diff options
Diffstat (limited to 'src/devices/sound/saa1099.cpp')
-rw-r--r-- | src/devices/sound/saa1099.cpp | 66 |
1 files changed, 25 insertions, 41 deletions
diff --git a/src/devices/sound/saa1099.cpp b/src/devices/sound/saa1099.cpp index 9c41ca7a939..9edec91113a 100644 --- a/src/devices/sound/saa1099.cpp +++ b/src/devices/sound/saa1099.cpp @@ -70,8 +70,10 @@ #include "emu.h" #include "saa1099.h" -#define LEFT 0x00 -#define RIGHT 0x01 +static constexpr int clock_divider = 256; + +static constexpr int LEFT = 0x00; +static constexpr int RIGHT = 0x01; static constexpr u16 amplitude_lookup[16] = { 0*32767/16, 1*32767/16, 2*32767/16, 3*32767/16, @@ -123,7 +125,6 @@ static constexpr u8 envelope[8][64] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,10,11,12,13,14,15 } }; - // device type definition DEFINE_DEVICE_TYPE(SAA1099, saa1099_device, "saa1099", "Philips SAA1099") @@ -149,7 +150,6 @@ saa1099_device::saa1099_device(const machine_config &mconfig, const char *tag, d , m_all_ch_enable(false) , m_sync_state(false) , m_selected_reg(0) - , m_sample_rate(0.0) { } @@ -160,11 +160,8 @@ saa1099_device::saa1099_device(const machine_config &mconfig, const char *tag, d void saa1099_device::device_start() { - /* copy global parameters */ - m_sample_rate = clock() / 256; - /* for each chip allocate one stream */ - m_stream = stream_alloc(0, 2, m_sample_rate); + m_stream = stream_alloc(0, 2, clock()/clock_divider); save_item(NAME(m_noise_params)); save_item(NAME(m_env_enable)); @@ -184,7 +181,6 @@ void saa1099_device::device_start() save_item(STRUCT_MEMBER(m_channels, amplitude)); save_item(STRUCT_MEMBER(m_channels, envelope)); save_item(STRUCT_MEMBER(m_channels, counter)); - save_item(STRUCT_MEMBER(m_channels, freq)); save_item(STRUCT_MEMBER(m_channels, level)); save_item(STRUCT_MEMBER(m_noise, counter)); @@ -199,9 +195,7 @@ void saa1099_device::device_start() void saa1099_device::device_clock_changed() { - m_sample_rate = clock() / 256; - - m_stream->set_sample_rate(m_sample_rate); + m_stream->set_sample_rate(clock()/clock_divider); } @@ -225,10 +219,10 @@ void saa1099_device::sound_stream_update(sound_stream &stream, stream_sample_t * { switch (m_noise_params[ch]) { - case 0: m_noise[ch].freq = clock()/256.0 * 2; break; - case 1: m_noise[ch].freq = clock()/512.0 * 2; break; - case 2: m_noise[ch].freq = clock()/1024.0 * 2; break; - case 3: m_noise[ch].freq = m_channels[ch * 3].freq; break; // todo: this case will be clock()/[ch*3's octave divisor, 0 is = 256*2, higher numbers are higher] * 2 if the tone generator phase reset bit (0x1c bit 1) is set. + case 0: + case 1: + case 2: m_noise[ch].freq = 256 << m_noise_params[ch]; break; + case 3: m_noise[ch].freq = m_channels[ch * 3].freq(); break; // todo: this case will be clock()/[ch*3's octave divisor, 0 is = 256*2, higher numbers are higher] * 2 if the tone generator phase reset bit (0x1c bit 1) is set. } } @@ -240,19 +234,11 @@ void saa1099_device::sound_stream_update(sound_stream &stream, stream_sample_t * /* for each channel */ for (ch = 0; ch < 6; ch++) { - if (m_channels[ch].freq == 0.0) - m_channels[ch].freq = (double)((2 * clock() / 512) << m_channels[ch].octave) / - (511.0 - (double)m_channels[ch].frequency); - /* check the actual position in the square wave */ - m_channels[ch].counter -= m_channels[ch].freq; - while (m_channels[ch].counter < 0) + while (m_channels[ch].counter <= 0) { /* calculate new frequency now after the half wave is updated */ - m_channels[ch].freq = (double)((2 * clock() / 512) << m_channels[ch].octave) / - (511.0 - (double)m_channels[ch].frequency); - - m_channels[ch].counter += m_sample_rate; + m_channels[ch].counter += m_channels[ch].freq(); m_channels[ch].level ^= 1; /* eventually clock the envelope counters */ @@ -261,27 +247,25 @@ void saa1099_device::sound_stream_update(sound_stream &stream, stream_sample_t * if (ch == 4 && m_env_clock[1] == 0) envelope_w(1); } + m_channels[ch].counter -= clock_divider; // if the noise is enabled + u8 level = 0; if (m_channels[ch].noise_enable) { // if the noise level is high (noise 0: chan 0-2, noise 1: chan 3-5) - if (m_noise[ch/3].level & 1) - { - // subtract to avoid overflows, also use only half amplitude - output_l -= m_channels[ch].amplitude[ LEFT] * m_channels[ch].envelope[ LEFT] / 16 / 2; - output_r -= m_channels[ch].amplitude[RIGHT] * m_channels[ch].envelope[RIGHT] / 16 / 2; - } + level ^= m_noise[ch/3].level & 1; } // if the square wave is enabled if (m_channels[ch].freq_enable) { // if the channel level is high - if (m_channels[ch].level & 1) - { - output_l += m_channels[ch].amplitude[ LEFT] * m_channels[ch].envelope[ LEFT] / 16; - output_r += m_channels[ch].amplitude[RIGHT] * m_channels[ch].envelope[RIGHT] / 16; - } + level ^= m_channels[ch].level & 1; + } + if (level) + { + output_l += m_channels[ch].amplitude[ LEFT] * m_channels[ch].envelope[ LEFT] / 16; + output_r += m_channels[ch].amplitude[RIGHT] * m_channels[ch].envelope[RIGHT] / 16; } } @@ -290,15 +274,15 @@ void saa1099_device::sound_stream_update(sound_stream &stream, stream_sample_t * /* update the state of the noise generator * polynomial is x^18 + x^11 + x (i.e. 0x20400) and is a plain XOR, initial state is probably all 1s * see http://www.vogons.org/viewtopic.php?f=9&t=51695 */ - m_noise[ch].counter -= m_noise[ch].freq; - while (m_noise[ch].counter < 0) + while (m_noise[ch].counter <= 0) { - m_noise[ch].counter += m_sample_rate; + m_noise[ch].counter += m_noise[ch].freq; // clock / ((511 - frequency) * 2^(8 - octave)) or clock / 2^(8 + noise period) if( ((m_noise[ch].level & 0x20000) == 0) != ((m_noise[ch].level & 0x0400) == 0) ) m_noise[ch].level = (m_noise[ch].level << 1) | 1; else m_noise[ch].level <<= 1; } + m_noise[ch].counter -= clock_divider; } /* write sound data to the buffer */ outputs[LEFT][j] = output_l / 6; @@ -434,7 +418,7 @@ void saa1099_device::data_w(u8 data) for (int i = 0; i < 6; i++) { m_channels[i].level = 0; - m_channels[i].counter = 0.0; + m_channels[i].counter = m_channels[i].freq(); } } break; |