mirror of
https://github.com/trudnai/Steve2.git
synced 2024-06-13 09:29:36 +00:00
Eliminated lots of click unwanted noise
This commit is contained in:
tudnai
parent
70fd9484c8
commit
47cae17e6d
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@ -8,44 +8,52 @@
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###
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import os
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import sys
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import struct
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binpath = '/Users/trudnai/Library/Containers/com.trudnai.steveii/Data/'
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binfilename = binpath + 'steve2_audio_debug.bin'
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data_size = os.path.getsize(binfilename)
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channels = 2
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sample_rate = 192000 # must be the same as in speaker.c : spkr_sample_rate
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bits_per_sample = 16
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header_size = 44
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# binpath = '/Users/trudnai/Library/Containers/com.trudnai.steveii/Data/'
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# binfilename = binpath + 'steve2_audio_debug.bin'
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with open( binfilename, 'rb') as binfile:
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with open( 'steve2_audio_debug.wav', 'wb+') as wavfile:
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# WAV HEADER
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wavfile.write( 'RIFF'.encode() ) # Marks the file as a riff file. Characters are each 1 byte long
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wavfile.write( struct.pack('I', data_size + header_size ) ) # Size of the overall file - 8 bytes, in bytes (32-bit integer). Typically you’d fill this in after creation
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wavfile.write( 'WAVE'.encode() ) # File Type Header. For our purposes, it always equals “WAVE”
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# FMT HEADER
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wavfile.write( 'fmt '.encode() ) # Format chunk marker. Includes trailing null (tr: Space works, null don't)
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wavfile.write( struct.pack('I', 16 ) ) # Length of format data as listed above
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# FMT DATA
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wavfile.write( struct.pack('H', 1 ) ) # Type of format (1 is PCM) - 2 byte integer
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wavfile.write( struct.pack('H', channels ) ) # Number of Channels - 2 byte integer
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wavfile.write( struct.pack('I', sample_rate ) ) # Sample Rate - 32 byte integer. Common values are 44100 (CD), 48000 (DAT). Sample Rate = Number of Samples per second, or Hertz
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wavfile.write( struct.pack('I', int(sample_rate * bits_per_sample * channels / 8) ) ) # (Sample Rate * BitsPerSample * Channels) / 8
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wavfile.write( struct.pack('H', 4 ) ) # 0: (BitsPerSample * Channels) / 8
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# 1: 8 bit mono2
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# 2: 8 bit stereo
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# 3: 16 bit mono
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# 4: 16 bit stereo
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wavfile.write( struct.pack('H', bits_per_sample ) ) # Bits per sample
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# DATA HEADER
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wavfile.write( 'data'.encode() ) # “data” chunk header. Marks the beginning of the data section
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wavfile.write( struct.pack('I', data_size ) ) # Size of the data section
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# // WAV DATA
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wavfile.write( binfile.read() )
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def convert_bin_to_audio(filename):
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data_size = os.path.getsize(filename)
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channels = 2
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sample_rate = 192000 # must be the same as in speaker.c : spkr_sample_rate
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bits_per_sample = 16
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header_size = 44
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with open( filename, 'rb') as binfile:
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wavfilename = os.path.splitext( os.path.basename(filename) )[0] + '.wav'
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with open( wavfilename, 'wb+') as wavfile:
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# WAV HEADER
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wavfile.write( 'RIFF'.encode() ) # Marks the file as a riff file. Characters are each 1 byte long
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wavfile.write( struct.pack('I', data_size + header_size ) ) # Size of the overall file - 8 bytes, in bytes (32-bit integer). Typically you’d fill this in after creation
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wavfile.write( 'WAVE'.encode() ) # File Type Header. For our purposes, it always equals “WAVE”
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# FMT HEADER
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wavfile.write( 'fmt '.encode() ) # Format chunk marker. Includes trailing null (tr: Space works, null don't)
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wavfile.write( struct.pack('I', 16 ) ) # Length of format data as listed above
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# FMT DATA
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wavfile.write( struct.pack('H', 1 ) ) # Type of format (1 is PCM) - 2 byte integer
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wavfile.write( struct.pack('H', channels ) ) # Number of Channels - 2 byte integer
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wavfile.write( struct.pack('I', sample_rate ) ) # Sample Rate - 32 byte integer. Common values are 44100 (CD), 48000 (DAT). Sample Rate = Number of Samples per second, or Hertz
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wavfile.write( struct.pack('I', int(sample_rate * bits_per_sample * channels / 8) ) ) # (Sample Rate * BitsPerSample * Channels) / 8
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wavfile.write( struct.pack('H', 4 ) ) # 0: (BitsPerSample * Channels) / 8
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# 1: 8 bit mono2
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# 2: 8 bit stereo
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# 3: 16 bit mono
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# 4: 16 bit stereo
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wavfile.write( struct.pack('H', bits_per_sample ) ) # Bits per sample
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# DATA HEADER
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wavfile.write( 'data'.encode() ) # “data” chunk header. Marks the beginning of the data section
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wavfile.write( struct.pack('I', data_size ) ) # Size of the data section
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# // WAV DATA
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wavfile.write( binfile.read() )
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for filename in sys.argv:
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convert_bin_to_audio(filename)
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@ -175,11 +175,12 @@ void spkr_load_sfx( const char * bundlePath ) {
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#undef SPKR_KEEP_PITCH
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#undef SPKR_DEBUG
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#undef SPKR_DEBUG_BEFORE_EMA
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#ifdef SPKR_DEBUG
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static const char * spkr_debug_filename = "steve2_audio_debug.bin";
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FILE * spkr_debug_file = NULL;
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static const char * spkr_debug_raw_filename = "steve2_audio_debug_raw.bin";
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static const char * spkr_debug_ema_filename = "steve2_audio_debug_ema.bin";
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FILE * spkr_debug_raw_file = NULL;
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FILE * spkr_debug_ema_file = NULL;
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#endif
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@ -202,20 +203,39 @@ void spkr_fade(float fadeLevel, unsigned idx) {
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}
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// Fill with Zero to avoid pops
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memset(spkr_samples + idx * sizeof(spkr_sample_t), 0, spkr_buf_size * sizeof(spkr_sample_t) * SPKR_CHANNELS);
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memset(spkr_samples + idx, 0, spkr_buf_size * sizeof(spkr_sample_t) * SPKR_CHANNELS);
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}
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/// opens debug dump files for audio data -- if enabled
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#ifdef SPKR_DEBUG
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static void spkr_debug_init() {
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spkr_debug_raw_file = fopen(spkr_debug_raw_filename, "w+");
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spkr_debug_ema_file = fopen(spkr_debug_ema_filename, "w+");
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}
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#else
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#define spkr_debug_init()
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#endif
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/// dumps audio data -- if enabled
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#ifdef SPKR_DEBUG
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static void spkr_debug(FILE * file) {
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fwrite(spkr_samples, sizeof(spkr_sample_t), spkr_buf_size, file);
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fflush(file);
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}
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#else
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#define spkr_debug(f)
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#endif
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// initialize OpenAL
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void spkr_init() {
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const char *defname = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
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printf( "Default device: %s\n", defname );
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// DEBUG ONLY!!!
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#ifdef SPKR_DEBUG
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spkr_debug_file = fopen(spkr_debug_filename, "w+");
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#endif
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// opens debug dump files for audio data -- if enabled
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spkr_debug_init();
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// restart OpenAL when restarting the virtual machine
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spkr_exit();
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@ -232,10 +252,8 @@ void spkr_init() {
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freeBuffers--;
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// DEBUG ONLY!!!
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#ifdef SPKR_DEBUG
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fwrite(spkr_samples, sizeof(spkr_sample_t), spkr_sample_idx, spkr_debug_file);
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fflush(spkr_debug_file);
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#endif
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spkr_debug(spkr_debug_raw_file);
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spkr_debug(spkr_debug_ema_file);
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alBufferData(spkr_buffers[freeBuffers], AL_FORMAT_STEREO16, spkr_samples, spkr_sample_idx * sizeof(spkr_sample_t), spkr_sample_rate);
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al_check_error();
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@ -359,7 +377,7 @@ int spkr_unqueue( ALuint src ) {
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// printf("pf:%d\n", processed);
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alSourceUnqueueBuffers( src, processed, &spkr_buffers[freeBuffers]);
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al_check_error();
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freeBuffers = clamp( 1, freeBuffers + processed, BUFFER_COUNT );
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freeBuffers = clamp( 0, freeBuffers + processed, BUFFER_COUNT );
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// alGetSourcei ( src, AL_BUFFERS_PROCESSED, &processed );
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// al_check_error();
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@ -435,15 +453,15 @@ void spkr_toggle_square ( int level_max, const unsigned idx_diff ) {
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}
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// start a new speaker level -- it might not be necessary, but...
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spkr_samples[ spkr_sample_idx++ ] = level_max;
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spkr_samples[ spkr_sample_idx++ ] = level_max; // stereo
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// spkr_samples[ spkr_sample_idx++ ] = level_max;
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// spkr_samples[ spkr_sample_idx++ ] = level_max; // stereo
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spkr_level = level_max;
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}
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void spkr_toggle_tick ( int level_max, const unsigned idx_diff ) {
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spkr_level = SPKR_LEVEL_ZERO;
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spkr_level = level_max;
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spkr_samples[ spkr_sample_idx++ ] = level_max;
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spkr_samples[ spkr_sample_idx++ ] = level_max; // stereo
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spkr_sample_last_idx = spkr_sample_idx;
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@ -476,8 +494,6 @@ void spkr_toggle() {
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if ( clk_6502_per_frm_set < clk_6502_per_frm_max_sound ) {
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spkr_play_time = spkr_play_timeout;
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// push a click into the speaker buffer
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// (we will play the entire buffer at the end of the frame)
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double multiplier = (double)spkr_sample_rate / MHZ(MHz_6502);
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@ -492,6 +508,13 @@ void spkr_toggle() {
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spkr_sample_idx = round( (double)(spkr_clk + m6502.clkfrm) * multiplier ) * SPKR_CHANNELS;
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spkr_sample_idx &= UINTMAX_MAX - 1;
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// if play stopped, we should make sure we are not creating a false initial quare wave
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if (spkr_play_time <= 0) {
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spkr_sample_last_idx = spkr_sample_idx;
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}
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spkr_play_time = spkr_play_timeout;
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unsigned spkr_sample_idx_diff = spkr_sample_idx - spkr_sample_last_idx;
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if ( (int)spkr_sample_idx_diff < 0 ) {
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@ -521,11 +544,11 @@ void spkr_toggle() {
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}
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#define SPKR_PLAY_DELAY 2
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#define SPKR_PLAY_DELAY 1
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int playDelay = SPKR_PLAY_DELAY;
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INLINE void filter_reset() {
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INLINE static void spkr_filter_reset() {
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spkr_level = SPKR_LEVEL_ZERO;
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spkr_level_ema = SPKR_LEVEL_ZERO;
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spkr_level_dema = SPKR_LEVEL_ZERO;
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@ -533,14 +556,10 @@ INLINE void filter_reset() {
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}
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INLINE void filter() {
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INLINE static void spkr_filter(int buf_len) {
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#ifdef SPKR_DEBUG // Debug SPKR Buffer Before EMA
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#ifdef SPKR_DEBUG_BEFORE_EMA
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fwrite(spkr_samples, 1, buf_len, spkr_debug_file);
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fflush(spkr_debug_file);
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#endif
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#endif
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// Debug SPKR Buffer Before EMA
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spkr_debug(spkr_debug_raw_file);
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// to use with EMA
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// static const int ema_len_sharper = 30;
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@ -550,33 +569,28 @@ INLINE void filter() {
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// to use with TEMA
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static const int ema_len_sharper = 7;
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static const int ema_len_sharp = 14;
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static const int ema_len_normal = 20;
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static const int ema_len_soft = 30;
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static const int ema_len_normal = 18;
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static const int ema_len_soft = 20;
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static const int ema_len_supersoft = 40;
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static const int ema_len = ema_len_normal;
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for ( int i = 0; i <= spkr_buf_size; ) {
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for ( int i = 0; i < spkr_buf_size; ) {
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spkr_level_ema = ema(spkr_samples[i], spkr_level_ema, ema_len);
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spkr_level_dema = ema(spkr_level_ema, spkr_level_dema, ema_len);
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spkr_level_tema = ema(spkr_level_dema, spkr_level_tema, ema_len);
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// smoothing with Tripple EMA
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// smoothing with Tripple EMA
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spkr_samples[i++] = spkr_level_tema;
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spkr_samples[i++] = spkr_level_tema;
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}
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#ifdef SPKR_DEBUG // Debug SPKR Buffer After EMA
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#ifdef SPKR_DEBUG_AFTER_EMA
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fwrite(spkr_samples, 1, buf_len, spkr_debug_file);
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fflush(spkr_debug_file);
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#endif
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#endif
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// Debug SPKR Buffer After EMA
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spkr_debug(spkr_debug_ema_file);
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}
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INLINE void debug_spike() {
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INLINE void spkr_debug_spike() {
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spkr_sample_idx = 0;
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spkr_samples[ spkr_sample_idx++ ] = -28000;
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spkr_samples[ spkr_sample_idx++ ] = 28000; // stereo
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@ -599,136 +613,139 @@ void spkr_update() {
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al_check_error();
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// printf("q:%d clkfrm:%d frm:%llu max:%llu\n", queued, clkfrm, clk_6502_per_frm, clk_6502_per_frm_max);
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if ( ( spkr_play_time ) && ( queued < SPKR_MAX_QUEUED ) ) {
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if ( freeBuffers ) {
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double multiplier = 1;
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#ifdef SPKR_KEEP_PITCH
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if (MHz_6502 > default_MHz_6502 ) {
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multiplier = 1 / MHz_6502;
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}
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#endif
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// in Game Mode do not fade out and stop playing
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if ( /*( cpuMode_game != cpuMode ) && */( --spkr_play_time == 0 ) ) {
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if ( queued < SPKR_MAX_QUEUED ) {
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if ( spkr_play_time ) {
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if ( freeBuffers ) {
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// double multiplier = 1;
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// #ifdef SPKR_KEEP_PITCH
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// if (MHz_6502 > default_MHz_6502 ) {
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// multiplier = 1 / MHz_6502;
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// }
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// #endif
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if (--freeBuffers < 0) {
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printf("freeBuffer < 0 (%i)\n", freeBuffers);
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freeBuffers = 0;
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// in Game Mode do not fade out and stop playing
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if ( /*( cpuMode_game != cpuMode ) && */( --spkr_play_time == 0 ) ) {
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if (--freeBuffers < 0) {
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printf("freeBuffer < 0 (%i)\n", freeBuffers);
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freeBuffers = 0;
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}
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// Need to Cool Down Speaker -- Soft Fade to Zero
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else {
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spkr_fade(spkr_level, 0);
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#ifdef SPKR_DEBUG
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spkr_debug(spkr_debug_raw_file);
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spkr_debug(spkr_debug_ema_file);
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#endif
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alBufferData(spkr_buffers[freeBuffers], AL_FORMAT_STEREO16, spkr_samples, spkr_buf_size * sizeof(spkr_sample_t), spkr_sample_rate);
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al_check_error();
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// // DEBUG ONLY!!!
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// fwrite(spkr_samples + spkr_buf_size * (BUFFER_COUNT - 2), sizeof(spkr_sample_t), spkr_buf_size, af);
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// fflush(af);
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alSourceQueueBuffers(spkr_src[SPKR_SRC_GAME_SFX], 1, &spkr_buffers[freeBuffers]);
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al_check_error();
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}
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memset(spkr_samples, 0, sample_buf_array_len * sizeof(spkr_sample_t)); // spkr_buf_size * sizeof(spkr_sample_t) * SPKR_CHANNELS);
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spkr_sample_idx = 0;
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spkr_sample_last_idx = 0;
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spkr_filter_reset();
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}
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// Need to Cool Down Speaker -- Soft Fade to Zero
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else {
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spkr_fade(spkr_level, 0);
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#ifdef SPKR_DEBUG
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fwrite(spkr_samples, sizeof(spkr_sample_t), spkr_buf_size, spkr_debug_file);
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fflush(spkr_debug_file);
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#endif
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// push a click into the speaker buffer
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// (we will play the entire buffer at the end of the frame)
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// spkr_sample_idx = ( (spkr_clk + m6502.clkfrm) / ( MHZ(default_MHz_6502) / spkr_sample_rate)) * SPKR_CHANNELS;
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alBufferData(spkr_buffers[freeBuffers], AL_FORMAT_STEREO16, spkr_samples, spkr_buf_size * sizeof(spkr_sample_t), spkr_sample_rate);
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al_check_error();
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// spkr_debug_spike();
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// // DEBUG ONLY!!!
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// fwrite(spkr_samples + spkr_buf_size * (BUFFER_COUNT - 2), sizeof(spkr_sample_t), spkr_buf_size, af);
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// fflush(af);
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alSourceQueueBuffers(spkr_src[SPKR_SRC_GAME_SFX], 1, &spkr_buffers[freeBuffers]);
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al_check_error();
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// //spkr_samples[sample_idx] = spkr_level;
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// memset(spkr_samples + spkr_sample_idx, spkr_level, spkr_buf_alloc_size * DEFAULT_FPS - spkr_sample_idx);
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if (--freeBuffers < 0) {
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printf("freeBuffer < 0 (%i)\n", freeBuffers);
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freeBuffers = 0;
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}
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// Normal Sound Buffer Feed
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else {
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// finish the aquare wave
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while ( spkr_sample_last_idx < spkr_buf_size ) {
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spkr_samples[ spkr_sample_last_idx++ ] = spkr_level;
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spkr_samples[ spkr_sample_last_idx++ ] = spkr_level; // stereo
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}
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int buf_len = round((double)spkr_buf_size + spkr_extra_buf) * sizeof(spkr_sample_t);
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// digital filtering the audio stream -- most notably smoothing
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spkr_filter(buf_len);
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alBufferData(spkr_buffers[freeBuffers], AL_FORMAT_STEREO16, spkr_samples, buf_len, spkr_sample_rate);
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al_check_error();
|
||||
alSourceQueueBuffers(spkr_src[SPKR_SRC_GAME_SFX], 1, &spkr_buffers[freeBuffers]);
|
||||
al_check_error();
|
||||
}
|
||||
}
|
||||
|
||||
memset(spkr_samples, 0, sample_buf_array_len * sizeof(spkr_sample_t)); // spkr_buf_size * sizeof(spkr_sample_t) * SPKR_CHANNELS);
|
||||
ALenum state;
|
||||
alGetSourcei( spkr_src[SPKR_SRC_GAME_SFX], AL_SOURCE_STATE, &state );
|
||||
// al_check_error();
|
||||
|
||||
switch (state) {
|
||||
case AL_PAUSED:
|
||||
if ( --playDelay < 0 ) {
|
||||
alSourcePlay(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
playDelay = SPKR_PLAY_DELAY;
|
||||
}
|
||||
break;
|
||||
|
||||
case AL_PLAYING:
|
||||
// already playing
|
||||
break;
|
||||
|
||||
case AL_INITIAL:
|
||||
case AL_STOPPED:
|
||||
default:
|
||||
alSourcePlay(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
// this is so we will set state to AL_PAUSED immediately
|
||||
// As a result there will be an extra queued buffer
|
||||
// which gives us a glitch free sound
|
||||
alSourcePause(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
playDelay = SPKR_PLAY_DELAY;
|
||||
break;
|
||||
}
|
||||
|
||||
// int dst = 0;
|
||||
int src = spkr_buf_size + spkr_extra_buf;
|
||||
// int size = max(0, (int)spkr_sample_idx - src);
|
||||
|
||||
// if (size) {
|
||||
memcpy(spkr_samples, spkr_samples + src, spkr_buf_size * sizeof(spkr_sample_t) );
|
||||
// }
|
||||
|
||||
memset(spkr_samples + src, 0, spkr_buf_size * sizeof(spkr_sample_t));
|
||||
|
||||
spkr_sample_idx = 0;
|
||||
spkr_sample_last_idx = 0;
|
||||
|
||||
filter_reset();
|
||||
|
||||
}
|
||||
else {
|
||||
// push a click into the speaker buffer
|
||||
// (we will play the entire buffer at the end of the frame)
|
||||
// spkr_sample_idx = ( (spkr_clk + m6502.clkfrm) / ( MHZ(default_MHz_6502) / spkr_sample_rate)) * SPKR_CHANNELS;
|
||||
|
||||
// debug_spike();
|
||||
|
||||
// //spkr_samples[sample_idx] = spkr_level;
|
||||
// memset(spkr_samples + spkr_sample_idx, spkr_level, spkr_buf_alloc_size * DEFAULT_FPS - spkr_sample_idx);
|
||||
|
||||
if (--freeBuffers < 0) {
|
||||
printf("freeBuffer < 0 (%i)\n", freeBuffers);
|
||||
freeBuffers = 0;
|
||||
}
|
||||
|
||||
// Normal Sound Buffer Feed
|
||||
else {
|
||||
// finish the aquare wave
|
||||
while ( spkr_sample_last_idx < spkr_buf_size ) {
|
||||
spkr_samples[ spkr_sample_last_idx++ ] = spkr_level;
|
||||
spkr_samples[ spkr_sample_last_idx++ ] = spkr_level; // stereo
|
||||
}
|
||||
|
||||
|
||||
int buf_len = round((double)spkr_buf_size * multiplier + spkr_extra_buf) * sizeof(spkr_sample_t);
|
||||
|
||||
// digital filtering the audio stream -- most notably smoothing
|
||||
filter();
|
||||
|
||||
alBufferData(spkr_buffers[freeBuffers], AL_FORMAT_STEREO16, spkr_samples, buf_len, spkr_sample_rate * multiplier);
|
||||
al_check_error();
|
||||
alSourceQueueBuffers(spkr_src[SPKR_SRC_GAME_SFX], 1, &spkr_buffers[freeBuffers]);
|
||||
al_check_error();
|
||||
}
|
||||
printf("Warning: No FreeBuffers!\n");
|
||||
}
|
||||
|
||||
ALenum state;
|
||||
alGetSourcei( spkr_src[SPKR_SRC_GAME_SFX], AL_SOURCE_STATE, &state );
|
||||
// al_check_error();
|
||||
|
||||
switch (state) {
|
||||
case AL_PAUSED:
|
||||
if ( --playDelay <= 0 ) {
|
||||
alSourcePlay(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
playDelay = SPKR_PLAY_DELAY;
|
||||
}
|
||||
break;
|
||||
|
||||
case AL_PLAYING:
|
||||
// already playing
|
||||
break;
|
||||
|
||||
case AL_INITIAL:
|
||||
case AL_STOPPED:
|
||||
default:
|
||||
alSourcePlay(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
// this is so we will set state to AL_PAUSED immediately
|
||||
// As a result there will be an extra queued buffer
|
||||
// which gives us a glitch free sound
|
||||
alSourcePause(spkr_src[SPKR_SRC_GAME_SFX]);
|
||||
break;
|
||||
}
|
||||
|
||||
// int dst = 0;
|
||||
int src = spkr_buf_size * multiplier + spkr_extra_buf;
|
||||
int size = max(0, (int)spkr_sample_idx - src);
|
||||
|
||||
if (size) {
|
||||
memcpy(spkr_samples, spkr_samples + src * sizeof(spkr_sample_t), size * sizeof(spkr_sample_t) * SPKR_CHANNELS );
|
||||
}
|
||||
|
||||
memset(spkr_samples + size, 0, spkr_buf_size * sizeof(spkr_sample_t));
|
||||
|
||||
spkr_sample_idx = 0;
|
||||
spkr_sample_last_idx = 0;
|
||||
|
||||
}
|
||||
else {
|
||||
printf("Warning: No FreeBuffers!\n");
|
||||
}
|
||||
|
||||
// else {
|
||||
// printf("spkr_play_time: %u, queued: %i\n", spkr_play_time, queued);
|
||||
// }
|
||||
}
|
||||
// else {
|
||||
// printf("spkr_play_time: %u, queued: %i\n", spkr_play_time, queued);
|
||||
// }
|
||||
|
||||
spkr_clk = 0;
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue
Block a user