by one, causing a wait for it to succeed
SetEvent(stopnotify);
}