len0rd/rockbox
1
0
Fork 1
mirror of https://github.com/Rockbox/rockbox.git synced 2026-10-10 08:03:04 -04:00
Code Issues Projects Releases Packages Wiki Activity Actions

doom: Rework audio init to properly respect pcm_sink capabilities

Browse source 
Try to use 11KHz if possible, if that's not an option, fall
back to 44KHz which is always available.  Simulators always use
44KHz.

Change-Id: I7547e7c1bddf7ce4634f7fd7bc64d4a5c9b62c29
This commit is contained in:
Solomon Peachy 2026-03-05 07:51:27 -05:00
parent f81126bcf0
commit ba69fb9bbf
1 changed files with 24 additions and 14 deletions
Download patch file Download diff file Expand all files Collapse all files

38
apps/plugins/doom/i_sound.c
View file

@ -51,16 +51,11 @@
#define SAMPLECOUNT 128 #define SAMPLECOUNT 128
#define NUM_CHANNELS 24 #define NUM_CHANNELS 24
// It is 2 for 16bit, and 2 for two channels. #define BUFMUL 2 // 2 channels
#define BUFMUL 2 #define SAMPLESIZE sizeof(short) // 16-bit
#define MIXBUFFERSIZE (SAMPLECOUNT*BUFMUL) #define MIXBUFFERSIZE (SAMPLECOUNT*BUFMUL)
#ifdef HW_HAVE_11 static unsigned short samplerate;
#define SAMPLERATE SAMPR_11 // 44100 22050 11025
#else
#define SAMPLERATE SAMPR_44 // 44100 22050 11025
#endif
#define SAMPLESIZE 2 // 16bit
// The global mixing buffer. // The global mixing buffer.
// Basically, samples from all active internal channels // Basically, samples from all active internal channels
@ -207,9 +202,9 @@ static void updateSoundParams(int handle, int volume, int seperation, int pitch)
// Patched to shift left *then* divide, to minimize roundoff errors // Patched to shift left *then* divide, to minimize roundoff errors
// as well as to use SAMPLERATE as defined above, not to assume 11025 Hz // as well as to use SAMPLERATE as defined above, not to assume 11025 Hz
if (pitched_sounds) if (pitched_sounds)
channelinfo[slot].step = step + (((channelinfo[slot].samplerate<<16)/SAMPLERATE)-65536); channelinfo[slot].step = step + (((channelinfo[slot].samplerate<<16)/samplerate)-65536);
else else
channelinfo[slot].step = ((channelinfo[slot].samplerate<<16)/SAMPLERATE); channelinfo[slot].step = ((channelinfo[slot].samplerate<<16)/samplerate);
// Separation, that is, orientation/stereo. // Separation, that is, orientation/stereo.
// range is: 1 - 256 // range is: 1 - 256
@ -265,7 +260,7 @@ void I_SetChannels()
// This table provides step widths for pitch parameters. // This table provides step widths for pitch parameters.
for (i=-128 ; i<128 ; i++) for (i=-128 ; i<128 ; i++)
steptablemid[i]=2; steptablemid[i]=2;
// steptablemid[i] = (int)(pow(1.2, ((double)i/(64.0*SAMPLERATE/11025)))*65536.0); // steptablemid[i] = (int)(pow(1.2, ((double)i/(64.0*samplerate/11025)))*65536.0);
// Generates volume lookup tables // Generates volume lookup tables
// which also turn the unsigned samples // which also turn the unsigned samples
@ -462,7 +457,7 @@ void get_more(const void** start, size_t* size)
I_UpdateSound(); // Force sound update I_UpdateSound(); // Force sound update
*start = mixbuffer; *start = mixbuffer;
*size = MIXBUFFERSIZE*sizeof(short); *size = MIXBUFFERSIZE*SAMPLESIZE;
} }
@ -480,10 +475,25 @@ void I_ShutdownSound(void)
rb->mixer_set_frequency(HW_SAMPR_DEFAULT); rb->mixer_set_frequency(HW_SAMPR_DEFAULT);
} }
#if defined(SIMULATOR)
#define PREFERRED_SAMPR SAMPR_44
#else
#define PREFERRED_SAMPR SAMPR_11
#endif
void I_InitSound() void I_InitSound()
{ {
int i; int i;
const struct pcm_sink_caps* caps = rb->pcm_current_sink_caps();
for (i = 0 ; i < caps->num_samprs; i++) {
if (caps->samprs[i] == PREFERRED_SAMPR)
break;
}
if (i == caps->num_samprs)
i = SAMPR_44;
samplerate = caps->samprs[i];
// Initialize external data (all sounds) at start, keep static. // Initialize external data (all sounds) at start, keep static.
printf( "I_InitSound: "); printf( "I_InitSound: ");
rb->audio_stop(); rb->audio_stop();
@ -493,11 +503,11 @@ void I_InitSound()
rb->audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK); rb->audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
rb->audio_set_output_source(AUDIO_SRC_PLAYBACK); rb->audio_set_output_source(AUDIO_SRC_PLAYBACK);
#endif #endif
rb->mixer_set_frequency(SAMPLERATE); rb->mixer_set_frequency(samplerate);
vol_lookup=malloc(128*256*sizeof(int)); vol_lookup=malloc(128*256*sizeof(int));
mixbuffer=malloc(MIXBUFFERSIZE*sizeof(short)); mixbuffer=malloc(MIXBUFFERSIZE*SAMPLESIZE);
steptable=malloc(256*sizeof(int)); steptable=malloc(256*sizeof(int));
for (i=1 ; i<NUMSFX ; i++) for (i=1 ; i<NUMSFX ; i++)