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Diffstat (limited to 'src/devices/sound/spkrdev.cpp')
-rw-r--r--src/devices/sound/spkrdev.cpp21
1 files changed, 10 insertions, 11 deletions
diff --git a/src/devices/sound/spkrdev.cpp b/src/devices/sound/spkrdev.cpp
index 7d9333d02cb..16dc6199c2f 100644
--- a/src/devices/sound/spkrdev.cpp
+++ b/src/devices/sound/spkrdev.cpp
@@ -72,7 +72,7 @@
*/
#include "emu.h"
-#include "sound/spkrdev.h"
+#include "spkrdev.h"
// The default is 1-bit, but can be customized with set_levels.
static constexpr double default_levels[2] = { 0.0, 1.0 };
@@ -187,19 +187,18 @@ 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, std::vector<read_stream_view> const &inputs, std::vector<write_stream_view> &outputs)
+void speaker_sound_device::sound_stream_update(sound_stream &stream)
{
- auto &buffer = outputs[0];
double volume = m_levels[m_level];
double filtered_volume;
attotime sampled_time = attotime::zero;
- if (buffer.samples() > 0)
+ if (stream.samples() > 0)
{
/* Prepare to update time state */
sampled_time = attotime(0, m_channel_sample_period);
- if (buffer.samples() > 1)
- sampled_time *= buffer.samples();
+ if (stream.samples() > 1)
+ sampled_time *= stream.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)
@@ -207,19 +206,19 @@ void speaker_sound_device::sound_stream_update(sound_stream &stream, std::vector
*/
}
- for (int sampindex = 0; sampindex < buffer.samples(); )
+ for (int sampindex = 0; sampindex < stream.samples(); )
{
/* Note that first intermediate sample may be composed... */
filtered_volume = update_interm_samples_get_filtered_volume(volume);
/* Composite volume is now quantized to the stream resolution */
- buffer.put(sampindex++, filtered_volume);
+ stream.put(0, sampindex++, filtered_volume);
/* Any additional samples will be homogeneous, however may need filtering across samples: */
- while (sampindex < buffer.samples())
+ while (sampindex < stream.samples())
{
filtered_volume = update_interm_samples_get_filtered_volume(volume);
- buffer.put(sampindex++, filtered_volume);
+ stream.put(0, sampindex++, filtered_volume);
}
/* Update the time state */
@@ -269,7 +268,7 @@ void speaker_sound_device::level_w(int new_level)
/* This is redundant because time update has to be done within sound_stream_update() anyway,
* however this ensures synchronization between the speaker and stream timing:
*/
- m_channel_last_sample_time = m_channel->sample_time();
+ m_channel_last_sample_time = m_channel->end_time();
/* sample_time() may be ahead of us */
if (m_channel_last_sample_time > time)