AstralTransRocketries/ClassiCube
1
0
Fork 0
mirror of https://github.com/ClassiCube/ClassiCube.git synced 2025-09-13 09:35:23 -04:00
Code Issues Packages Projects Releases Wiki Activity

Drop support for 8 bit wav files that were never used anyways.

Browse Source
This commit is contained in:
UnknownShadow200 2020-02-16 09:04:23 +11:00
parent 63a06b0a17
commit a2dfccc560
3 changed files with 16 additions and 29 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

40
src/Audio.c
View File

@ -75,13 +75,6 @@ static void Volume_Mix16(cc_int16* samples, int count, int volume) {
}
}
static void Volume_Mix8(cc_uint8* samples, int count, int volume) {
int i;
for (i = 0; i < count; i++, samples++) {
samples[0] = (127 + (samples[0] - 127) * volume / 100);
}
}
/*########################################################################################################################*
*------------------------------------------------Native implementation----------------------------------------------------*
@ -140,14 +133,14 @@ cc_result Audio_SetFormat(AudioHandle handle, struct AudioFormat* format) {
if (AudioFormat_Eq(cur, format)) return 0;
if (ctx->Handle && (res = waveOutClose(ctx->Handle))) return res;
int sampleSize = format->Channels * format->BitsPerSample / 8;
int sampleSize = format->Channels * 2; /* 16 bits per sample / 8 */
WAVEFORMATEX fmt;
fmt.wFormatTag = WAVE_FORMAT_PCM;
fmt.nChannels = format->Channels;
fmt.nSamplesPerSec = format->SampleRate;
fmt.nAvgBytesPerSec = format->SampleRate * sampleSize;
fmt.nBlockAlign = sampleSize;
fmt.wBitsPerSample = format->BitsPerSample;
fmt.wBitsPerSample = 16;
fmt.cbSize = 0;
ctx->Format = *format;
@ -311,14 +304,9 @@ cc_result Audio_Close(AudioHandle handle) {
return 0;
}
static ALenum GetALFormat(int channels, int bitsPerSample) {
if (bitsPerSample == 16) {
if (channels == 1) return AL_FORMAT_MONO16;
if (channels == 2) return AL_FORMAT_STEREO16;
} else if (bitsPerSample == 8) {
if (channels == 1) return AL_FORMAT_MONO8;
if (channels == 2) return AL_FORMAT_STEREO8;
}
static ALenum GetALFormat(int channels) {
if (channels == 1) return AL_FORMAT_MONO16;
if (channels == 2) return AL_FORMAT_STEREO16;
Logger_Abort("Unsupported audio format"); return 0;
}
@ -328,7 +316,7 @@ cc_result Audio_SetFormat(AudioHandle handle, struct AudioFormat* format) {
ALenum err;
if (AudioFormat_Eq(cur, format)) return 0;
ctx->DataFormat = GetALFormat(format->Channels, format->BitsPerSample);
ctx->DataFormat = GetALFormat(format->Channels);
ctx->Format = *format;
if ((err = Audio_FreeSource(ctx))) return err;
@ -453,6 +441,7 @@ static cc_result Sound_ReadWaveData(struct Stream* stream, struct Sound* snd) {
cc_uint32 fourCC, size;
cc_uint8 tmp[WAV_FMT_SIZE];
cc_result res;
int bitsPerSample;
if ((res = Stream_Read(stream, tmp, 12))) return res;
fourCC = Stream_GetU32_BE(&tmp[0]);
@ -473,7 +462,9 @@ static cc_result Sound_ReadWaveData(struct Stream* stream, struct Sound* snd) {
snd->Format.Channels = Stream_GetU16_LE(&tmp[2]);
snd->Format.SampleRate = Stream_GetU32_LE(&tmp[4]);
/* tmp[8] (6) alignment data and stuff */
snd->Format.BitsPerSample = Stream_GetU16_LE(&tmp[14]);
bitsPerSample = Stream_GetU16_LE(&tmp[14]);
if (bitsPerSample != 16) return WAV_ERR_SAMPLE_BITS;
size -= WAV_FMT_SIZE;
} else if (fourCC == WAV_FourCC('d','a','t','a')) {
snd->Data = (cc_uint8*)Mem_TryAlloc(size, 1);
@ -615,11 +606,7 @@ static void Sounds_PlayRaw(struct SoundOutput* output, struct Sound* snd, struct
data = output->Buffer;
Mem_Copy(data, snd->Data, snd->Size);
if (fmt->BitsPerSample == 8) {
Volume_Mix8((cc_uint8*)data, snd->Size, volume);
} else {
Volume_Mix16((cc_int16*)data, snd->Size / 2, volume);
}
Volume_Mix16((cc_int16*)data, snd->Size / 2, volume);
}
if ((res = Audio_BufferData(output->Handle, 0, data, snd->Size))) { Sounds_Fail(res); return; }
@ -772,9 +759,8 @@ static cc_result Music_PlayOgg(struct Stream* source) {
vorbis.source = &stream;
if ((res = Vorbis_DecodeHeaders(&vorbis))) goto cleanup;
fmt.Channels = vorbis.channels;
fmt.SampleRate = vorbis.sampleRate;
fmt.BitsPerSample = 16;
fmt.Channels = vorbis.channels;
fmt.SampleRate = vorbis.sampleRate;
if ((res = Audio_SetFormat(music_out, &fmt))) goto cleanup;
/* largest possible vorbis frame decodes to blocksize1 * channels samples */

4
src/Audio.h
View File

@ -21,8 +21,8 @@ void Audio_PlayStepSound(cc_uint8 type);
#define AUDIO_MAX_BUFFERS 4
/* Information about a source of audio. */
struct AudioFormat { cc_uint16 Channels, BitsPerSample; int SampleRate; };
#define AudioFormat_Eq(a, b) ((a)->Channels == (b)->Channels && (a)->BitsPerSample == (b)->BitsPerSample && (a)->SampleRate == (b)->SampleRate)
struct AudioFormat { cc_uint16 Channels; int SampleRate; };
#define AudioFormat_Eq(a, b) ((a)->Channels == (b)->Channels && (a)->SampleRate == (b)->SampleRate)
typedef int AudioHandle;
/* Acquires an audio context. */

1
src/Errors.h
View File

@ -13,6 +13,7 @@ enum ERRORS_ALL {
OGG_ERR_INVALID_SIG, OGG_ERR_VERSION,
/* WAV audio decoding errors*/
WAV_ERR_STREAM_HDR, WAV_ERR_STREAM_TYPE, WAV_ERR_DATA_TYPE, WAV_ERR_NO_DATA,
WAV_ERR_SAMPLE_BITS,
/* Vorbis audio decoding errors */
VORBIS_ERR_HEADER,
VORBIS_ERR_WRONG_HEADER, VORBIS_ERR_FRAMING,