Aegisub/aegisub/audio_player_dsound.cpp
Amar Takhar 0763541848 Change how configuration works.
* move win32/config0.h -> config/config_windows0.h
  * move win32/stdint.h -> msvc/stdint.h
  * move posix/defines.h -> config/config_unix.h
  * add config.h - brings in config_(windows|unix).h as required
  * add config.h to *.cpp
  * Self-contain FFmpegSource2 with required function defines wrapped with
    __UNIX__ instead of the now removed defines.h.
  * Edit aegisub_vs2008.vcproj to remove explicit inclusion of config.h, also
    change VCPreBuildEventTool to create windows_config.h from
    windows_config0.h.  Add msvc to include path.
  * Change configure to create ./acconf.h instead of posix/acconf.h

This will allow us to create a more standard and platform agnostic way of
configuration aegsisub during build time.

Originally committed to SVN as r2621.
2009-01-04 06:31:48 +00:00

405 lines
11 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
//
///////////
// Headers
#include "config.h"
#ifdef WITH_DIRECTSOUND
#include <wx/wxprec.h>
#include "utils.h"
#include "main.h"
#include "frame_main.h"
#include "audio_player_dsound.h"
///////////////
// 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 = sizeof(waveFormat);
// 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;
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_t start,int64_t 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*provider->GetBytesPerSample() > 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) {
if (thread->IsAlive()) {
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_t pos) {
if (playing) endPos = pos;
}
////////////////////////
// Set current position
void DirectSoundPlayer::SetCurrentPosition(int64_t pos) {
startPos = playPos = pos;
startTime = GetTickCount();
}
////////////////////////
// Get current position
int64_t 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_t 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() {
CoInitialize(0);
// 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();
CoUninitialize();
return 0;
}
////////////////////////
// Stop playback thread
void DirectSoundPlayerThread::Stop() {
// Increase the stopnotify by one, causing a wait for it to succeed
SetEvent(stopnotify);
}
#endif // WITH_DIRECTSOUND