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Diffstat (limited to 'src/osd/windows/sound.c')
-rw-r--r--src/osd/windows/sound.c14
1 files changed, 7 insertions, 7 deletions
diff --git a/src/osd/windows/sound.c b/src/osd/windows/sound.c
index 12f5f873bbc..1e7dc3e0d6d 100644
--- a/src/osd/windows/sound.c
+++ b/src/osd/windows/sound.c
@@ -101,7 +101,7 @@ static int buffer_overflows;
//============================================================
static void sound_exit(running_machine &machine);
-static HRESULT dsound_init(running_machine *machine);
+static HRESULT dsound_init(running_machine &machine);
static void dsound_kill(void);
static HRESULT dsound_create_buffers(void);
static void dsound_destroy_buffers(void);
@@ -112,14 +112,14 @@ static void dsound_destroy_buffers(void);
// winsound_init
//============================================================
-void winsound_init(running_machine *machine)
+void winsound_init(running_machine &machine)
{
// if no sound, don't create anything
- if (!machine->options().sound())
+ if (!machine.options().sound())
return;
// ensure we get called on the way out
- machine->add_notifier(MACHINE_NOTIFY_EXIT, sound_exit);
+ machine.add_notifier(MACHINE_NOTIFY_EXIT, sound_exit);
// attempt to initialize directsound
// don't make it fatal if we can't -- we'll just run without sound
@@ -267,7 +267,7 @@ void windows_osd_interface::set_mastervolume(int attenuation)
// dsound_init
//============================================================
-static HRESULT dsound_init(running_machine *machine)
+static HRESULT dsound_init(running_machine &machine)
{
HRESULT result;
@@ -300,12 +300,12 @@ static HRESULT dsound_init(running_machine *machine)
stream_format.wBitsPerSample = 16;
stream_format.wFormatTag = WAVE_FORMAT_PCM;
stream_format.nChannels = 2;
- stream_format.nSamplesPerSec = machine->sample_rate();
+ stream_format.nSamplesPerSec = machine.sample_rate();
stream_format.nBlockAlign = stream_format.wBitsPerSample * stream_format.nChannels / 8;
stream_format.nAvgBytesPerSec = stream_format.nSamplesPerSec * stream_format.nBlockAlign;
// compute the buffer size based on the output sample rate
- stream_buffer_size = stream_format.nSamplesPerSec * stream_format.nBlockAlign * downcast<windows_options &>(machine->options()).audio_latency() / 10;
+ stream_buffer_size = stream_format.nSamplesPerSec * stream_format.nBlockAlign * downcast<windows_options &>(machine.options()).audio_latency() / 10;
stream_buffer_size = (stream_buffer_size / 1024) * 1024;
if (stream_buffer_size < 1024)
stream_buffer_size = 1024;