diff options
42 files changed, 239 insertions, 249 deletions
diff --git a/src/devices/cpu/tms57002/tms57002.cpp b/src/devices/cpu/tms57002/tms57002.cpp index dcc6e615cea..c2cf28f3f92 100644 --- a/src/devices/cpu/tms57002/tms57002.cpp +++ b/src/devices/cpu/tms57002/tms57002.cpp @@ -924,16 +924,16 @@ void tms57002_device::sound_stream_update(sound_stream &stream, std::vector<read assert(outputs[0].samples() == 1); stream_buffer::sample_t in_scale = 32768.0 * ((st0 & ST0_SIM) ? 256.0 : 1.0); - si[0] = s32(inputs[0].get() * in_scale) & 0xffffff; - si[1] = s32(inputs[1].get() * in_scale) & 0xffffff; - si[2] = s32(inputs[2].get() * in_scale) & 0xffffff; - si[3] = s32(inputs[3].get() * in_scale) & 0xffffff; + si[0] = s32(inputs[0].get(0) * in_scale) & 0xffffff; + si[1] = s32(inputs[1].get(0) * in_scale) & 0xffffff; + si[2] = s32(inputs[2].get(0) * in_scale) & 0xffffff; + si[3] = s32(inputs[3].get(0) * in_scale) & 0xffffff; stream_buffer::sample_t out_scale = 1.0 / (32768.0 * 65536.0); - outputs[0].put(stream_buffer::sample_t(s32(so[0] << 8)) * out_scale); - outputs[1].put(stream_buffer::sample_t(s32(so[1] << 8)) * out_scale); - outputs[2].put(stream_buffer::sample_t(s32(so[2] << 8)) * out_scale); - outputs[3].put(stream_buffer::sample_t(s32(so[3] << 8)) * out_scale); + outputs[0].put(0, stream_buffer::sample_t(s32(so[0] << 8)) * out_scale); + outputs[1].put(0, stream_buffer::sample_t(s32(so[1] << 8)) * out_scale); + outputs[2].put(0, stream_buffer::sample_t(s32(so[2] << 8)) * out_scale); + outputs[3].put(0, stream_buffer::sample_t(s32(so[3] << 8)) * out_scale); sync_w(1); } diff --git a/src/devices/machine/netlist.cpp b/src/devices/machine/netlist.cpp index 06e574890d2..988c21c33e1 100644 --- a/src/devices/machine/netlist.cpp +++ b/src/devices/machine/netlist.cpp @@ -825,10 +825,10 @@ void netlist_mame_stream_output_device::sound_update_fill(write_stream_view &tar if (target.samples() < m_buffer.size()) osd_printf_warning("sound %s: samples %d less bufsize %d\n", name(), target.samples(), m_buffer.size()); - int sampindex = 0; - while (!target.done() && sampindex < m_buffer.size()) - target.put(m_buffer[sampindex++]); - target.fill(m_cur); + int sampindex; + for (sampindex = 0; sampindex < m_buffer.size(); sampindex++) + target.put(sampindex, m_buffer[sampindex]); + target.fill(m_cur, sampindex); } diff --git a/src/devices/machine/netlist.h b/src/devices/machine/netlist.h index 269f98e415c..8b9e156adba 100644 --- a/src/devices/machine/netlist.h +++ b/src/devices/machine/netlist.h @@ -241,7 +241,7 @@ public: netlist_mame_sound_input_buffer(read_stream_view const &src) : read_stream_view(src) { } - stream_buffer::sample_t operator[](std::size_t index) { return get_indexed(index); } + stream_buffer::sample_t operator[](std::size_t index) { return get(index); } }; // ---------------------------------------------------------------------------------------- diff --git a/src/devices/sound/aica.cpp b/src/devices/sound/aica.cpp index a55199cc3fc..7bedf5d660a 100644 --- a/src/devices/sound/aica.cpp +++ b/src/devices/sound/aica.cpp @@ -1234,7 +1234,7 @@ void aica_device::DoMasterSamples(std::vector<read_stream_view> const &inputs, w int i; constexpr stream_buffer::sample_t sample_scale = 1.0 / stream_buffer::sample_t(32768 << SHIFT); - while (!bufl.done()) + for (int s = 0; s < bufl.samples(); ++s) { s32 smpl = 0, smpr = 0; @@ -1275,7 +1275,7 @@ void aica_device::DoMasterSamples(std::vector<read_stream_view> const &inputs, w { if (EFSDL(i + 16)) // 16,17 for EXTS { - m_DSP.EXTS[i] = s16(inputs[i].get() * 32767.0); + m_DSP.EXTS[i] = s16(inputs[i].get(s) * 32767.0); u32 Enc = ((EFPAN(i + 16)) << 0x8) | ((EFSDL(i + 16)) << 0xd); smpl += (m_DSP.EXTS[i] * m_LPANTABLE[Enc]) >> SHIFT; smpr += (m_DSP.EXTS[i] * m_RPANTABLE[Enc]) >> SHIFT; @@ -1293,8 +1293,8 @@ void aica_device::DoMasterSamples(std::vector<read_stream_view> const &inputs, w smpr = clip16(smpr >> 3); } - bufl.put(stream_buffer::sample_t(smpl * m_LPANTABLE[MVOL() << 0xd]) * sample_scale); - bufr.put(stream_buffer::sample_t(smpr * m_LPANTABLE[MVOL() << 0xd]) * sample_scale); + bufl.put(s, stream_buffer::sample_t(smpl * m_LPANTABLE[MVOL() << 0xd]) * sample_scale); + bufr.put(s, stream_buffer::sample_t(smpr * m_LPANTABLE[MVOL() << 0xd]) * sample_scale); } } diff --git a/src/devices/sound/asc.cpp b/src/devices/sound/asc.cpp index c7e467b617c..e03494040ab 100644 --- a/src/devices/sound/asc.cpp +++ b/src/devices/sound/asc.cpp @@ -119,7 +119,7 @@ void asc_device::device_timer(emu_timer &timer, device_timer_id tid, int param, void asc_device::sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) { - int ch; + int i, ch; static uint32_t wtoffs[2] = { 0, 0x200 }; auto &outL = outputs[0]; @@ -149,7 +149,7 @@ void asc_device::sound_stream_update(sound_stream &stream, std::vector<read_stre case 1: // FIFO mode { constexpr stream_buffer::sample_t sample_scale = 64.0 / 32768.0; - while (!outL.done()) + for (i = 0; i < outL.samples(); i++) { int8_t smpll, smplr; @@ -234,8 +234,8 @@ void asc_device::sound_stream_update(sound_stream &stream, std::vector<read_stre break; } - outL.put(stream_buffer::sample_t(smpll) * sample_scale); - outR.put(stream_buffer::sample_t(smplr) * sample_scale); + outL.put(i, stream_buffer::sample_t(smpll) * sample_scale); + outR.put(i, stream_buffer::sample_t(smplr) * sample_scale); } break; } @@ -243,7 +243,7 @@ void asc_device::sound_stream_update(sound_stream &stream, std::vector<read_stre case 2: // wavetable mode { constexpr stream_buffer::sample_t sample_scale = 1.0 / (32768.0 * 4.0); - while (!outL.done()) + for (i = 0; i < outL.samples(); i++) { int32_t mixL, mixR; int8_t smpl; @@ -269,8 +269,8 @@ void asc_device::sound_stream_update(sound_stream &stream, std::vector<read_stre mixR += smpl*256; } - outL.put(stream_buffer::sample_t(mixL) * sample_scale); - outR.put(stream_buffer::sample_t(mixR) * sample_scale); + outL.put(i, stream_buffer::sample_t(mixL) * sample_scale); + outR.put(i, stream_buffer::sample_t(mixR) * sample_scale); } break; } diff --git a/src/devices/sound/astrocde.cpp b/src/devices/sound/astrocde.cpp index fe39bd4ac32..17df6999989 100644 --- a/src/devices/sound/astrocde.cpp +++ b/src/devices/sound/astrocde.cpp @@ -144,13 +144,14 @@ void astrocade_io_device::sound_stream_update(sound_stream &stream, std::vector< /* loop over samples */ int samples_this_time; constexpr stream_buffer::sample_t sample_scale = 1.0f / 60.0f; - while (!dest.done()) + for (int sampindex = 0; sampindex < dest.samples(); sampindex += samples_this_time) { stream_sample_t cursample = 0; /* compute the number of cycles until the next master oscillator reset */ /* or until the next noise boundary */ - samples_this_time = std::min(256 - master_count, 64 - noise_clock); + samples_this_time = std::min<int>(dest.samples() - sampindex, 256 - master_count); + samples_this_time = std::min(samples_this_time, 64 - noise_clock); /* sum the output of the tone generators */ if (m_a_state) @@ -165,7 +166,7 @@ void astrocade_io_device::sound_stream_update(sound_stream &stream, std::vector< cursample += m_reg[7] >> 4; /* scale to max and output */ - samples_this_time = dest.fill(stream_buffer::sample_t(cursample) * sample_scale, samples_this_time); + dest.fill(stream_buffer::sample_t(cursample) * sample_scale, sampindex, samples_this_time); /* clock the noise; a 2-bit counter clocks a 4-bit counter which clocks the LFSR */ noise_clock += samples_this_time; diff --git a/src/devices/sound/awacs.cpp b/src/devices/sound/awacs.cpp index 007217db048..19f734cd468 100644 --- a/src/devices/sound/awacs.cpp +++ b/src/devices/sound/awacs.cpp @@ -96,10 +96,10 @@ void awacs_device::sound_stream_update(sound_stream &stream, std::vector<read_st constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; if (m_playback_enable) { - while (!outL.done()) + for (int i = 0; i < outL.samples(); i++) { - outL.put(stream_buffer::sample_t(s16(m_dma_space->read_word(offset + m_play_ptr))) * sample_scale); - outR.put(stream_buffer::sample_t(s16(m_dma_space->read_word(offset + m_play_ptr + 2))) * sample_scale); + outL.put(i, stream_buffer::sample_t(s16(m_dma_space->read_word(offset + m_play_ptr))) * sample_scale); + outR.put(i, stream_buffer::sample_t(s16(m_dma_space->read_word(offset + m_play_ptr + 2))) * sample_scale); m_play_ptr += 4; } diff --git a/src/devices/sound/ay8910.cpp b/src/devices/sound/ay8910.cpp index 0152d83620a..26e99cf1bc7 100644 --- a/src/devices/sound/ay8910.cpp +++ b/src/devices/sound/ay8910.cpp @@ -1086,10 +1086,12 @@ void ay8910_device::sound_stream_update(sound_stream &stream, std::vector<read_s tone_t *tone; envelope_t *envelope; + int samples = outputs[0].samples(); + /* hack to prevent us from hanging when starting filtered outputs */ if (!m_ready) { - for (int chan = 0; chan < outputs.size(); chan++) + for (int chan = 0; chan < m_streams; chan++) outputs[chan].fill(0); } @@ -1101,7 +1103,7 @@ void ay8910_device::sound_stream_update(sound_stream &stream, std::vector<read_s /* is 1, not 0, and can be modulated changing the volume. */ /* buffering loop */ - while (!outputs[0].done()) + for (int sampindex = 0; sampindex < samples; sampindex++) { for (int chan = 0; chan < NUM_CHANNELS; chan++) { @@ -1197,38 +1199,38 @@ void ay8910_device::sound_stream_update(sound_stream &stream, std::vector<read_s { env_volume >>= 1; if (m_feature & PSG_EXTENDED_ENVELOPE) // AY8914 Has a two bit tone_envelope field - outputs[chan].put(m_vol_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); + outputs[chan].put(sampindex, m_vol_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); else - outputs[chan].put(m_vol_table[chan][m_vol_enabled[chan] ? env_volume : 0]); + outputs[chan].put(sampindex, m_vol_table[chan][m_vol_enabled[chan] ? env_volume : 0]); } else { if (m_feature & PSG_EXTENDED_ENVELOPE) // AY8914 Has a two bit tone_envelope field - outputs[chan].put(m_env_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); + outputs[chan].put(sampindex, m_env_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); else - outputs[chan].put(m_env_table[chan][m_vol_enabled[chan] ? env_volume : 0]); + outputs[chan].put(sampindex, m_env_table[chan][m_vol_enabled[chan] ? env_volume : 0]); } } else { if (m_feature & PSG_EXTENDED_ENVELOPE) // AY8914 Has a two bit tone_envelope field - outputs[chan].put(m_env_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); + outputs[chan].put(sampindex, m_env_table[chan][m_vol_enabled[chan] ? env_volume >> (3-tone_envelope(tone)) : 0]); else - outputs[chan].put(m_env_table[chan][m_vol_enabled[chan] ? env_volume : 0]); + outputs[chan].put(sampindex, m_env_table[chan][m_vol_enabled[chan] ? env_volume : 0]); } } else { if (is_expanded_mode()) - outputs[chan].put(m_env_table[chan][m_vol_enabled[chan] ? tone_volume(tone) : 0]); + outputs[chan].put(sampindex, m_env_table[chan][m_vol_enabled[chan] ? tone_volume(tone) : 0]); else - outputs[chan].put(m_vol_table[chan][m_vol_enabled[chan] ? tone_volume(tone) : 0]); + outputs[chan].put(sampindex, m_vol_table[chan][m_vol_enabled[chan] ? tone_volume(tone) : 0]); } } } else { - outputs[0].put(mix_3D()); + outputs[0].put(sampindex, mix_3D()); } } } diff --git a/src/devices/sound/beep.cpp b/src/devices/sound/beep.cpp index a9cfd6747cc..58096c4b561 100644 --- a/src/devices/sound/beep.cpp +++ b/src/devices/sound/beep.cpp @@ -83,9 +83,9 @@ void beep_device::sound_stream_update(sound_stream &stream, std::vector<read_str } /* fill in the sample */ - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { - buffer.put(signal); + buffer.put(sampindex, signal); incr -= clock; while( incr < 0 ) { diff --git a/src/devices/sound/c140.cpp b/src/devices/sound/c140.cpp index 0227fd47d08..0edb1920a8c 100644 --- a/src/devices/sound/c140.cpp +++ b/src/devices/sound/c140.cpp @@ -327,14 +327,14 @@ void c140_device::sound_stream_update(sound_stream &stream, std::vector<read_str auto &dest1 = outputs[0]; auto &dest2 = outputs[1]; constexpr stream_buffer::sample_t sample_scale = 8.0 / 32768.0; - while (!dest1.done()) + for (int i = 0; i < samples; i++) { stream_buffer::sample_t val; val = stream_buffer::sample_t(*lmix++) * sample_scale; - dest1.put(limit(val)); + dest1.put(i, limit(val)); val = stream_buffer::sample_t(*rmix++) * sample_scale; - dest2.put(limit(val)); + dest2.put(i, limit(val)); } } } @@ -466,14 +466,14 @@ void c219_device::sound_stream_update(sound_stream &stream, std::vector<read_str auto &dest1 = outputs[0]; auto &dest2 = outputs[1]; constexpr stream_buffer::sample_t sample_scale = 8.0 / 32768.0; - while (!dest1.done()) + for (int i = 0; i < samples; i++) { stream_buffer::sample_t val; val = stream_buffer::sample_t(*lmix++) * sample_scale; - dest1.put(limit(val)); + dest1.put(i, limit(val)); val = stream_buffer::sample_t(*rmix++) * sample_scale; - dest2.put(limit(val)); + dest2.put(i, limit(val)); } } } diff --git a/src/devices/sound/c352.cpp b/src/devices/sound/c352.cpp index f2b7dd27506..826f194437a 100644 --- a/src/devices/sound/c352.cpp +++ b/src/devices/sound/c352.cpp @@ -130,7 +130,7 @@ void c352_device::sound_stream_update(sound_stream &stream, std::vector<read_str auto &buffer_rr = outputs[3]; constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!buffer_fl.done()) + for (int i = 0; i < buffer_fl.samples(); i++) { int out[4] = { 0, 0, 0, 0 }; @@ -174,10 +174,10 @@ void c352_device::sound_stream_update(sound_stream &stream, std::vector<read_str out[3] += (((v.flags & C352_FLG_PHASEFR) ? -s : s) * v.curr_vol[3]) >> 8; } - buffer_fl.put(stream_buffer::sample_t(s16(out[0] >> 3)) * sample_scale); - buffer_fr.put(stream_buffer::sample_t(s16(out[1] >> 3)) * sample_scale); - buffer_rl.put(stream_buffer::sample_t(s16(out[2] >> 3)) * sample_scale); - buffer_rr.put(stream_buffer::sample_t(s16(out[3] >> 3)) * sample_scale); + buffer_fl.put(i, stream_buffer::sample_t(s16(out[0] >> 3)) * sample_scale); + buffer_fr.put(i, stream_buffer::sample_t(s16(out[1] >> 3)) * sample_scale); + buffer_rl.put(i, stream_buffer::sample_t(s16(out[2] >> 3)) * sample_scale); + buffer_rr.put(i, stream_buffer::sample_t(s16(out[3] >> 3)) * sample_scale); } } diff --git a/src/devices/sound/c6280.cpp b/src/devices/sound/c6280.cpp index e8de8151be6..d01dd74125a 100644 --- a/src/devices/sound/c6280.cpp +++ b/src/devices/sound/c6280.cpp @@ -55,10 +55,6 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st for (int ch = 0; ch < 6; ch++) { channel *chan = &m_channel[ch]; - - outputs[0].reset(); - outputs[1].reset(); - /* Only look at enabled channels */ if (chan->control & 0x80) { @@ -81,7 +77,7 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st { /* Noise mode */ const u32 step = (chan->noise_control & 0x1f) ^ 0x1f; - while (!outputs[0].done()) + for (int i = 0; i < outputs[0].samples(); i += 1) { s16 data = BIT(chan->noise_seed, 0) ? 0x1f : 0; chan->noise_counter--; @@ -92,18 +88,18 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st // based on Charles MacDonald's research chan->noise_seed = (seed >> 1) | ((BIT(seed, 0) ^ BIT(seed, 1) ^ BIT(seed, 11) ^ BIT(seed, 12) ^ BIT(seed, 17)) << 17); } - outputs[0].add(stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); - outputs[1].add(stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); + outputs[0].add(i, stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); + outputs[1].add(i, stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); } } else if (chan->control & 0x40) { /* DDA mode */ - while (!outputs[0].done()) + for (int i = 0; i < outputs[0].samples(); i++) { - outputs[0].add(stream_buffer::sample_t(s16(vll * (chan->dda - 16))) * sample_scale); - outputs[1].add(stream_buffer::sample_t(s16(vlr * (chan->dda - 16))) * sample_scale); + outputs[0].add(i, stream_buffer::sample_t(s16(vll * (chan->dda - 16))) * sample_scale); + outputs[1].add(i, stream_buffer::sample_t(s16(vlr * (chan->dda - 16))) * sample_scale); } } else @@ -116,7 +112,7 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st channel *lfo_srcchan = &m_channel[1]; channel *lfo_dstchan = &m_channel[0]; const u16 lfo_step = lfo_srcchan->frequency ? lfo_srcchan->frequency : 0x1000; - while (!outputs[0].done()) + for (int i = 0; i < outputs[0].samples(); i += 1) { s32 step = lfo_dstchan->frequency ? lfo_dstchan->frequency : 0x1000; if (m_lfo_control & 0x80) // reset LFO @@ -142,8 +138,8 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st lfo_dstchan->tick = step; lfo_dstchan->index = (lfo_dstchan->index + 1) & 0x1f; } - outputs[0].add(stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); - outputs[1].add(stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); + outputs[0].add(i, stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); + outputs[1].add(i, stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); } } } @@ -151,7 +147,7 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st { /* Waveform mode */ const u32 step = chan->frequency ? chan->frequency : 0x1000; - while (!outputs[0].done()) + for (int i = 0; i < outputs[0].samples(); i += 1) { const s16 data = chan->waveform[chan->index]; chan->tick--; @@ -160,8 +156,8 @@ void c6280_device::sound_stream_update(sound_stream &stream, std::vector<read_st chan->tick = step; chan->index = (chan->index + 1) & 0x1f; } - outputs[0].add(stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); - outputs[1].add(stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); + outputs[0].add(i, stream_buffer::sample_t(s16(vll * (data - 16))) * sample_scale); + outputs[1].add(i, stream_buffer::sample_t(s16(vlr * (data - 16))) * sample_scale); } } } diff --git a/src/devices/sound/cdda.cpp b/src/devices/sound/cdda.cpp index 4cecdc95ee7..ca2bd280de8 100644 --- a/src/devices/sound/cdda.cpp +++ b/src/devices/sound/cdda.cpp @@ -18,8 +18,8 @@ void cdda_device::sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) { get_audio_data(outputs[0], outputs[1]); - m_audio_volume[0] = outputs[0].get_indexed(0); - m_audio_volume[1] = outputs[1].get_indexed(0); + m_audio_volume[0] = outputs[0].get(0); + m_audio_volume[1] = outputs[1].get(0); } //------------------------------------------------- @@ -165,7 +165,7 @@ void cdda_device::get_audio_data(write_stream_view &bufL, write_stream_view &buf int16_t *audio_cache = (int16_t *) m_audio_cache.get(); constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!bufL.done()) + for (int sampindex = 0; sampindex < bufL.samples(); ) { /* if no file, audio not playing, audio paused, or out of disc data, just zero fill */ @@ -177,12 +177,12 @@ void cdda_device::get_audio_data(write_stream_view &bufL, write_stream_view &buf m_audio_ended_normally = true; } - bufL.fill(0); - bufR.fill(0); + bufL.fill(0, sampindex); + bufR.fill(0, sampindex); return; } - int samples = bufL.remaining(); + int samples = bufL.samples() - sampindex; if (samples > m_audio_samples) { samples = m_audio_samples; @@ -191,10 +191,11 @@ void cdda_device::get_audio_data(write_stream_view &bufL, write_stream_view &buf for (i = 0; i < samples; i++) { /* CD-DA data on the disc is big-endian */ - bufL.put(stream_buffer::sample_t(s16(big_endianize_int16( audio_cache[ m_audio_bptr ] ))) * sample_scale); m_audio_bptr++; - bufR.put(stream_buffer::sample_t(s16(big_endianize_int16( audio_cache[ m_audio_bptr ] ))) * sample_scale); m_audio_bptr++; + bufL.put(sampindex + i, stream_buffer::sample_t(s16(big_endianize_int16( audio_cache[ m_audio_bptr ] ))) * sample_scale); m_audio_bptr++; + bufR.put(sampindex + i, stream_buffer::sample_t(s16(big_endianize_int16( audio_cache[ m_audio_bptr ] ))) * sample_scale); m_audio_bptr++; } + sampindex += samples; m_audio_samples -= samples; if (m_audio_samples == 0) diff --git a/src/devices/sound/cdp1863.cpp b/src/devices/sound/cdp1863.cpp index fd84ff0cd75..984a306b28a 100644 --- a/src/devices/sound/cdp1863.cpp +++ b/src/devices/sound/cdp1863.cpp @@ -114,9 +114,9 @@ void cdp1863_device::sound_stream_update(sound_stream &stream, std::vector<read_ signal = 1.0; } - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { - buffer.put(signal); + buffer.put(sampindex, signal); incr -= frequency; while( incr < 0 ) { diff --git a/src/devices/sound/cdp1864.cpp b/src/devices/sound/cdp1864.cpp index 87901a6dd1b..e8168004811 100644 --- a/src/devices/sound/cdp1864.cpp +++ b/src/devices/sound/cdp1864.cpp @@ -279,9 +279,9 @@ void cdp1864_device::sound_stream_update(sound_stream &stream, std::vector<read_ signal = 1.0; } - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { - buffer.put(signal); + buffer.put(sampindex, signal); incr -= frequency; while( incr < 0 ) { diff --git a/src/devices/sound/cdp1869.cpp b/src/devices/sound/cdp1869.cpp index d4b789f3297..98580d7ae54 100644 --- a/src/devices/sound/cdp1869.cpp +++ b/src/devices/sound/cdp1869.cpp @@ -534,9 +534,9 @@ void cdp1869_device::sound_stream_update(sound_stream &stream, std::vector<read_ signal = stream_buffer::sample_t(m_toneamp) / 15.0; } - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { - buffer.put(signal); + buffer.put(sampindex, signal); incr -= frequency; while( incr < 0 ) { diff --git a/src/devices/sound/cem3394.cpp b/src/devices/sound/cem3394.cpp index e6966e49a3d..5b288d2d700 100644 --- a/src/devices/sound/cem3394.cpp +++ b/src/devices/sound/cem3394.cpp @@ -219,7 +219,7 @@ void cem3394_device::sound_stream_update(sound_stream &stream, std::vector<read_ logerror("%f V didn't cut it\n", m_values[WAVE_SELECT]); // loop over samples - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { // get the current VCO position and step it forward double vco_position = m_vco_position; @@ -250,7 +250,7 @@ void cem3394_device::sound_stream_update(sound_stream &stream, std::vector<read_ // compute extension input (for Bally/Sente this is the noise) if (ENABLE_EXTERNAL) - result += EXTERNAL_VOLUME * m_mixer_external * external.get(); + result += EXTERNAL_VOLUME * m_mixer_external * external.get(sampindex); // compute the modulated filter frequency and apply the filter // modulation tracks the VCO triangle @@ -258,7 +258,7 @@ void cem3394_device::sound_stream_update(sound_stream &stream, std::vector<read_ result = filter(result, filter_freq); // write the sample - buffer.put(result * m_volume); + buffer.put(sampindex, result * m_volume); } } diff --git a/src/devices/sound/dac.h b/src/devices/sound/dac.h index b45dc66dfad..7a48ba80998 100644 --- a/src/devices/sound/dac.h +++ b/src/devices/sound/dac.h @@ -87,12 +87,12 @@ protected: virtual void sound_stream_update_tag(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) override { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_pos = inputs[DAC_VREF_POS_INPUT].get() * this->m_gain; - double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get() * this->m_gain; + double const vref_pos = inputs[DAC_VREF_POS_INPUT].get(samp) * this->m_gain; + double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = vref_neg + dac_multiply<bits>(vref_pos - vref_neg, this->m_code); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } }; @@ -107,20 +107,20 @@ protected: { if (this->m_code & (1 << (bits - 1))) { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get() * this->m_gain; + double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = dac_multiply<bits - 1>(vref_neg, this->m_code ^ ~(~0U << bits)); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } else { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_pos = inputs[DAC_VREF_POS_INPUT].get() * this->m_gain; + double const vref_pos = inputs[DAC_VREF_POS_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = dac_multiply<bits - 1>(vref_pos, this->m_code); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } } @@ -134,12 +134,12 @@ protected: virtual void sound_stream_update_tag(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) override { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_pos = inputs[DAC_VREF_POS_INPUT].get() * this->m_gain; - double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get() * this->m_gain; + double const vref_pos = inputs[DAC_VREF_POS_INPUT].get(samp) * this->m_gain; + double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = vref_neg + dac_multiply<bits>(vref_pos - vref_neg, this->m_code ^ (1 << (bits - 1))); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } }; @@ -154,20 +154,20 @@ protected: { if (this->m_code & (1 << (bits - 1))) { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get() * this->m_gain; + double const vref_neg = inputs[DAC_VREF_NEG_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = dac_multiply<bits - 1>(vref_neg, this->m_code ^ (1 << (bits - 1))); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } else { - while (!outputs[0].done()) + for (int samp = 0; samp < outputs[0].samples(); samp++) { - double const vref_pos = inputs[DAC_VREF_POS_INPUT].get() * this->m_gain; + double const vref_pos = inputs[DAC_VREF_POS_INPUT].get(samp) * this->m_gain; stream_buffer::sample_t const vout = dac_multiply<bits - 1>(vref_pos, this->m_code); - outputs[0].put(vout); + outputs[0].put(samp, vout); } } } diff --git a/src/devices/sound/dave.cpp b/src/devices/sound/dave.cpp index 1c82ab6691e..f26ea891879 100644 --- a/src/devices/sound/dave.cpp +++ b/src/devices/sound/dave.cpp @@ -206,7 +206,7 @@ void dave_device::sound_stream_update(sound_stream &stream, std::vector<read_str auto &buffer2 = outputs[1]; constexpr stream_buffer::sample_t sample_scale = 1.0 / (4.0 * 32768.0); - while (!buffer1.done()) + for (int sampindex = 0; sampindex < buffer1.samples(); sampindex++) { int vol[4]; @@ -264,8 +264,8 @@ void dave_device::sound_stream_update(sound_stream &stream, std::vector<read_str left_volume = output_volumes[0] + output_volumes[2] + output_volumes[4] + output_volumes[6]; right_volume = output_volumes[1] + output_volumes[3] + output_volumes[5] + output_volumes[7]; - buffer1.put(stream_buffer::sample_t(left_volume) * sample_scale); - buffer2.put(stream_buffer::sample_t(right_volume) * sample_scale); + buffer1.put(sampindex, stream_buffer::sample_t(left_volume) * sample_scale); + buffer2.put(sampindex, stream_buffer::sample_t(right_volume) * sample_scale); } } diff --git a/src/devices/sound/digitalk.cpp b/src/devices/sound/digitalk.cpp index 6fe7cfe75b0..9ca0e6fa5f0 100644 --- a/src/devices/sound/digitalk.cpp +++ b/src/devices/sound/digitalk.cpp @@ -547,20 +547,26 @@ void digitalker_device::digitalker_step() void digitalker_device::sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) { auto &sout = outputs[0]; + int cpos = 0; constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!sout.done()) { + while(cpos != sout.samples()) { if(m_zero_count == 0 && m_dac_index == 128) digitalker_step(); if(m_zero_count) { - m_zero_count -= sout.fill(0, m_zero_count); + int n = sout.samples() - cpos; + if(n > m_zero_count) + n = m_zero_count; + sout.fill(0, cpos, n); + cpos += n; + m_zero_count -= n; } else if(m_dac_index != 128) { - while(!sout.done() && m_dac_index != 128) { + while(cpos != sout.samples() && m_dac_index != 128) { stream_buffer::sample_t v = stream_buffer::sample_t(m_dac[m_dac_index]) * sample_scale; int pp = m_pitch_pos; - while(!sout.done() && pp != m_pitch) { - sout.put(v); + while(cpos != sout.samples() && pp != m_pitch) { + sout.put(cpos++, v); pp++; } if(pp == m_pitch) { @@ -571,7 +577,7 @@ void digitalker_device::sound_stream_update(sound_stream &stream, std::vector<re } } else { - sout.fill(0); + sout.fill(0, cpos); break; } } diff --git a/src/devices/sound/disc_cls.h b/src/devices/sound/disc_cls.h index 3162d956f22..cb043557734 100644 --- a/src/devices/sound/disc_cls.h +++ b/src/devices/sound/disc_cls.h @@ -118,12 +118,13 @@ public: /* Add gain to the output and put into the buffers */ /* Clipping will be handled by the main sound system */ double val = DISCRETE_INPUT(0) * DISCRETE_INPUT(1); - m_outview->put(val * (1.0 / 32768.0)); + m_outview->put(m_outview_sample++, val * (1.0 / 32768.0)); } virtual int max_output(void) override { return 0; } - virtual void set_output_ptr(write_stream_view &view) override { m_outview = &view; m_outview->reset(); } + virtual void set_output_ptr(write_stream_view &view) override { m_outview = &view; m_outview_sample = 0; } private: write_stream_view *m_outview; + u32 m_outview_sample; }; DISCRETE_CLASS(dso_csvlog, 0, @@ -231,6 +232,7 @@ public: //protected: uint32_t m_stream_in_number; read_stream_view const *m_inview; /* current in ptr for stream */ + uint32_t m_inview_sample; private: void stream_generate(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs); diff --git a/src/devices/sound/disc_inp.hxx b/src/devices/sound/disc_inp.hxx index 729227dea24..1c6c9a6dba5 100644 --- a/src/devices/sound/disc_inp.hxx +++ b/src/devices/sound/disc_inp.hxx @@ -250,7 +250,10 @@ DISCRETE_STEP(dss_input_stream) { /* the context pointer is set to point to the current input stream data in discrete_stream_update */ if (EXPECTED(m_inview)) - set_output(0, m_inview->get() * 32768.0 * m_gain + m_offset); + { + set_output(0, m_inview->get(m_inview_sample) * 32768.0 * m_gain + m_offset); + m_inview_sample++; + } else set_output(0, 0); } diff --git a/src/devices/sound/discrete.cpp b/src/devices/sound/discrete.cpp index 9e3b4f60d0c..7d6a84a4fc7 100644 --- a/src/devices/sound/discrete.cpp +++ b/src/devices/sound/discrete.cpp @@ -1079,6 +1079,7 @@ void discrete_sound_device::sound_stream_update(sound_stream &stream, std::vecto for_each(discrete_dss_input_stream_node **, node, &m_input_stream_list) { (*node)->m_inview = &inputs[(*node)->m_stream_in_number]; + (*node)->m_inview_sample = 0; } /* just process it */ diff --git a/src/devices/sound/k054539.cpp b/src/devices/sound/k054539.cpp index 4a271009fa1..2ed7dc134a2 100644 --- a/src/devices/sound/k054539.cpp +++ b/src/devices/sound/k054539.cpp @@ -125,7 +125,7 @@ void k054539_device::sound_stream_update(sound_stream &stream, std::vector<read_ } constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!outputs[0].done()) { + for(int sample = 0; sample != outputs[0].samples(); sample++) { double lval, rval; if(!(flags & DISABLE_REVERB)) lval = rval = rbase[reverb_pos]; @@ -303,8 +303,8 @@ void k054539_device::sound_stream_update(sound_stream &stream, std::vector<read_ } } reverb_pos = (reverb_pos + 1) & 0x1fff; - outputs[0].put(stream_buffer::sample_t(lval) * sample_scale); - outputs[1].put(stream_buffer::sample_t(rval) * sample_scale); + outputs[0].put(sample, stream_buffer::sample_t(lval) * sample_scale); + outputs[1].put(sample, stream_buffer::sample_t(rval) * sample_scale); } } diff --git a/src/devices/sound/namco.cpp b/src/devices/sound/namco.cpp index cdad9636209..830f3713b45 100644 --- a/src/devices/sound/namco.cpp +++ b/src/devices/sound/namco.cpp @@ -237,9 +237,9 @@ void namco_audio_device::build_decoded_waveform(uint8_t *rgnbase) /* generate sound by oversampling */ uint32_t namco_audio_device::namco_update_one(write_stream_view &buffer, const int16_t *wave, uint32_t counter, uint32_t freq) { - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { - buffer.add(stream_buffer::sample_t(wave[WAVEFORM_POSITION(counter)]) * sample_scale); + buffer.add(sampindex, stream_buffer::sample_t(wave[WAVEFORM_POSITION(counter)]) * sample_scale); counter += freq; } @@ -672,8 +672,8 @@ void namco_audio_device::sound_stream_update(sound_stream &stream, std::vector<r /* loop over each voice and add its contribution */ for (sound_channel *voice = m_channel_list; voice < m_last_channel; voice++) { - auto &lmix = outputs[0].reset(); - auto &rmix = outputs[1].reset(); + auto &lmix = outputs[0]; + auto &rmix = outputs[1]; int lv = voice->volume[0]; int rv = voice->volume[1]; @@ -699,13 +699,13 @@ void namco_audio_device::sound_stream_update(sound_stream &stream, std::vector<r if (voice->noise_state) { - lmix.add(stream_buffer::sample_t(l_noise_data) * sample_scale); - rmix.add(stream_buffer::sample_t(r_noise_data) * sample_scale); + lmix.add(i, stream_buffer::sample_t(l_noise_data) * sample_scale); + rmix.add(i, stream_buffer::sample_t(r_noise_data) * sample_scale); } else { - lmix.add(stream_buffer::sample_t(-l_noise_data) * sample_scale); - rmix.add(stream_buffer::sample_t(-r_noise_data) * sample_scale); + lmix.add(i, stream_buffer::sample_t(-l_noise_data) * sample_scale); + rmix.add(i, stream_buffer::sample_t(-r_noise_data) * sample_scale); } if (hold) @@ -779,7 +779,6 @@ void namco_audio_device::sound_stream_update(sound_stream &stream, std::vector<r /* loop over each voice and add its contribution */ for (voice = m_channel_list; voice < m_last_channel; voice++) { - buffer.reset(); int v = voice->volume[0]; if (voice->noise_sw) { @@ -792,16 +791,17 @@ void namco_audio_device::sound_stream_update(sound_stream &stream, std::vector<r uint32_t delta = f << 4; uint32_t c = voice->noise_counter; int16_t noise_data = OUTPUT_LEVEL(0x07 * (v >> 1)); + int i; /* add our contribution */ - while (!buffer.done()) + for (i = 0; i < buffer.samples(); i++) { int cnt; if (voice->noise_state) - buffer.add(stream_buffer::sample_t(noise_data) * sample_scale); + buffer.add(i, stream_buffer::sample_t(noise_data) * sample_scale); else - buffer.add(stream_buffer::sample_t(-noise_data) * sample_scale); + buffer.add(i, stream_buffer::sample_t(-noise_data) * sample_scale); if (hold) { diff --git a/src/devices/sound/okim6295.cpp b/src/devices/sound/okim6295.cpp index 29461f7f37c..c3aee26e52e 100644 --- a/src/devices/sound/okim6295.cpp +++ b/src/devices/sound/okim6295.cpp @@ -182,7 +182,7 @@ void okim6295_device::sound_stream_update(sound_stream &stream, std::vector<read // iterate over voices and accumulate sample data for (auto & elem : m_voice) - elem.generate_adpcm(*this, outputs[0].reset()); + elem.generate_adpcm(*this, outputs[0]); } @@ -352,7 +352,7 @@ void okim6295_device::okim_voice::generate_adpcm(device_rom_interface &rom, writ // output to the buffer, scaling by the volume // signal in range -2048..2047 - buffer.add(m_adpcm.clock(nibble) * sample_scale * m_volume); + buffer.add(sampindex, m_adpcm.clock(nibble) * sample_scale * m_volume); // next! if (++m_sample >= m_count) diff --git a/src/devices/sound/pokey.cpp b/src/devices/sound/pokey.cpp index f405cd01255..f749afad2ae 100644 --- a/src/devices/sound/pokey.cpp +++ b/src/devices/sound/pokey.cpp @@ -702,11 +702,11 @@ void pokey_device::sound_stream_update(sound_stream &stream, std::vector<read_st double V0 = rTot / (rTot+m_r_pullup) * m_v_ref / 5.0; double mult = (m_cap == 0.0) ? 1.0 : 1.0 - exp(-(rTot + m_r_pullup) / (m_cap * m_r_pullup * rTot) * m_clock_period.as_double()); - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { /* store sum of output signals into the buffer */ m_out_filter += (V0 - m_out_filter) * mult; - buffer.put(m_out_filter); + buffer.put(sampindex, m_out_filter); } } else if (m_output_type == OPAMP_C_TO_GROUND) @@ -734,11 +734,11 @@ void pokey_device::sound_stream_update(sound_stream &stream, std::vector<read_st double V0 = (m_r_pullup / rTot) * m_v_ref / 5.0; double mult = (m_cap == 0.0) ? 1.0 : 1.0 - exp(-1.0 / (m_cap * m_r_pullup) * m_clock_period.as_double()); - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { /* store sum of output signals into the buffer */ m_out_filter += (V0 - m_out_filter) * mult; - buffer.put(m_out_filter); + buffer.put(sampindex, m_out_filter); } } else if (m_output_type == DISCRETE_VAR_R) diff --git a/src/devices/sound/samples.cpp b/src/devices/sound/samples.cpp index bfe7c02bb6b..5c8d129b799 100644 --- a/src/devices/sound/samples.cpp +++ b/src/devices/sound/samples.cpp @@ -334,7 +334,7 @@ void samples_device::sound_stream_update(sound_stream &stream, std::vector<read_ double endpos = chan.source_len; const int16_t *sample = chan.source; - while (!buffer.done()) + for (int sampindex = 0; sampindex < buffer.samples(); sampindex++) { // do a linear interp on the sample double pos_floor = floor(chan.pos); @@ -343,7 +343,7 @@ void samples_device::sound_stream_update(sound_stream &stream, std::vector<read_ stream_buffer::sample_t sample1 = stream_buffer::sample_t(sample[ipos++]); stream_buffer::sample_t sample2 = stream_buffer::sample_t(sample[(ipos + 1) % chan.source_len]); - buffer.put(sample_scale * ((1.0 - frac) * sample1 + frac * sample2)); + buffer.put(sampindex, sample_scale * ((1.0 - frac) * sample1 + frac * sample2)); // advance chan.pos += step; @@ -357,7 +357,7 @@ void samples_device::sound_stream_update(sound_stream &stream, std::vector<read_ { chan.source = nullptr; chan.source_num = -1; - buffer.fill(0); + buffer.fill(0, sampindex); break; } } diff --git a/src/devices/sound/sn76496.cpp b/src/devices/sound/sn76496.cpp index 6dcfe18c939..9bc686dedb2 100644 --- a/src/devices/sound/sn76496.cpp +++ b/src/devices/sound/sn76496.cpp @@ -379,7 +379,7 @@ void sn76496_base_device::sound_stream_update(sound_stream &stream, std::vector< int16_t out2 = 0; constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!lbuffer->done()) + for (int sampindex = 0; sampindex < lbuffer->samples(); sampindex++) { // clock chip once if (m_current_clock > 0) // not ready for new divided clock @@ -445,9 +445,9 @@ void sn76496_base_device::sound_stream_update(sound_stream &stream, std::vector< if (m_negate) { out = -out; out2 = -out2; } - lbuffer->put(stream_buffer::sample_t(out) * sample_scale); + lbuffer->put(sampindex, stream_buffer::sample_t(out) * sample_scale); if (m_stereo) - rbuffer->put(stream_buffer::sample_t(out2) * sample_scale); + rbuffer->put(sampindex, stream_buffer::sample_t(out2) * sample_scale); } } diff --git a/src/devices/sound/sp0250.cpp b/src/devices/sound/sp0250.cpp index da836cde53c..e18143a29b0 100644 --- a/src/devices/sound/sp0250.cpp +++ b/src/devices/sound/sp0250.cpp @@ -259,12 +259,12 @@ void sp0250_device::sound_stream_update(sound_stream &stream, std::vector<read_s if (!m_pwm_mode) { constexpr stream_buffer::sample_t sample_scale = 1.0 / 128.0; - while (!output.done()) - output.put(stream_buffer::sample_t(next()) * sample_scale); + for (int sampindex = 0; sampindex < output.samples(); sampindex++) + output.put(sampindex, stream_buffer::sample_t(next()) * sample_scale); } else { - while (!output.done()) + for (int sampindex = 0; sampindex < output.samples(); ) { // see where we're at in the current PWM cycle if (m_pwm_index >= PWM_CLOCKS) @@ -291,8 +291,14 @@ void sp0250_device::sound_stream_update(sound_stream &stream, std::vector<read_s remaining = PWM_CLOCKS - m_pwm_index; } + // clamp to the number of samples requested and advance the counters + if (remaining > output.samples() - sampindex) + remaining = output.samples() - sampindex; + m_pwm_index += remaining; + // fill the output - m_pwm_index += outputs[0].fill(value, remaining); + while (remaining-- != 0) + outputs[0].put(sampindex++, value); } } } diff --git a/src/devices/sound/tms3615.cpp b/src/devices/sound/tms3615.cpp index 53c1df7d52a..e59c64eab04 100644 --- a/src/devices/sound/tms3615.cpp +++ b/src/devices/sound/tms3615.cpp @@ -57,7 +57,7 @@ void tms3615_device::sound_stream_update(sound_stream &stream, std::vector<read_ int samplerate = buffer8.sample_rate(); constexpr stream_buffer::sample_t VMAX = 1.0f / stream_buffer::sample_t(TMS3615_TONES); - while (!buffer8.done()) + for (int sampindex = 0; sampindex < buffer8.samples(); sampindex++) { stream_buffer::sample_t sum8 = 0, sum16 = 0; @@ -94,8 +94,8 @@ void tms3615_device::sound_stream_update(sound_stream &stream, std::vector<read_ } } - buffer8.put(sum8); - buffer16.put(sum16); + buffer8.put(sampindex, sum8); + buffer16.put(sampindex, sum16); } } diff --git a/src/devices/sound/tms5220.cpp b/src/devices/sound/tms5220.cpp index 227de0f2be9..9c4974c34e2 100644 --- a/src/devices/sound/tms5220.cpp +++ b/src/devices/sound/tms5220.cpp @@ -2062,14 +2062,14 @@ void tms5220_device::sound_stream_update(sound_stream &stream, std::vector<read_ /* loop while we still have samples to generate */ constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!output.done()) + for (int sampindex = 0; sampindex < output.samples(); ) { - int length = (output.remaining() > MAX_SAMPLE_CHUNK) ? MAX_SAMPLE_CHUNK : output.remaining(); + int length = (output.samples() > MAX_SAMPLE_CHUNK) ? MAX_SAMPLE_CHUNK : output.samples(); /* generate the samples and copy to the target buffer */ process(sample_data, length); for (int index = 0; index < length; index++) - output.put(sample_data[index] * sample_scale); + output.put(sampindex++, sample_data[index] * sample_scale); } } diff --git a/src/devices/sound/upd7759.cpp b/src/devices/sound/upd7759.cpp index 1da8fd7c11c..fc07c0f95ab 100644 --- a/src/devices/sound/upd7759.cpp +++ b/src/devices/sound/upd7759.cpp @@ -738,11 +738,12 @@ void upd775x_device::sound_stream_update(sound_stream &stream, std::vector<read_ uint32_t pos = m_pos; /* loop until done */ + u32 index = 0; if (m_state != STATE_IDLE) - while (!outputs[0].done()) + for ( ; index < outputs[0].samples(); index++) { /* store the current sample */ - outputs[0].put(sample); + outputs[0].put(index, sample); /* advance by the number of clocks/output sample */ pos += step; @@ -774,7 +775,8 @@ void upd775x_device::sound_stream_update(sound_stream &stream, std::vector<read_ } /* if we got out early, just zap the rest of the buffer */ - outputs[0].fill(0); + for (; index < outputs[0].samples(); index++) + outputs[0].put(index, 0); /* flush the state back */ m_clocks_left = clocks_left; diff --git a/src/devices/sound/vgm_visualizer.cpp b/src/devices/sound/vgm_visualizer.cpp index 46471760328..c1f438f2b2c 100644 --- a/src/devices/sound/vgm_visualizer.cpp +++ b/src/devices/sound/vgm_visualizer.cpp @@ -289,12 +289,12 @@ void vgmviz_device::sound_stream_update(sound_stream &stream, std::vector<read_s device_mixer_interface::sound_stream_update(stream, inputs, outputs); // now consume the outputs - while (!outputs[0].done()) + for (int pos = 0; pos < outputs[0].samples(); pos++) { for (int i = 0; i < outputs.size(); i++) { // Original code took 16-bit sample / 65536.0 instead of 32768.0, so multiply by 0.5 here but is it necessary? - const float sample = outputs[i].get() * 0.5f; + const float sample = outputs[i].get(pos) * 0.5f; m_audio_buf[m_audio_fill_index][i][m_audio_count[m_audio_fill_index]] = sample + 0.5f; } diff --git a/src/devices/sound/votrax.cpp b/src/devices/sound/votrax.cpp index d8f98bb65cb..000a568d2a3 100644 --- a/src/devices/sound/votrax.cpp +++ b/src/devices/sound/votrax.cpp @@ -133,11 +133,11 @@ void votrax_sc01_device::inflection_w(uint8_t data) void votrax_sc01_device::sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) { - while (!outputs[0].done()) { + for(int i=0; i<outputs[0].samples(); i++) { m_sample_count++; if(m_sample_count & 1) chip_update(); - outputs[0].put(analog_calc()); + outputs[0].put(i, analog_calc()); } } diff --git a/src/devices/sound/ym2151.cpp b/src/devices/sound/ym2151.cpp index 82481fb79e4..b519d9a1224 100644 --- a/src/devices/sound/ym2151.cpp +++ b/src/devices/sound/ym2151.cpp @@ -1846,7 +1846,7 @@ void ym2151_device::sound_stream_update(sound_stream &stream, std::vector<read_s } constexpr stream_buffer::sample_t sample_scale = 1.0 / 32768.0; - while (!outputs[0].done()) + for (int sampindex=0; sampindex<outputs[0].samples(); sampindex++) { advance_eg(); @@ -1872,8 +1872,8 @@ void ym2151_device::sound_stream_update(sound_stream &stream, std::vector<read_s outr = 32767; else if (outr < -32768) outr = -32768; - outputs[0].put(stream_buffer::sample_t(outl) * sample_scale); - outputs[1].put(stream_buffer::sample_t(outr) * sample_scale); + outputs[0].put(sampindex, stream_buffer::sample_t(outl) * sample_scale); + outputs[1].put(sampindex, stream_buffer::sample_t(outr) * sample_scale); advance(); } diff --git a/src/devices/video/i8244.cpp b/src/devices/video/i8244.cpp index 3d7ca031a6e..8288dd9e552 100644 --- a/src/devices/video/i8244.cpp +++ b/src/devices/video/i8244.cpp @@ -745,11 +745,11 @@ void i8244_device::sound_stream_update(sound_stream &stream, std::vector<read_st u8 volume = m_vdc.s.sound & 0xf; stream_buffer::sample_t sample_on = (m_sh_output & m_vdc.s.sound >> 7) * 0.5; - while (!outputs[0].done()) + for (int i = 0; i < outputs[0].samples(); i++) { // clock duty cycle m_sh_duty = (m_sh_duty + 1) & 0xf; - outputs[0].put((m_sh_duty < volume) ? sample_on : 0.0); + outputs[0].put(i, (m_sh_duty < volume) ? sample_on : 0.0); } } diff --git a/src/emu/disound.cpp b/src/emu/disound.cpp index 1e7be43b32f..29d8b4e13b5 100644 --- a/src/emu/disound.cpp +++ b/src/emu/disound.cpp @@ -512,9 +512,9 @@ void device_mixer_interface::sound_stream_update(sound_stream &stream, std::vect int outputnum = m_outputmap[inputnum]; auto &output = outputs[outputnum]; if (!m_output_clear[outputnum]) - output.copy_indexed(0, input); + output.copy(input); else - output.add_indexed(0, input); + output.add(input); m_output_clear[outputnum] = true; } @@ -522,5 +522,5 @@ void device_mixer_interface::sound_stream_update(sound_stream &stream, std::vect // clear anything unused for (int outputnum = 0; outputnum < m_outputs; outputnum++) if (!m_output_clear[outputnum]) - outputs[outputnum].fill_indexed(0, 0); + outputs[outputnum].fill(0); } diff --git a/src/emu/sound.cpp b/src/emu/sound.cpp index d2c968ca370..a00546f7af3 100644 --- a/src/emu/sound.cpp +++ b/src/emu/sound.cpp @@ -208,7 +208,7 @@ void stream_buffer::flush_wav() // convert and fill for (int sampindex = 0; sampindex < cursamples; sampindex++) - buffer[sampindex] = s16(view.get_indexed(samplebase + sampindex) * 32768.0); + buffer[sampindex] = s16(view.get(samplebase + sampindex) * 32768.0); // write to the WAV wav_add_data_16(m_wav_file, buffer, cursamples); @@ -743,7 +743,7 @@ read_stream_view sound_stream::update_view(attotime start, attotime end, u32 out #if (SOUND_DEBUG) // clear each output view to NANs before we call the callback for (unsigned int outindex = 0; outindex < m_output.size(); outindex++) - m_output_view[outindex].fill_indexed(0, NAN); + m_output_view[outindex].fill(NAN); #endif // if we have an extended callback, that's all we need @@ -753,7 +753,7 @@ read_stream_view sound_stream::update_view(attotime start, attotime end, u32 out // make sure everything was overwritten for (unsigned int outindex = 0; outindex < m_output.size(); outindex++) for (int sampindex = 0; sampindex < m_output_view[outindex].samples(); sampindex++) - m_output_view[outindex].get_indexed(sampindex); + m_output_view[outindex].get(sampindex); for (unsigned int outindex = 0; outindex < m_output.size(); outindex++) m_output[outindex].m_buffer.flush_wav(); @@ -934,7 +934,7 @@ void sound_stream::stream_update_legacy(sound_stream &stream, std::vector<read_s { stream_sample_t *dest = inputptr[inputnum]; for (int index = 0; index < cursamples; index++) - dest[index] = stream_sample_t(inputs[inputnum].get_indexed(baseindex + index) * stream_buffer::sample_t(32768.0)); + dest[index] = stream_sample_t(inputs[inputnum].get(baseindex + index) * stream_buffer::sample_t(32768.0)); } // run the callback @@ -945,7 +945,7 @@ void sound_stream::stream_update_legacy(sound_stream &stream, std::vector<read_s { stream_sample_t *src = outputptr[outputnum]; for (int index = 0; index < cursamples; index++) - outputs[outputnum].put_indexed(baseindex + index, stream_buffer::sample_t(src[index]) * stream_buffer::sample_t(1.0 / 32768.0)); + outputs[outputnum].put(baseindex + index, stream_buffer::sample_t(src[index]) * stream_buffer::sample_t(1.0 / 32768.0)); } } } @@ -1012,6 +1012,7 @@ void default_resampler_stream::resampler_sound_update(sound_stream &stream, std: } // if we have equal sample rates, we just need to copy + auto numsamples = output.samples(); if (input.sample_rate() == output.sample_rate()) { output.copy(input); @@ -1033,13 +1034,14 @@ void default_resampler_stream::resampler_sound_update(sound_stream &stream, std: attotime latency = latency_samples * input.sample_period(); // clamp the latency to the start (only relevant at the beginning) + s32 dstindex = 0; attotime output_start = output.start_time(); - while (latency > output_start && !output.done()) + while (latency > output_start && dstindex < numsamples) { - output.put(0); + output.put(dstindex++, 0); output_start += output.sample_period(); } - if (output.done()) + if (dstindex >= numsamples) return; // create a rebased input buffer around the adjusted start time @@ -1053,14 +1055,15 @@ void default_resampler_stream::resampler_sound_update(sound_stream &stream, std: sound_assert(srcpos <= 1.0f); // input is undersampled: point sample except where our sample period covers a boundary + s32 srcindex = 0; if (step < 1.0) { - stream_buffer::sample_t cursample = rebased.get(); - while (!output.done()) + stream_buffer::sample_t cursample = rebased.get(srcindex++); + for ( ; dstindex < numsamples; dstindex++) { // if still within the current sample, just replicate if (srcpos <= 1.0) - output.put(cursample); + output.put(dstindex, cursample); // if crossing a sample boundary, blend with the neighbor else @@ -1068,18 +1071,19 @@ void default_resampler_stream::resampler_sound_update(sound_stream &stream, std: srcpos -= 1.0; sound_assert(srcpos <= step + 1e-5); stream_buffer::sample_t prevsample = cursample; - cursample = rebased.get(); - output.put(stepinv * (prevsample * (step - srcpos) + srcpos * cursample)); + cursample = rebased.get(srcindex++); + output.put(dstindex, stepinv * (prevsample * (step - srcpos) + srcpos * cursample)); } srcpos += step; } + sound_assert(srcindex <= rebased.samples()); } // input is oversampled: sum the energy else { - float cursample = rebased.get(); - while (!output.done()) + float cursample = rebased.get(srcindex++); + for ( ; dstindex < numsamples; dstindex++) { // compute the partial first sample and advance stream_buffer::sample_t scale = 1.0 - srcpos; @@ -1089,17 +1093,18 @@ void default_resampler_stream::resampler_sound_update(sound_stream &stream, std: stream_buffer::sample_t remaining = step - scale; while (remaining >= 1.0) { - sample += rebased.get(); + sample += rebased.get(srcindex++); remaining -= 1.0; } // add in the final partial sample - cursample = rebased.get(); + cursample = rebased.get(srcindex++); sample += cursample * remaining; - output.put(sample * stepinv); + output.put(dstindex, sample * stepinv); // our position is now the remainder srcpos = remaining; + sound_assert(srcindex <= rebased.samples()); } } } diff --git a/src/emu/sound.h b/src/emu/sound.h index d7a833c749d..5ccc78c015b 100644 --- a/src/emu/sound.h +++ b/src/emu/sound.h @@ -239,7 +239,6 @@ public: m_buffer(buffer), m_end(end), m_start(start), - m_offset(0), m_gain(gain) { normalize_start_end(); @@ -276,7 +275,6 @@ public: m_start = rhs.m_start; m_end = rhs.m_end; m_gain = rhs.m_gain; - m_offset = 0; normalize_start_end(); return *this; } @@ -294,18 +292,6 @@ public: // return the number of samples represented by the buffer u32 samples() const { return m_end - m_start; } - // return the current offset within the buffer - s32 offset() const { return m_offset; } - - // return the number of samples remaining from the current offset - s32 remaining() const { return samples() - m_offset; } - - // (re)set the current offset within the buffer - read_stream_view const &reset(s32 offset = 0) const { m_offset = offset; return *this; } - - // return true if we've consumed all the samples in the view - bool done() const { return m_offset >= samples(); } - // return the starting or ending time of the buffer attotime start_time() const { return m_buffer->index_time(m_start); } attotime end_time() const { return m_buffer->index_time(m_end); } @@ -317,7 +303,7 @@ public: read_stream_view &apply_gain(float gain) { m_gain *= gain; return *this; } // safely fetch a gain-scaled sample from the buffer - sample_t get_indexed(s32 index) const + sample_t get(s32 index) const { sound_assert(u32(index) < samples()); index += m_start; @@ -328,7 +314,7 @@ public: // safely fetch a raw sample from the buffer; if you use this, you need to // apply the gain yourself for correctness - sample_t getraw_indexed(s32 index) const + sample_t getraw(s32 index) const { sound_assert(u32(index) < samples()); index += m_start; @@ -337,10 +323,6 @@ public: return m_buffer->get(index); } - // non-indexed versions of the above that use the implied offset - sample_t get() const { sound_assert(m_offset < samples()); return get_indexed(m_offset++); } - sample_t getraw() const { sound_assert(m_offset < samples()); return getraw_indexed(m_offset++); } - protected: // normalize start/end void normalize_start_end() @@ -356,7 +338,6 @@ protected: stream_buffer *m_buffer; // pointer to the stream buffer we're viewing s32 m_end; // ending sample index (always >= start) s32 m_start; // starting sample index - mutable s32 m_offset; // current read/write offset sample_t m_gain; // overall gain factor }; @@ -383,11 +364,8 @@ public: { } - // (re)set the current offset within the buffer - write_stream_view &reset(s32 offset = 0) { m_offset = offset; return *this; } - // safely write a gain-applied sample to the buffer - void put_indexed(s32 index, sample_t sample) + void put(s32 index, sample_t sample) { sound_assert(u32(index) < samples()); index += m_start; @@ -397,7 +375,7 @@ public: } // safely add a gain-applied sample to the buffer - void add_indexed(s32 index, sample_t sample) + void add(s32 index, sample_t sample) { sound_assert(u32(index) < samples()); index += m_start; @@ -407,9 +385,9 @@ public: } // fill part of the view with the given value - s32 fill_indexed(s32 start, sample_t value, s32 count = -1) + void fill(sample_t value, s32 start, s32 count) { - if (count < 0 || start + count > samples()) + if (start + count > samples()) count = samples() - start; u32 index = start + m_start; for (s32 sampindex = 0; sampindex < count; sampindex++) @@ -417,51 +395,39 @@ public: m_buffer->put(index, value); index = m_buffer->next_index(index); } - return count; } + void fill(sample_t value, s32 start) { fill(value, start, samples() - start); } + void fill(sample_t value) { fill(value, 0, samples()); } // copy data from another view - s32 copy_indexed(s32 start, read_stream_view const &src, s32 srcstart = -1, s32 count = -1) + void copy(read_stream_view const &src, s32 start, s32 count) { - if (count < 0 || start + count > samples()) + if (start + count > samples()) count = samples() - start; - if (srcstart < 0) - srcstart = start; - if (srcstart + count > src.samples()) - count = src.samples() - srcstart; u32 index = start + m_start; for (s32 sampindex = 0; sampindex < count; sampindex++) { - m_buffer->put(index, src.get_indexed(srcstart + sampindex)); + m_buffer->put(index, src.get(start + sampindex)); index = m_buffer->next_index(index); } - return count; } + void copy(read_stream_view const &src, s32 start) { copy(src, start, samples() - start); } + void copy(read_stream_view const &src) { copy(src, 0, samples()); } // add data from another view to our current values - s32 add_indexed(s32 start, read_stream_view const &src, s32 srcstart = -1, s32 count = -1) + void add(read_stream_view const &src, s32 start, s32 count) { - if (count < 0 || start + count > samples()) + if (start + count > samples()) count = samples() - start; - if (srcstart < 0) - srcstart = start; - if (srcstart + count > src.samples()) - count = src.samples() - srcstart; u32 index = start + m_start; for (s32 sampindex = 0; sampindex < count; sampindex++) { - m_buffer->put(index, m_buffer->get(index) + src.get_indexed(start + sampindex)); + m_buffer->put(index, m_buffer->get(index) + src.get(start + sampindex)); index = m_buffer->next_index(index); } - return count; } - - // non-indexed versions of the above that use the implied offset - void put(sample_t sample) { sound_assert(m_offset < samples()); put_indexed(m_offset++, sample); } - void add(sample_t sample) { sound_assert(m_offset < samples()); add_indexed(m_offset++, sample); } - s32 fill(sample_t value, s32 count = -1) { m_offset += count = fill_indexed(m_offset, value, count); return count; } - s32 copy(read_stream_view const &src, s32 count = -1) { count = copy_indexed(m_offset, src, m_offset, count); m_offset += count; src.reset(src.offset() + count); return count; } - s32 add(read_stream_view const &src, s32 count = -1) { count = add_indexed(m_offset, src, m_offset, count); m_offset += count; src.reset(src.offset() + count); return count; } + void add(read_stream_view const &src, s32 start) { add(src, start, samples() - start); } + void add(read_stream_view const &src) { add(src, 0, samples()); } }; diff --git a/src/emu/speaker.cpp b/src/emu/speaker.cpp index b487b5d8aea..3cbb14355a4 100644 --- a/src/emu/speaker.cpp +++ b/src/emu/speaker.cpp @@ -74,9 +74,9 @@ void speaker_device::mix(stream_buffer::sample_t *leftmix, stream_buffer::sample if (machine().options().speaker_report() != 0) { u32 samples_per_bucket = m_mixer_stream->sample_rate() / BUCKETS_PER_SECOND; - while (!view.done()) + for (int sample = 0; sample < expected_samples; sample++) { - m_current_max = std::max(m_current_max, fabsf(view.get())); + m_current_max = std::max(m_current_max, fabsf(view.get(sample))); if (++m_samples_this_bucket >= samples_per_bucket) { m_max_sample.push_back(m_current_max); @@ -84,7 +84,6 @@ void speaker_device::mix(stream_buffer::sample_t *leftmix, stream_buffer::sample m_samples_this_bucket = 0; } } - view.reset(); } // mix if sound is enabled @@ -94,7 +93,7 @@ void speaker_device::mix(stream_buffer::sample_t *leftmix, stream_buffer::sample if (m_x == 0) for (int sample = 0; sample < expected_samples; sample++) { - stream_buffer::sample_t cursample = view.get(); + stream_buffer::sample_t cursample = view.get(sample); leftmix[sample] += cursample; rightmix[sample] += cursample; } @@ -102,12 +101,12 @@ void speaker_device::mix(stream_buffer::sample_t *leftmix, stream_buffer::sample // if the speaker is to the left, send only to the left else if (m_x < 0) for (int sample = 0; sample < expected_samples; sample++) - leftmix[sample] += view.get(); + leftmix[sample] += view.get(sample); // if the speaker is to the right, send only to the right else for (int sample = 0; sample < expected_samples; sample++) - rightmix[sample] += view.get(); + rightmix[sample] += view.get(sample); } } diff --git a/src/mame/audio/sente6vb.cpp b/src/mame/audio/sente6vb.cpp index 311f8929c17..22c9721a18a 100644 --- a/src/mame/audio/sente6vb.cpp +++ b/src/mame/audio/sente6vb.cpp @@ -98,9 +98,9 @@ protected: // sound stream update overrides virtual void sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs) override { - while (!outputs[0].done()) + for (int sampindex = 0; sampindex < outputs[0].samples(); sampindex++) { - outputs[0].put(BIT(m_noise_state, 0) ? 1.0 : 0.0); + outputs[0].put(sampindex, BIT(m_noise_state, 0) ? 1.0 : 0.0); m_noise_state = (m_noise_state >> 1) | ((BIT(m_noise_state, 0) ^ BIT(m_noise_state, 3)) << 16); } } |