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Diffstat (limited to 'src/devices/sound/spkrdev.cpp')
-rw-r--r--src/devices/sound/spkrdev.cpp30
1 files changed, 15 insertions, 15 deletions
diff --git a/src/devices/sound/spkrdev.cpp b/src/devices/sound/spkrdev.cpp
index c44e00f5708..184a628fe73 100644
--- a/src/devices/sound/spkrdev.cpp
+++ b/src/devices/sound/spkrdev.cpp
@@ -76,7 +76,7 @@
#include "sound/spkrdev.h"
-static constexpr int16_t default_levels[2] = {0, 32767};
+static constexpr double default_levels[2] = {0.0, 1.0};
// Internal oversampling factor (interm. samples vs stream samples)
static constexpr int RATE_MULTIPLIER = 4;
@@ -102,7 +102,7 @@ void speaker_sound_device::device_start()
int i;
double x;
- m_channel = machine().sound().stream_alloc(*this, 0, 1, machine().sample_rate());
+ m_channel = stream_alloc(0, 1, machine().sample_rate());
m_level = 0;
for (i = 0; i < FILTER_LENGTH; i++)
@@ -202,19 +202,19 @@ void speaker_sound_device::device_post_load()
//-------------------------------------------------
// This can be triggered by the core (based on emulated time) or via level_w().
-void speaker_sound_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **outputs, int samples)
+void speaker_sound_device::sound_stream_update(sound_stream &stream, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs)
{
- stream_sample_t *buffer = outputs[0];
- int volume = m_levels[m_level];
+ auto &buffer = outputs[0];
+ double volume = m_levels[m_level];
double filtered_volume;
attotime sampled_time = attotime::zero;
- if (samples > 0)
+ if (buffer.samples() > 0)
{
/* Prepare to update time state */
sampled_time = attotime(0, m_channel_sample_period);
- if (samples > 1)
- sampled_time *= samples;
+ if (buffer.samples() > 1)
+ sampled_time *= buffer.samples();
/* Note: since the stream is in the process of being updated,
* stream->sample_time() will return the time before the update! (MAME 0.130)
@@ -222,19 +222,19 @@ void speaker_sound_device::sound_stream_update(sound_stream &stream, stream_samp
*/
}
- if (samples-- > 0)
+ for (int sampindex = 0; sampindex < buffer.samples(); )
{
/* Note that first interm. sample may be composed... */
filtered_volume = update_interm_samples_get_filtered_volume(volume);
/* Composite volume is now quantized to the stream resolution */
- *buffer++ = (stream_sample_t)filtered_volume;
+ buffer.put(sampindex++, filtered_volume);
/* Any additional samples will be homogeneous, however may need filtering across samples: */
- while (samples-- > 0)
+ while (sampindex < buffer.samples())
{
filtered_volume = update_interm_samples_get_filtered_volume(volume);
- *buffer++ = (stream_sample_t)filtered_volume;
+ buffer.put(sampindex++, filtered_volume);
}
/* Update the time state */
@@ -322,7 +322,7 @@ void speaker_sound_device::update_interm_samples(const attotime &time, int volum
}
-double speaker_sound_device::update_interm_samples_get_filtered_volume(int volume)
+double speaker_sound_device::update_interm_samples_get_filtered_volume(double volume)
{
double filtered_volume, tempx;
@@ -355,7 +355,7 @@ double speaker_sound_device::update_interm_samples_get_filtered_volume(int volum
}
-void speaker_sound_device::finalize_interm_sample(int volume)
+void speaker_sound_device::finalize_interm_sample(double volume)
{
double fraction;
@@ -401,7 +401,7 @@ double speaker_sound_device::get_filtered_volume()
{
if (i >= FILTER_LENGTH) i = 0;
filtered_volume += m_composed_volume[i] * m_ampl[c];
- ampsum += m_ampl[c];
+ ampsum += std::abs(m_ampl[c]);
}
filtered_volume /= ampsum;