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itgmania212121/stepmania/src/arch/Sound/RageSoundDriver_Generic_Software.cpp
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14 KiB
C++

#include "global.h"
#include "RageSoundDriver_Generic_Software.h"
#include "RageLog.h"
#include "RageSound.h"
#include "RageUtil.h"
#include "RageSoundManager.h"
#include "RageSoundMixBuffer.h"
static const int channels = 2;
static const int bytes_per_frame = channels*2; /* 16-bit */
/* When a sound has fewer than min_fill_frames buffered, buffer at maximum speed.
* Once beyond that, fill at a limited rate. */
static const int min_fill_frames = 1024*4;
static int frames_to_buffer;
static int min_streaming_buffer_size = 1024*32;
/* 512 is about 10ms, which is big enough for the tolerance of most schedulers. */
static int chunksize() { return 512; }
static int underruns = 0, logged_underruns = 0;
RageSound_Generic_Software::sound::sound()
{
snd = NULL;
state = STOPPED;
available = true;
}
void RageSound_Generic_Software::sound::Allocate( int frames )
{
/* Reserve enough blocks in the buffer to hold the buffer. Add one, to account for
* the fact that we may have a partial block due to a previous Mix() call. */
const int frames_per_block = samples_per_block / channels;
const int blocks_to_prebuffer = frames / frames_per_block;
buffer.reserve( blocks_to_prebuffer + 1 );
}
void RageSound_Generic_Software::sound::Deallocate()
{
buffer.reserve( 0 );
}
int RageSound_Generic_Software::DecodeThread_start( void *p )
{
((RageSound_Generic_Software *) p)->DecodeThread();
return 0;
}
static int g_TotalAhead = 0;
static int g_TotalAheadCount = 0;
void RageSound_Generic_Software::Mix( int16_t *buf, int frames, int64_t frameno, int64_t current_frameno )
{
ASSERT_M( m_DecodeThread.IsCreated(), "RageSound_Generic_Software::StartDecodeThread() was never called" );
if( frameno - current_frameno + frames > 0 )
{
g_TotalAhead += (int) (frameno - current_frameno + frames);
++g_TotalAheadCount;
}
static RageSoundMixBuffer mix;
for( unsigned i = 0; i < ARRAYSIZE(sounds); ++i )
{
/* s.snd can not safely be accessed from here. */
sound &s = sounds[i];
if( s.state == sound::HALTING )
{
/* This indicates that this stream can be reused. */
s.Deallocate();
s.state = sound::STOPPED;
s.available = true; /* do this last */
// LOG->Trace("set %p from HALTING to STOPPED", sounds[i].snd);
continue;
}
if( s.state != sound::STOPPING && s.state != sound::PLAYING )
continue;
/* STOPPING or PLAYING. Read sound data. */
int got_frames = 0;
int frames_left = frames;
/* Does the sound have a start time? */
if( !s.start_time.IsZero() && current_frameno != -1 )
{
/* If the sound is supposed to start at a time past this buffer, insert silence. */
const int64_t iFramesUntilThisBuffer = frameno - current_frameno;
const float fSecondsBeforeStart = -s.start_time.Ago();
const int64_t iFramesBeforeStart = int64_t(fSecondsBeforeStart * GetSampleRate(0));
const int iSilentFramesInThisBuffer = clamp( int(iFramesBeforeStart-iFramesUntilThisBuffer), 0, frames_left );
got_frames += iSilentFramesInThisBuffer;
frames_left -= iSilentFramesInThisBuffer;
/* If we didn't completely fill the buffer, then we've written all of the silence. */
if( frames_left )
s.start_time.SetZero();
}
/* Fill actual data. */
sound_block *p[2];
unsigned pSize[2];
s.buffer.get_read_pointers( p, pSize );
while( frames_left && pSize[0] )
{
if( !p[0]->frames_in_buffer )
{
/* We've processed all of the sound in this block. Mark it read. */
s.buffer.advance_read_pointer( 1 );
++p[0];
--pSize[0];
/* If we have more data in p[0], keep going. */
if( pSize[0] )
continue; // more data
/* We've used up p[0]. Try p[1]. */
swap( p[0], p[1] );
swap( pSize[0], pSize[1] );
continue;
}
/* Note that, until we call advance_read_pointer, we can safely write to p[0]. */
const int frames_to_read = min( frames_left, p[0]->frames_in_buffer );
mix.SetVolume( s.volume );
mix.SetWriteOffset( got_frames*channels );
mix.write( p[0]->p, frames_to_read * channels );
SOUNDMAN->CommitPlayingPosition( s.sound_id, frameno+got_frames, p[0]->position, frames_to_read );
p[0]->p += frames_to_read*channels;
p[0]->frames_in_buffer -= frames_to_read;
p[0]->position += frames_to_read;
// LOG->Trace( "incr fr rd += %i (state %i) (%p)",
// (int) frames_to_read, s.state, s.snd );
got_frames += frames_to_read;
frames_left -= frames_to_read;
}
/* If we don't have enough to fill the buffer, we've underrun. */
if( got_frames < frames && s.state == sound::PLAYING )
++underruns;
}
memset( buf, 0, frames*bytes_per_frame );
mix.read( buf );
}
void RageSound_Generic_Software::DecodeThread()
{
/* SOUNDMAN will be set once RageSoundManager's ctor returns and
* assigns it; we might get here before that happens, though. */
while( !SOUNDMAN && !shutdown_decode_thread )
usleep( 10000 );
SetupDecodingThread();
while( !shutdown_decode_thread )
{
/* Fill each playing sound, round-robin. */
usleep( 1000000*chunksize() / GetSampleRate(0) );
LockMut( m_Mutex );
// LOG->Trace("begin mix");
for( unsigned i = 0; i < ARRAYSIZE(sounds); ++i )
{
/* The volume can change while the sound is playing; update it. */
if( sounds[i].state == sound::PLAYING || sounds[i].state == sound::STOPPING )
sounds[i].volume = sounds[i].snd->GetVolume();
}
/*
* If a buffer is low on data, keep filling until it has a reasonable amount.
* However, once beyond a certain threshold, clamp the rate at which we fill
* it. For example, if the threshold is 4k frames, and we have a 32k frame
* buffer, fill the buffer as fast as we can until it reaches 4k frames; but
* beyond that point, only fill it at a rate relative to realtime (for example,
* at 2x realtime).
*
* This allows a stream to have a large buffer, for higher reliability, without
* causing major CPU bursts when the stream starts or underruns. (Filling 32k
* takes more CPU than filling 4k frames, and may cause a gameplay skip.)
*/
for( unsigned i = 0; i < ARRAYSIZE(sounds); ++i )
{
if( sounds[i].state != sound::PLAYING )
continue;
sound *pSound = &sounds[i];
CHECKPOINT;
int frames_filled = 0;
while( pSound->buffer.num_writable() )
{
/* If there are more than min_fill_frames available, check for
* rate clamping. */
if( pSound->buffer.num_readable()*samples_per_block >= unsigned(min_fill_frames) )
{
/* Don't write more than two chunks worth of data in one
* iteration. Since we delay for one chunk period per loop,
* this means we'll fill at no more than ealtime. */
if( frames_filled >= chunksize()*4 )
break;
}
int wrote = GetDataForSound( *pSound );
if( !wrote )
{
/* This sound is finishing. */
pSound->state = sound::STOPPING;
break;
// LOG->Trace("mixer: (#%i) eof (%p)", i, pSound->snd );
}
frames_filled += wrote;
}
}
// LOG->Trace("end mix");
}
}
/* Buffer a block of sound data for the given sound. Return the number of
* frames buffered. If end of file is reached, return 0. */
int RageSound_Generic_Software::GetDataForSound( sound &s )
{
sound_block *p[2];
unsigned psize[2];
s.buffer.get_write_pointers( p, psize );
/* If we have no open buffer slot, we have a buffer overflow. */
ASSERT( psize[0] > 0 );
sound_block *b = p[0];
int size = ARRAYSIZE(b->buf)/channels;
bool eof = !s.snd->GetDataToPlay( b->buf, size, b->position, b->frames_in_buffer );
b->p = b->buf;
s.buffer.advance_write_pointer( 1 );
// LOG->Trace( "incr fr wr %i (state %i) (%p)",
// (int) b->frames_in_buffer, s.state, s.snd );
return eof? 0:size;
}
void RageSound_Generic_Software::Update(float delta)
{
/* We must not lock here, since the decoder thread might hold the lock for a
* while at a time. This is threadsafe, because once a sound is in STOPPING,
* this is the only place it'll be changed (to STOPPED). */
for( unsigned i = 0; i < ARRAYSIZE(sounds); ++i )
{
if( sounds[i].state != sound::STOPPING )
continue;
if( sounds[i].buffer.num_readable() != 0 )
continue;
// LOG->Trace("finishing sound %i", i);
sounds[i].snd->SoundIsFinishedPlaying();
/* This sound is done. Set it to HALTING, since the mixer thread might
* be accessing it; it'll change it back to STOPPED once it's ready to
* be used again. */
sounds[i].state = sound::HALTING;
// LOG->Trace("set (#%i) %p from STOPPING to HALTING", i, sounds[i].snd);
}
static float fNext = 0;
if( RageTimer::GetTimeSinceStart() >= fNext )
{
/* Lockless: only Mix() can write to underruns. */
int current_underruns = underruns;
if( current_underruns > logged_underruns )
{
LOG->MapLog( "GenericMixingUnderruns", "Mixing underruns: %i", current_underruns - logged_underruns );
LOG->Trace( "Mixing underruns: %i", current_underruns - logged_underruns );
logged_underruns = current_underruns;
/* Don't log again for at least a second, or we'll burst output
* and possibly cause more underruns. */
fNext = RageTimer::GetTimeSinceStart() + 1;
}
}
}
void RageSound_Generic_Software::StartMixing( RageSoundBase *snd )
{
/* Lock available sounds[], and reserve a slot. */
m_SoundListMutex.Lock();
unsigned i;
for( i = 0; i < ARRAYSIZE(sounds); ++i )
if( sounds[i].available )
break;
if( i == ARRAYSIZE(sounds) )
{
m_SoundListMutex.Unlock();
return;
}
sound &s = sounds[i];
s.available = false;
/* We've reserved our slot; we can safely unlock now. Don't hold onto it longer
* than needed, since prebuffering might take some time. */
m_SoundListMutex.Unlock();
s.snd = snd;
s.start_time = snd->GetStartTime();
s.sound_id = snd->GetID();
s.volume = snd->GetVolume();
s.buffer.clear();
/* Initialize the sound buffer. */
int BufferSize = frames_to_buffer;
/* If a sound is streaming from disk, use a bigger buffer, so we don't underrun
* the BGM if a drive seek takes too long. Note that in this case, we'll still
* underrun any other sounds that are playing, even if they're preloaded. This
* is less of a problem, since the music position isn't based on those. It could
* be fixed by having two decoding threads; one for streaming sounds and one for
* non-streaming sounds. */
if( s.snd->IsStreamingFromDisk() )
BufferSize = max( BufferSize, min_streaming_buffer_size );
s.Allocate( BufferSize );
// LOG->Trace("StartMixing(%s) (%p)", s.snd->GetLoadedFilePath().c_str(), s.snd );
/* Prebuffer some frames before changing the sound to PLAYING. */
bool ReachedEOF = false;
int frames_filled = 0;
while( !ReachedEOF && frames_filled < min_fill_frames && frames_filled < BufferSize )
{
// LOG->Trace("StartMixing: (#%i) buffering %i (%i writable) (%p)", i, (int) frames_to_buffer, s.buffer.num_writable(), s.snd );
int wrote = GetDataForSound( s );
frames_filled += wrote;
if( !wrote )
{
// LOG->Trace("StartMixing: XXX hit EOF (%p)", s.snd );
ReachedEOF = true;
}
}
/* If we hit EOF already, while prebuffering, then go right to STOPPING. */
s.state = ReachedEOF? sound::STOPPING: sound::PLAYING;
// LOG->Trace("StartMixing: (#%i) finished prebuffering(%s) (%p)", i, s.snd->GetLoadedFilePath().c_str(), s.snd );
}
void RageSound_Generic_Software::StopMixing( RageSoundBase *snd )
{
/* Lock, to make sure the decoder thread isn't running on this sound while we do this. */
LockMut( m_Mutex );
/* Find the sound. */
unsigned i;
for( i = 0; i < ARRAYSIZE(sounds); ++i )
if( !sounds[i].available && sounds[i].snd == snd )
break;
if( i == ARRAYSIZE(sounds) )
{
LOG->Trace( "not stopping a sound because it's not playing" );
return;
}
/* If we're already in STOPPED, there's nothing to do. */
if( sounds[i].state == sound::STOPPED )
{
LOG->Trace( "not stopping a sound because it's already in STOPPED" );
return;
}
// LOG->Trace("StopMixing: set %p (%s) to HALTING", sounds[i].snd, sounds[i].snd->GetLoadedFilePath().c_str());
/* Tell the mixing thread to flush the buffer. We don't have to worry about
* the decoding thread, since we've locked m_Mutex. */
sounds[i].state = sound::HALTING;
/* Invalidate the snd pointer to guarantee we don't make any further references to
* it. Once this call returns, the sound may no longer exist. */
sounds[i].snd = NULL;
// LOG->Trace("end StopMixing");
}
void RageSound_Generic_Software::StartDecodeThread()
{
ASSERT( !m_DecodeThread.IsCreated() );
m_DecodeThread.Create( DecodeThread_start, this );
}
void RageSound_Generic_Software::SetDecodeBufferSize( int frames )
{
ASSERT( !m_DecodeThread.IsCreated() );
frames_to_buffer = frames;
}
RageSound_Generic_Software::RageSound_Generic_Software():
m_Mutex("RageSound_Generic_Software"),
m_SoundListMutex("SoundListMutex")
{
shutdown_decode_thread = false;
SetDecodeBufferSize( 4096 );
m_DecodeThread.SetName("Decode thread");
}
RageSound_Generic_Software::~RageSound_Generic_Software()
{
/* Signal the decoding thread to quit. */
if( m_DecodeThread.IsCreated() )
{
shutdown_decode_thread = true;
LOG->Trace("Shutting down decode thread ...");
LOG->Flush();
m_DecodeThread.Wait();
LOG->Trace("Decode thread shut down.");
LOG->Flush();
LOG->Info( "Mixing %f ahead in %i Mix() calls",
float(g_TotalAhead) / max( g_TotalAheadCount, 1 ), g_TotalAheadCount );
}
}
/*
* (c) 2002-2004 Glenn Maynard
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the
* "Software"), to deal in the Software without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, and/or sell copies of the Software, and to permit persons to
* whom the Software is furnished to do so, provided that the above
* copyright notice(s) and this permission notice appear in all copies of
* the Software and that both the above copyright notice(s) and this
* permission notice appear in supporting documentation.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF
* THIRD PARTY RIGHTS. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR HOLDERS
* INCLUDED IN THIS NOTICE BE LIABLE FOR ANY CLAIM, OR ANY SPECIAL INDIRECT
* OR CONSEQUENTIAL DAMAGES, OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS
* OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR
* OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
* PERFORMANCE OF THIS SOFTWARE.
*/