Aegisub/aegisub/src/audio_player_dsound2.cpp

970 lines
25 KiB
C++

// Copyright (c) 2008, 2010, Niels Martin Hansen
// 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 Project http://www.aegisub.org/
/// @file audio_player_dsound2.cpp
/// @brief New DirectSound-based audio output
/// @ingroup audio_output
///
#include "config.h"
#ifdef WITH_DIRECTSOUND
#ifndef AGI_PRE
#include <mmsystem.h>
#endif
#include <process.h>
#include <dsound.h>
#include <libaegisub/log.h>
#include "audio_player_dsound2.h"
#include "audio_controller.h"
#include "include/aegisub/audio_provider.h"
#include "frame_main.h"
#include "main.h"
#include "utils.h"
/// @brief RAII support class to init and de-init the COM library
struct COMInitialization {
/// Flag set if an inited COM library is managed
bool inited;
/// @brief Constructor, sets inited false
COMInitialization()
{
inited = false;
}
/// @brief Destructor, de-inits COM if it is inited
~COMInitialization()
{
if (inited) CoUninitialize();
}
/// @brief Initialise the COM library as single-threaded apartment if isn't already inited by us
void Init()
{
if (!inited)
{
if (FAILED(CoInitialize(NULL)))
throw std::exception();
inited = true;
}
}
};
/// @class COMObjectRetainer
/// @brief Simple auto_ptr-like class for COM objects
template<class T>
struct COMObjectRetainer {
/// Managed object
T *obj;
/// @brief Constructor for null object
COMObjectRetainer()
{
obj = 0;
}
/// @brief Constructor to take object immediately
/// @param _obj Object to manage
COMObjectRetainer(T *_obj)
{
obj = _obj;
}
/// @brief Destructor, releases object if there is one
~COMObjectRetainer()
{
if (obj) obj->Release();
}
/// @brief Dereference the managed object
/// @return The managed object
T * operator -> ()
{
return obj;
}
};
/// @brief RAII wrapper around Win32 HANDLE type
struct Win32KernelHandle : public agi::scoped_holder<HANDLE, BOOL (__stdcall *)(HANDLE)> {
/// @brief Create with a managed handle
/// @param handle Win32 handle to manage
Win32KernelHandle(HANDLE handle = 0)
: scoped_holder(handle, CloseHandle)
{
}
Win32KernelHandle& operator=(HANDLE new_handle)
{
scoped_holder::operator=(new_handle);
return *this;
}
};
/// @class DirectSoundPlayer2Thread
/// @brief Playback thread class for DirectSoundPlayer2
///
/// Not based on wxThread, but uses Win32 threads directly
class DirectSoundPlayer2Thread {
/// @brief Win32 thread entry point
/// @param parameter Pointer to our thread object
/// @return Thread return value, always 0 here
static unsigned int __stdcall ThreadProc(void *parameter);
/// @brief Thread entry point
void Run();
/// @brief Fill audio data into a locked buffer-pair and unlock the buffers
/// @param buf1 First buffer in pair
/// @param buf1sz Byte-size of first buffer in pair
/// @param buf2 Second buffer in pair, or null
/// @param buf2sz Byte-size of second buffer in pair
/// @param input_frame First audio frame to fill into buffers
/// @param bfr DirectSound buffer object owning the buffer pair
/// @return Number of bytes written
DWORD FillAndUnlockBuffers(void *buf1, DWORD buf1sz, void *buf2, DWORD buf2sz, int64_t &input_frame, IDirectSoundBuffer8 *bfr);
/// @brief Check for error state and throw exception if one occurred
void CheckError();
/// Win32 handle to the thread
Win32KernelHandle thread_handle;
/// Event object, world to thread, set to start playback
Win32KernelHandle event_start_playback;
/// Event object, world to thread, set to stop playback
Win32KernelHandle event_stop_playback;
/// Event object, world to thread, set if playback end time was updated
Win32KernelHandle event_update_end_time;
/// Event object, world to thread, set if the volume was changed
Win32KernelHandle event_set_volume;
/// Event object, world to thread, set if the thread should end as soon as possible
Win32KernelHandle event_kill_self;
/// Event object, thread to world, set when the thread has entered its main loop
Win32KernelHandle thread_running;
/// Event object, thread to world, set when playback is ongoing
Win32KernelHandle is_playing;
/// Event object, thread to world, set if an error state has occurred (implies thread is dying)
Win32KernelHandle error_happened;
/// Statically allocated error message text describing reason for error_happened being set
const char *error_message;
/// Playback volume, 1.0 is "unchanged"
double volume;
/// Audio frame to start playback at
int64_t start_frame;
/// Audio frame to end playback at
int64_t end_frame;
/// Desired length in milliseconds to write ahead of the playback cursor
int wanted_latency;
/// Multiplier for WantedLatency to get total buffer length
int buffer_length;
/// System millisecond timestamp of last playback start, used to calculate playback position
DWORD last_playback_restart;
/// Audio provider to take sample data from
AudioProvider *provider;
public:
/// @brief Constructor, creates and starts playback thread
/// @param provider Audio provider to take sample data from
/// @param WantedLatency Desired length in milliseconds to write ahead of the playback cursor
/// @param BufferLength Multiplier for WantedLatency to get total buffer length
DirectSoundPlayer2Thread(AudioProvider *provider, int WantedLatency, int BufferLength);
/// @brief Destructor, waits for thread to have died
~DirectSoundPlayer2Thread();
/// @brief Start audio playback
/// @param start Audio frame to start playback at
/// @param count Number of audio frames to play
void Play(int64_t start, int64_t count);
/// @brief Stop audio playback
void Stop();
/// @brief Change audio playback end point
/// @param new_end_frame New last audio frame to play
///
/// Playback stops instantly if new_end_frame is before the current playback position
void SetEndFrame(int64_t new_end_frame);
/// @brief Change audio playback volume
/// @param new_volume New playback amplification factor, 1.0 is "unchanged"
void SetVolume(double new_volume);
/// @brief Tell whether audio playback is active
/// @return True if audio is being played back, false if it is not
bool IsPlaying();
/// @brief Get first audio frame in current playback range
/// @return Audio frame index
int64_t GetStartFrame();
/// @brief Get approximate current audio frame being heard by the user
/// @return Audio frame index
///
/// Returns 0 if not playing
int64_t GetCurrentFrame();
/// @brief Get audio playback end point
/// @return Audio frame index
int64_t GetEndFrame();
/// @brief Get current playback volume
/// @return Audio amplification factor
double GetVolume();
/// @brief Tell whether playback thread has died
/// @return True if thread is no longer running
bool IsDead();
};
unsigned int __stdcall DirectSoundPlayer2Thread::ThreadProc(void *parameter)
{
static_cast<DirectSoundPlayer2Thread*>(parameter)->Run();
return 0;
}
/// Macro used to set error_message, error_happened and end the thread
#define REPORT_ERROR(msg) \
{ \
ResetEvent(is_playing); \
error_message = "DirectSoundPlayer2Thread: " msg; \
SetEvent(error_happened); \
return; \
}
void DirectSoundPlayer2Thread::Run()
{
COMInitialization COM_library;
try { COM_library.Init(); }
catch (std::exception e)
REPORT_ERROR("Could not initialise COM")
// Create DirectSound object
COMObjectRetainer<IDirectSound8> ds;
if (FAILED(DirectSoundCreate8(&DSDEVID_DefaultPlayback, &ds.obj, NULL)))
REPORT_ERROR("Cound not create DirectSound object")
// Ensure we can get interesting wave formats (unless we have PRIORITY we can only use a standard 8 bit format)
ds->SetCooperativeLevel((HWND)static_cast<AegisubApp*>(wxApp::GetInstance())->frame->GetHandle(), DSSCL_PRIORITY);
// Describe the wave format
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);
// And the buffer itself
int aim = waveFormat.nAvgBytesPerSec * (wanted_latency*buffer_length)/1000;
int min = DSBSIZE_MIN;
int max = DSBSIZE_MAX;
DWORD bufSize = mid(min,aim,max); // size of entire playback buffer
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;
// And then create the buffer
IDirectSoundBuffer *bfr7 = 0;
if FAILED(ds->CreateSoundBuffer(&desc, &bfr7, 0))
REPORT_ERROR("Could not create buffer")
// But it's an old version interface we get, query it for the DSound8 interface
COMObjectRetainer<IDirectSoundBuffer8> bfr;
if (FAILED(bfr7->QueryInterface(IID_IDirectSoundBuffer8, (LPVOID*)&bfr.obj)))
REPORT_ERROR("Buffer doesn't support version 8 interface")
bfr7->Release();
bfr7 = 0;
//wx Log Debug("DirectSoundPlayer2: Created buffer of %d bytes, supposed to be %d milliseconds or %d frames", bufSize, WANTED_LATENCY*BUFFER_LENGTH, bufSize/provider->GetBytesPerSample());
// Now we're ready to roll!
SetEvent(thread_running);
bool running = true;
HANDLE events_to_wait[] = {
event_start_playback,
event_stop_playback,
event_update_end_time,
event_set_volume,
event_kill_self
};
int64_t next_input_frame = 0;
DWORD buffer_offset = 0;
bool playback_should_be_running = false;
int current_latency = wanted_latency;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*provider->GetBytesPerSample()/1000;
while (running)
{
DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait)/sizeof(HANDLE), events_to_wait, FALSE, current_latency);
switch (wait_result)
{
case WAIT_OBJECT_0+0:
{
// Start or restart playback
bfr->Stop();
next_input_frame = start_frame;
DWORD buf_size; // size of buffer locked for filling
void *buf;
buffer_offset = 0;
if (FAILED(bfr->SetCurrentPosition(0)))
REPORT_ERROR("Could not reset playback buffer cursor before filling first buffer.")
HRESULT res = bfr->Lock(buffer_offset, 0, &buf, &buf_size, 0, 0, DSBLOCK_ENTIREBUFFER);
if (FAILED(res))
{
if (res == DSERR_BUFFERLOST)
{
// Try to regain the buffer
if (FAILED(bfr->Restore()) ||
FAILED(bfr->Lock(buffer_offset, 0, &buf, &buf_size, 0, 0, DSBLOCK_ENTIREBUFFER)))
{
REPORT_ERROR("Lost buffer and could not restore it.")
}
}
else
{
REPORT_ERROR("Could not lock buffer for playback.")
}
}
// Clear the buffer in case we can't fill it completely
memset(buf, 0, buf_size);
DWORD bytes_filled = FillAndUnlockBuffers(buf, buf_size, 0, 0, next_input_frame, bfr.obj);
buffer_offset += bytes_filled;
if (buffer_offset >= bufSize) buffer_offset -= bufSize;
if (FAILED(bfr->SetCurrentPosition(0)))
REPORT_ERROR("Could not reset playback buffer cursor before playback.")
if (bytes_filled < wanted_latency_bytes)
{
// Very short playback length, do without streaming playback
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
if (FAILED(bfr->Play(0, 0, 0)))
REPORT_ERROR("Could not start single-buffer playback.")
}
else
{
// We filled the entire buffer so there's reason to do streaming playback
current_latency = wanted_latency;
if (FAILED(bfr->Play(0, 0, DSBPLAY_LOOPING)))
REPORT_ERROR("Could not start looping playback.")
}
SetEvent(is_playing);
playback_should_be_running = true;
break;
}
case WAIT_OBJECT_0+1:
stop_playback:
// Stop playing
bfr->Stop();
ResetEvent(is_playing);
playback_should_be_running = false;
break;
case WAIT_OBJECT_0+2:
// Set end frame
if (end_frame <= next_input_frame)
{
goto stop_playback;
}
// If the user is dragging the start or end point in the audio display
// the set end frame events might come in faster than the timeouts happen
// and then new data never get filled into the buffer. See bug #915.
goto do_fill_buffer;
case WAIT_OBJECT_0+3:
// Change volume
// We aren't thread safe right now, filling the buffers grabs volume directly
// from the field set by the controlling thread, but it shouldn't be a major
// problem if race conditions do occur, just some momentary distortion.
goto do_fill_buffer;
case WAIT_OBJECT_0+4:
// Perform suicide
running = false;
goto stop_playback;
case WAIT_TIMEOUT:
do_fill_buffer:
{
// Time to fill more into buffer
if (!playback_should_be_running)
break;
DWORD status;
if (FAILED(bfr->GetStatus(&status)))
REPORT_ERROR("Could not get playback buffer status")
if (!(status & DSBSTATUS_LOOPING))
{
// Not looping playback...
// hopefully we only triggered timeout after being done with the buffer
goto stop_playback;
}
DWORD play_cursor;
if (FAILED(bfr->GetCurrentPosition(&play_cursor, 0)))
REPORT_ERROR("Could not get play cursor position for filling buffer.")
int bytes_needed = (int)play_cursor - (int)buffer_offset;
if (bytes_needed < 0) bytes_needed += (int)bufSize;
// Requesting zero buffer makes Windows cry, and zero buffer seemed to be
// a common request on Windows 7. (Maybe related to the new timer coalescing?)
// We'll probably get non-zero bytes requested on the next iteration.
if (bytes_needed == 0)
break;
DWORD buf1sz, buf2sz;
void *buf1, *buf2;
assert(bytes_needed > 0);
assert(buffer_offset < bufSize);
assert((DWORD)bytes_needed <= bufSize);
HRESULT res = bfr->Lock(buffer_offset, bytes_needed, &buf1, &buf1sz, &buf2, &buf2sz, 0);
switch (res)
{
case DSERR_BUFFERLOST:
// Try to regain the buffer
// When the buffer was lost the entire contents was lost too, so we have to start over
if (SUCCEEDED(bfr->Restore()) &&
SUCCEEDED(bfr->Lock(0, bufSize, &buf1, &buf1sz, &buf2, &buf2sz, 0)) &&
SUCCEEDED(bfr->Play(0, 0, DSBPLAY_LOOPING)))
{
LOG_D("audio/player/dsound") << "Lost and restored buffer";
break;
}
REPORT_ERROR("Lost buffer and could not restore it.")
case DSERR_INVALIDPARAM:
REPORT_ERROR("Invalid parameters to IDirectSoundBuffer8::Lock().")
case DSERR_INVALIDCALL:
REPORT_ERROR("Invalid call to IDirectSoundBuffer8::Lock().")
case DSERR_PRIOLEVELNEEDED:
REPORT_ERROR("Incorrect priority level set on DirectSoundBuffer8 object.")
default:
if (FAILED(res))
REPORT_ERROR("Could not lock audio buffer, unknown error.")
break;
}
DWORD bytes_filled = FillAndUnlockBuffers(buf1, buf1sz, buf2, buf2sz, next_input_frame, bfr.obj);
buffer_offset += bytes_filled;
if (buffer_offset >= bufSize) buffer_offset -= bufSize;
if (bytes_filled < 1024)
{
// Arbitrary low number, we filled in very little so better get back to filling in the rest with silence
// really fast... set latency to zero in this case.
current_latency = 0;
}
else if (bytes_filled < wanted_latency_bytes)
{
// Didn't fill as much as we wanted to, let's get back to filling sooner than normal
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
}
else
{
// Plenty filled in, do regular latency
current_latency = wanted_latency;
}
break;
}
default:
REPORT_ERROR("Something bad happened while waiting on events in playback loop, either the wait failed or an event object was abandoned.")
break;
}
}
}
#undef REPORT_ERROR
DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, void *buf2, DWORD buf2sz, int64_t &input_frame, IDirectSoundBuffer8 *bfr)
{
// Assume buffers have been locked and are ready to be filled
DWORD bytes_per_frame = provider->GetChannels() * provider->GetBytesPerSample();
DWORD buf1szf = buf1sz / bytes_per_frame;
DWORD buf2szf = buf2sz / bytes_per_frame;
if (input_frame >= end_frame)
{
// Silence
if (buf1)
memset(buf1, 0, buf1sz);
if (buf2)
memset(buf2, 0, buf2sz);
input_frame += buf1szf + buf2szf;
bfr->Unlock(buf1, buf1sz, buf2, buf2sz); // should be checking for success
return buf1sz + buf2sz;
}
if (buf1 && buf1sz)
{
if (buf1szf + input_frame > end_frame)
{
buf1szf = end_frame - input_frame;
buf1sz = buf1szf * bytes_per_frame;
buf2szf = 0;
buf2sz = 0;
}
provider->GetAudioWithVolume(buf1, input_frame, buf1szf, volume);
input_frame += buf1szf;
}
if (buf2 && buf2sz)
{
if (buf2szf + input_frame > end_frame)
{
buf2szf = end_frame - input_frame;
buf2sz = buf2szf * bytes_per_frame;
}
provider->GetAudioWithVolume(buf2, input_frame, buf2szf, volume);
input_frame += buf2szf;
}
bfr->Unlock(buf1, buf1sz, buf2, buf2sz); // bad? should check for success
return buf1sz + buf2sz;
}
void DirectSoundPlayer2Thread::CheckError()
{
try
{
switch (WaitForSingleObject(error_happened, 0))
{
case WAIT_OBJECT_0:
throw error_message;
case WAIT_ABANDONED:
throw "The DirectShowPlayer2Thread error signal event was abandoned, somehow. This should not happen.";
case WAIT_FAILED:
throw "Failed checking state of DirectShowPlayer2Thread error signal event.";
case WAIT_TIMEOUT:
default:
return;
}
}
catch (...)
{
ResetEvent(is_playing);
ResetEvent(thread_running);
throw;
}
}
DirectSoundPlayer2Thread::DirectSoundPlayer2Thread(AudioProvider *provider, int WantedLatency, int BufferLength)
: event_start_playback (CreateEvent(0, FALSE, FALSE, 0))
, event_stop_playback (CreateEvent(0, FALSE, FALSE, 0))
, event_update_end_time (CreateEvent(0, FALSE, FALSE, 0))
, event_set_volume (CreateEvent(0, FALSE, FALSE, 0))
, event_kill_self (CreateEvent(0, FALSE, FALSE, 0))
, thread_running (CreateEvent(0, TRUE, FALSE, 0))
, is_playing (CreateEvent(0, TRUE, FALSE, 0))
, error_happened (CreateEvent(0, FALSE, FALSE, 0))
, wanted_latency(WantedLatency)
, buffer_length(BufferLength)
, provider(provider)
{
error_message = 0;
volume = 1.0;
start_frame = 0;
end_frame = 0;
thread_handle = (HANDLE)_beginthreadex(0, 0, ThreadProc, this, 0, 0);
if (!thread_handle)
throw agi::AudioPlayerOpenError("Failed creating playback thread in DirectSoundPlayer2. This is bad.", 0);
HANDLE running_or_error[] = { thread_running, error_happened };
switch (WaitForMultipleObjects(2, running_or_error, FALSE, INFINITE))
{
case WAIT_OBJECT_0:
// running, all good
return;
case WAIT_OBJECT_0 + 1:
// error happened, we fail
throw agi::AudioPlayerOpenError(error_message, 0);
default:
throw agi::AudioPlayerOpenError("Failed wait for thread start or thread error in DirectSoundPlayer2. This is bad.", 0);
}
}
DirectSoundPlayer2Thread::~DirectSoundPlayer2Thread()
{
SetEvent(event_kill_self);
WaitForSingleObject(thread_handle, INFINITE);
}
void DirectSoundPlayer2Thread::Play(int64_t start, int64_t count)
{
CheckError();
start_frame = start;
end_frame = start+count;
SetEvent(event_start_playback);
last_playback_restart = GetTickCount();
// Block until playback actually begins to avoid race conditions with
// checking if playback is in progress
HANDLE events_to_wait[] = { is_playing, error_happened };
switch (WaitForMultipleObjects(2, events_to_wait, FALSE, INFINITE))
{
case WAIT_OBJECT_0+0: // Playing
LOG_D("audio/player/dsound") << "Playback begun";
break;
case WAIT_OBJECT_0+1: // Error
throw error_message;
default:
throw agi::InternalError("Unexpected result from WaitForMultipleObjects in DirectSoundPlayer2Thread::Play", 0);
}
}
void DirectSoundPlayer2Thread::Stop()
{
CheckError();
SetEvent(event_stop_playback);
}
void DirectSoundPlayer2Thread::SetEndFrame(int64_t new_end_frame)
{
CheckError();
end_frame = new_end_frame;
SetEvent(event_update_end_time);
}
void DirectSoundPlayer2Thread::SetVolume(double new_volume)
{
CheckError();
volume = new_volume;
SetEvent(event_set_volume);
}
bool DirectSoundPlayer2Thread::IsPlaying()
{
CheckError();
switch (WaitForSingleObject(is_playing, 0))
{
case WAIT_ABANDONED:
throw "The DirectShowPlayer2Thread playback state event was abandoned, somehow. This should not happen.";
case WAIT_FAILED:
throw "Failed checking state of DirectShowPlayer2Thread playback state event.";
case WAIT_OBJECT_0:
return true;
case WAIT_TIMEOUT:
default:
return false;
}
}
int64_t DirectSoundPlayer2Thread::GetStartFrame()
{
CheckError();
return start_frame;
}
int64_t DirectSoundPlayer2Thread::GetCurrentFrame()
{
CheckError();
if (!IsPlaying()) return 0;
int64_t milliseconds_elapsed = GetTickCount() - last_playback_restart;
return start_frame + milliseconds_elapsed * provider->GetSampleRate() / 1000;
}
int64_t DirectSoundPlayer2Thread::GetEndFrame()
{
CheckError();
return end_frame;
}
double DirectSoundPlayer2Thread::GetVolume()
{
CheckError();
return volume;
}
bool DirectSoundPlayer2Thread::IsDead()
{
switch (WaitForSingleObject(thread_running, 0))
{
case WAIT_OBJECT_0:
return false;
default:
return true;
}
}
DirectSoundPlayer2::DirectSoundPlayer2(AudioProvider *provider)
: AudioPlayer(provider)
{
// The buffer will hold BufferLength times WantedLatency milliseconds of audio
WantedLatency = OPT_GET("Player/Audio/DirectSound/Buffer Latency")->GetInt();
BufferLength = OPT_GET("Player/Audio/DirectSound/Buffer Length")->GetInt();
// sanity checking
if (WantedLatency <= 0)
WantedLatency = 100;
if (BufferLength <= 0)
BufferLength = 5;
try
{
thread.reset(new DirectSoundPlayer2Thread(provider, WantedLatency, BufferLength));
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
throw agi::AudioPlayerOpenError(msg, 0);
}
}
DirectSoundPlayer2::~DirectSoundPlayer2()
{
}
bool DirectSoundPlayer2::IsThreadAlive()
{
if (thread && thread->IsDead())
{
thread.reset();
}
return thread;
}
void DirectSoundPlayer2::Play(int64_t start,int64_t count)
{
try
{
thread->Play(start, count);
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
}
}
void DirectSoundPlayer2::Stop()
{
try
{
if (IsThreadAlive()) thread->Stop();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
}
}
bool DirectSoundPlayer2::IsPlaying()
{
try
{
if (!IsThreadAlive()) return false;
return thread->IsPlaying();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
return false;
}
}
int64_t DirectSoundPlayer2::GetStartPosition()
{
try
{
if (!IsThreadAlive()) return 0;
return thread->GetStartFrame();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
return 0;
}
}
int64_t DirectSoundPlayer2::GetEndPosition()
{
try
{
if (!IsThreadAlive()) return 0;
return thread->GetEndFrame();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
return 0;
}
}
int64_t DirectSoundPlayer2::GetCurrentPosition()
{
try
{
if (!IsThreadAlive()) return 0;
return thread->GetCurrentFrame();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
return 0;
}
}
void DirectSoundPlayer2::SetEndPosition(int64_t pos)
{
try
{
if (IsThreadAlive()) thread->SetEndFrame(pos);
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
}
}
void DirectSoundPlayer2::SetCurrentPosition(int64_t pos)
{
try
{
if (IsThreadAlive()) thread->Play(pos, thread->GetEndFrame()-pos);
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
}
}
void DirectSoundPlayer2::SetVolume(double vol)
{
try
{
if (IsThreadAlive()) thread->SetVolume(vol);
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
}
}
double DirectSoundPlayer2::GetVolume()
{
try
{
if (!IsThreadAlive()) return 0;
return thread->GetVolume();
}
catch (const char *msg)
{
LOG_E("audio/player/dsound") << msg;
return 0;
}
}
#endif // WITH_DIRECTSOUND