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