mirror of https://github.com/odrling/Aegisub
The buffer length/wanted latency in the dsound2 audio player are now tweakable in config.dat; the options are called "audio dsound buffer latency" and "audio dsound buffer length" respectively. Defaults to 100/5 respectively (should work for the people who used to get buffer errors); for better responsiveness try 80/4.
Originally committed to SVN as r2905.
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@ -48,6 +48,7 @@
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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 "options.h"
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#include "audio_player_dsound2.h"
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@ -131,12 +132,15 @@ class DirectSoundPlayer2Thread {
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int64_t start_frame;
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int64_t end_frame;
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int wanted_latency;
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int buffer_length;
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DWORD last_playback_restart;
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AudioProvider *provider;
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public:
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DirectSoundPlayer2Thread(AudioProvider *provider);
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DirectSoundPlayer2Thread(AudioProvider *provider, int WantedLatency, int BufferLength);
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~DirectSoundPlayer2Thread();
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void Play(int64_t start, int64_t count);
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@ -159,11 +163,6 @@ unsigned int __stdcall DirectSoundPlayer2Thread::ThreadProc(void *parameter)
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}
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#define WANTED_LATENCY 80
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#define BUFFER_LENGTH 4
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// The buffer will hold BUFFER_LENGTH times WANTED_LATENCY milliseconds of audio
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void DirectSoundPlayer2Thread::Run()
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{
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#define REPORT_ERROR(msg) { error_message = _T("DirectSoundPlayer2Thread: ") _T(msg); SetEvent(error_happened); return; }
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@ -194,7 +193,7 @@ void DirectSoundPlayer2Thread::Run()
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waveFormat.cbSize = sizeof(waveFormat);
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// And the buffer itself
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int aim = waveFormat.nAvgBytesPerSec * (WANTED_LATENCY*BUFFER_LENGTH)/1000;
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int aim = waveFormat.nAvgBytesPerSec * (wanted_latency*buffer_length)/1000;
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int min = DSBSIZE_MIN;
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int max = DSBSIZE_MAX;
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DWORD bufSize = MIN(MAX(min,aim),max); // size of entier playback buffer
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@ -239,7 +238,7 @@ void DirectSoundPlayer2Thread::Run()
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while (running)
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{
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DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait)/sizeof(HANDLE), events_to_wait, FALSE, WANTED_LATENCY);
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DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait)/sizeof(HANDLE), events_to_wait, FALSE, wanted_latency);
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switch (wait_result)
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{
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@ -475,7 +474,7 @@ void DirectSoundPlayer2Thread::CheckError()
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}
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DirectSoundPlayer2Thread::DirectSoundPlayer2Thread(AudioProvider *provider)
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DirectSoundPlayer2Thread::DirectSoundPlayer2Thread(AudioProvider *provider, int _WantedLatency, int _BufferLength)
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{
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event_start_playback = CreateEvent(0, FALSE, FALSE, 0);
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event_stop_playback = CreateEvent(0, FALSE, FALSE, 0);
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@ -492,6 +491,9 @@ DirectSoundPlayer2Thread::DirectSoundPlayer2Thread(AudioProvider *provider)
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start_frame = 0;
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end_frame = 0;
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wanted_latency = _WantedLatency;
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buffer_length = _BufferLength;
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this->provider = provider;
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thread_handle = (HANDLE)_beginthreadex(0, 0, ThreadProc, this, 0, 0);
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@ -613,6 +615,16 @@ double DirectSoundPlayer2Thread::GetVolume()
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DirectSoundPlayer2::DirectSoundPlayer2()
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{
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thread = 0;
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// The buffer will hold BufferLength times WantedLatency milliseconds of audio
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WantedLatency = Options.AsInt(_T("Audio dsound buffer latency"));
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BufferLength = Options.AsInt(_T("Audio dsound buffer length"));
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// sanity checking
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if (WantedLatency <= 0)
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WantedLatency = 100;
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if (BufferLength <= 0)
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BufferLength = 5;
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}
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@ -628,7 +640,7 @@ void DirectSoundPlayer2::OpenStream()
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try
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{
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thread = new DirectSoundPlayer2Thread(GetProvider());
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thread = new DirectSoundPlayer2Thread(GetProvider(), WantedLatency, BufferLength);
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}
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catch (const wxChar *msg)
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{
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@ -661,7 +673,7 @@ void DirectSoundPlayer2::SetProvider(AudioProvider *provider)
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if (thread && provider != GetProvider())
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{
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delete thread;
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thread = new DirectSoundPlayer2Thread(provider);
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thread = new DirectSoundPlayer2Thread(provider, WantedLatency, BufferLength);
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}
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AudioPlayer::SetProvider(provider);
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@ -45,6 +45,10 @@ class DirectSoundPlayer2Thread;
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class DirectSoundPlayer2 : public AudioPlayer {
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DirectSoundPlayer2Thread *thread;
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protected:
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int WantedLatency;
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int BufferLength;
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public:
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DirectSoundPlayer2();
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~DirectSoundPlayer2();
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@ -219,6 +219,9 @@ void OptionsManager::LoadDefaults(bool onlyDefaults,bool doOverride) {
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SetInt(_T("Audio Spectrum Cutoff"),0);
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SetInt(_T("Audio Spectrum Quality"),1);
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SetInt(_T("Audio Spectrum Memory Max"),128); // megabytes
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// and this one could do with just reiniting the audio player
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SetInt(_T("Audio dsound buffer latency"), 100);
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SetInt(_T("Audio dsound buffer length"), 5);
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// Automation
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// The path changes only take effect when a script is (re)loaded but Automatic should be good enough, it certainly doesn't warrart a restart
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