mirror of
https://github.com/openmoh/openmohaa.git
synced 2025-04-28 13:47:58 +03:00
Add support for IMA-ADPCM sounds (4-bit)
Sounds such as sea waves (mohaa m3l1a) and subpen (mohaas flughafen) will be played accordingly. This uses OpenAL extensions such as AL_EXT_IMA4 and SOFT features such as AL_SOFT_block_alignment
This commit is contained in:
smallmodel
GitHub
parent
459e13dc76
commit
1e9decf193
8 changed files with 135 additions and 78 deletions
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@ -81,6 +81,7 @@ LPALGENBUFFERS qalGenBuffers;
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LPALDELETEBUFFERS qalDeleteBuffers;
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LPALISBUFFER qalIsBuffer;
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LPALBUFFERDATA qalBufferData;
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LPALBUFFERI qalBufferi;
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LPALGETBUFFERF qalGetBufferf;
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LPALGETBUFFERFV qalGetBufferfv;
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LPALGETBUFFERI qalGetBufferi;
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@ -201,6 +202,7 @@ qboolean QAL_Init(const char *libname)
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qalDeleteBuffers = GPA("alDeleteBuffers");
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qalIsBuffer = GPA("alIsBuffer");
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qalBufferData = GPA("alBufferData");
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qalBufferi = GPA("alBufferi");
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qalGetBufferf = GPA("alGetBufferf");
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qalGetBufferfv = GPA("alGetBufferfv");
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qalGetBufferi = GPA("alGetBufferi");
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@ -120,7 +120,7 @@ extern LPALBUFFERDATA qalBufferData;
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extern LPALBUFFERF qalBufferf;
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extern LPALBUFFER3F qalBuffer3f;
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extern LPALBUFFERFV qalBufferfv;
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extern LPALBUFFERF qalBufferi;
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extern LPALBUFFERI qalBufferi;
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extern LPALBUFFER3F qalBuffer3i;
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extern LPALBUFFERFV qalBufferiv;
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extern LPALGETBUFFERF qalGetBufferf;
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@ -34,11 +34,12 @@ extern "C" {
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typedef struct snd_info_s
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{
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int rate;
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int width;
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float width;
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int channels;
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int samples;
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int size;
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int dataofs;
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int dataalign;
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} snd_info_t;
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typedef struct snd_codec_s snd_codec_t;
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@ -622,7 +622,7 @@ int S_MP3_CodecReadStream(snd_stream_t* stream, int bytes, void* buffer)
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mp3info = stream->ptr;
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// Make sure we get complete frames all the way through.
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bytes -= bytes % (stream->info.channels * stream->info.width);
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bytes -= fmod(bytes, (stream->info.channels * stream->info.width));
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if (mp3info->buflen)
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{
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@ -133,6 +133,7 @@ static qboolean S_ReadRIFFHeader(fileHandle_t file, snd_info_t *info)
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char dump[16];
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int bits;
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int fmtlen = 0;
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int bytealign;
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// skip the riff wav header
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FS_Read(dump, 12, file);
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@ -149,17 +150,22 @@ static qboolean S_ReadRIFFHeader(fileHandle_t file, snd_info_t *info)
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info->channels = FGetLittleShort(file);
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info->rate = FGetLittleLong(file);
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FGetLittleLong(file);
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FGetLittleShort(file);
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bytealign = FGetLittleShort(file);
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bits = FGetLittleShort(file);
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if( bits < 8 )
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{
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Com_Printf( S_COLOR_RED "ERROR: Less than 8 bit sound is not supported\n");
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return qfalse;
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}
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//if( bits < 8 )
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//{
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// Com_Printf( S_COLOR_RED "ERROR: Less than 8 bit sound is not supported\n");
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// return qfalse;
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//}
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info->width = bits / 8;
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info->width = bits / 8.0;
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info->dataofs = 0;
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if (bits == 16) {
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info->dataalign = 1;
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} else {
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info->dataalign = (bytealign / info->channels - 4) / 4 * 8 + 1;
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}
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// Skip the rest of the format chunk if required
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if(fmtlen > 16)
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@ -62,6 +62,7 @@ typedef struct {
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int dataofs;
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int datasize;
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int dataalign;
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} wavinfo_t;
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typedef struct sfx_s {
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@ -43,13 +43,13 @@ short int GetLittleShort()
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byte bytes[2];
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} val;
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# ifdef Q3_LITTLE_ENDIAN
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#ifdef Q3_LITTLE_ENDIAN
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val.bytes[0] = data_p[0];
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val.bytes[1] = data_p[1];
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# else
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#else
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val.bytes[0] = data_p[1];
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val.bytes[1] = data_p[0];
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# endif
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#endif
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data_p += sizeof(short);
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return val.value;
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@ -67,17 +67,17 @@ int GetLittleLong()
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byte bytes[4];
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} val;
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# ifdef Q3_LITTLE_ENDIAN
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#ifdef Q3_LITTLE_ENDIAN
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val.bytes[0] = data_p[0];
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val.bytes[1] = data_p[1];
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val.bytes[2] = data_p[2];
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val.bytes[3] = data_p[3];
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# else
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#else
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val.bytes[0] = data_p[3];
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val.bytes[1] = data_p[2];
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val.bytes[2] = data_p[1];
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val.bytes[3] = data_p[0];
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# endif
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#endif
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data_p += sizeof(int);
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return val.value;
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@ -97,13 +97,13 @@ void SetLittleShort(int i)
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val.value = i;
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# ifdef Q3_LITTLE_ENDIAN
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#ifdef Q3_LITTLE_ENDIAN
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data_p[0] = val.bytes[0];
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data_p[1] = val.bytes[1];
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# else
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#else
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data_p[0] = val.bytes[1];
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data_p[1] = val.bytes[0];
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# endif
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#endif
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data_p += sizeof(short);
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}
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@ -122,17 +122,17 @@ void SetLittleLong(int i)
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val.value = i;
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# ifdef Q3_LITTLE_ENDIAN
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#ifdef Q3_LITTLE_ENDIAN
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data_p[0] = val.bytes[0];
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data_p[1] = val.bytes[1];
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data_p[2] = val.bytes[2];
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data_p[3] = val.bytes[3];
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# else
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#else
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data_p[0] = val.bytes[3];
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data_p[1] = val.bytes[2];
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data_p[2] = val.bytes[1];
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data_p[3] = val.bytes[0];
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# endif
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#endif
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data_p += sizeof(int);
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}
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@ -218,6 +218,7 @@ wavinfo_t GetWavinfo(const char *name, byte *wav, int wavlength)
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{
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wavinfo_t info;
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int samples;
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short bytealign;
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memset(&info, 0, sizeof(wavinfo_t));
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@ -247,7 +248,9 @@ wavinfo_t GetWavinfo(const char *name, byte *wav, int wavlength)
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if (info.format == 17) {
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info.channels = GetLittleShort();
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info.rate = (float)GetLittleLong();
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data_p += 6;
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data_p += 4;
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bytealign = GetLittleShort();
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info.width = (float)GetLittleShort() / 8.f;
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data_p += 2;
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@ -270,12 +273,16 @@ wavinfo_t GetWavinfo(const char *name, byte *wav, int wavlength)
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Com_Error(ERR_DROP, "Sound %s has a bad loop length", name);
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}
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info.dataofs = 0;
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info.dataofs = data_p - wav;
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info.datasize = iff_chunk_len - bytealign + info.dataofs;
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info.dataalign = (bytealign / info.channels - 4) / 4 * 8 + 1;
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} else if (info.format == 1) {
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info.channels = GetLittleShort();
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info.rate = (float)GetLittleLong();
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data_p += 6;
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info.width = (float)(GetLittleShort() / 8);
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data_p += 4;
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bytealign = GetLittleShort();
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info.width = (float)GetLittleShort() / 8.f;
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FindChunk("data");
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if (!data_p) {
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@ -292,14 +299,16 @@ wavinfo_t GetWavinfo(const char *name, byte *wav, int wavlength)
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Com_Error(ERR_DROP, "Sound %s has a bad loop length", name);
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}
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info.dataofs = data_p - wav;
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info.dataofs = data_p - wav;
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info.datasize = iff_chunk_len;
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info.dataalign = (bytealign / info.channels - 4) / 4 * 8 + 1;
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// dataalign should always be 1
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assert(info.dataalign == 1);
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} else {
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Com_Printf("Microsoft PCM format only\n");
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return info;
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}
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info.datasize = iff_chunk_len;
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return info;
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}
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@ -310,26 +319,24 @@ DownSampleWav
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*/
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qboolean DownSampleWav(wavinfo_t *info, byte *wav, int wavlength, int newkhz, byte **newdata)
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{
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int newdatasize;
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byte* datap;
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int i;
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int ii;
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int error;
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int width;
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int oldsamples;
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int oldrate;
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int newdatasize;
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byte *datap;
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int i;
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int ii;
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int error;
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int width;
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int oldsamples;
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int oldrate;
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newdatasize = 0;
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datap = &wav[info->dataofs];
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datap = &wav[info->dataofs];
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if (info->channels > 1)
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{
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if (info->channels > 1) {
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Com_DPrintf("Could not downsample WAV file. Stereo WAVs not supported!\n");
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return 0;
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}
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if (info->format != 1 || !info->dataofs)
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{
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if (info->format != 1 || !info->dataofs) {
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Com_DPrintf("Could not downsample WAV file. Not PCM format!\n");
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return 0;
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}
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@ -348,29 +355,27 @@ qboolean DownSampleWav(wavinfo_t *info, byte *wav, int wavlength, int newkhz, by
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}
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}
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oldsamples = info->samples;
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oldrate = info->rate;
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oldsamples = info->samples;
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oldrate = info->rate;
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info->samples = newdatasize / width;
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info->rate = (float)newkhz;
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info->rate = (float)newkhz;
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newdatasize += info->dataofs;
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*newdata = (byte*)Z_TagMalloc(newdatasize, TAG_SOUND);
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*newdata = (byte *)Z_TagMalloc(newdatasize, TAG_SOUND);
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memcpy(*newdata, wav, info->dataofs);
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iff_data = *newdata;
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iff_end = *newdata + newdatasize;
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iff_end = *newdata + newdatasize;
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FindChunk("RIFF");
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if (!data_p || strncmp((const char*)data_p + 8, "WAVE", 4u))
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{
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if (!data_p || strncmp((const char *)data_p + 8, "WAVE", 4u)) {
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Com_DPrintf("Missing RIFF/WAVE chunks\n");
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return 0;
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}
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iff_data = data_p + 12;
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FindChunk("fmt ");
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if (!data_p)
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{
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if (!data_p) {
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Com_DPrintf("Missing fmt chunk\n");
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return 0;
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}
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@ -380,8 +385,7 @@ qboolean DownSampleWav(wavinfo_t *info, byte *wav, int wavlength, int newkhz, by
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data_p += 8;
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FindChunk("data");
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if (!data_p)
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{
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if (!data_p) {
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Com_DPrintf("Missing data chunk\n");
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return 0;
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}
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@ -427,10 +431,10 @@ S_LoadSound
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*/
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qboolean S_LoadSound(const char *fileName, sfx_t *sfx, int streamed, qboolean force_load)
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{
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int size;
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int size;
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fileHandle_t file_handle;
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char tempName[MAX_RES_NAME + 1];
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int realKhz;
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char tempName[MAX_RES_NAME + 1];
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int realKhz;
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sfx->buffer = 0;
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@ -440,10 +444,10 @@ qboolean S_LoadSound(const char *fileName, sfx_t *sfx, int streamed, qboolean fo
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if (streamed) {
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sfx->length = 5000;
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sfx->width = 1;
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sfx->width = 1;
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sfx->iFlags |= SFX_FLAG_STREAMED;
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sfx->time_length = 5000.0;
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sfx->data = NULL;
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sfx->data = NULL;
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return qtrue;
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}
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@ -456,22 +460,20 @@ qboolean S_LoadSound(const char *fileName, sfx_t *sfx, int streamed, qboolean fo
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}
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size = FS_FOpenFileRead(fileName, &file_handle, qfalse, qtrue);
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if (size <= 0)
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{
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if (size <= 0) {
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if (file_handle) {
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FS_FCloseFile(file_handle);
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}
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return qfalse;
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}
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sfx->data = (byte*)Z_TagMalloc(size, TAG_SOUND);
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sfx->data = (byte *)Z_TagMalloc(size, TAG_SOUND);
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FS_Read(sfx->data, size, file_handle);
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FS_FCloseFile(file_handle);
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sfx->info = GetWavinfo(fileName, sfx->data, size);
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if (sfx->info.channels != 1 && !streamed)
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{
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if (sfx->info.channels != 1 && !streamed) {
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Com_Printf("%s is a stereo wav file\n", fileName);
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Z_Free(sfx->data);
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sfx->data = NULL;
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@ -496,8 +498,8 @@ qboolean S_LoadSound(const char *fileName, sfx_t *sfx, int streamed, qboolean fo
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}
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if (!(sfx->iFlags & SFX_FLAG_STREAMED) && realKhz < sfx->info.rate) {
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byte* newdata;
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int newdatasize;
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byte *newdata;
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int newdatasize;
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newdata = NULL;
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if (sfx->iFlags & SFX_FLAG_NO_OFFSET) {
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@ -508,13 +510,13 @@ qboolean S_LoadSound(const char *fileName, sfx_t *sfx, int streamed, qboolean fo
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if (newdatasize) {
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Z_Free(sfx->data);
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sfx->data = newdata;
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sfx->data = newdata;
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sfx->info.datasize = newdatasize;
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}
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}
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sfx->length = sfx->info.samples;
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sfx->width = sfx->info.width;
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sfx->length = sfx->info.samples;
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sfx->width = sfx->info.width;
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sfx->time_length = sfx->info.samples / sfx->info.rate * 1000.f;
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if (sfx->iFlags & SFX_FLAG_STREAMED) {
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@ -540,7 +542,7 @@ S_LoadMP3
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*/
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qboolean S_LoadMP3(const char *fileName, sfx_t *sfx)
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{
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int length;
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int length;
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fileHandle_t file_handle;
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length = FS_FOpenFileRead(fileName, &file_handle, 0, 1);
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@ -552,9 +554,9 @@ qboolean S_LoadMP3(const char *fileName, sfx_t *sfx)
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}
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memset(&sfx->info, 0, sizeof(sfx->info));
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sfx->data = (byte*)Z_TagMalloc(length, TAG_SOUND);
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sfx->data = (byte *)Z_TagMalloc(length, TAG_SOUND);
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sfx->length = length;
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sfx->width = 1;
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sfx->width = 1;
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FS_Read(sfx->data, length, file_handle);
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FS_FCloseFile(file_handle);
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@ -100,6 +100,9 @@ int music_currentsong = 0;
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static qboolean enumeration_ext = qfalse;
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static qboolean enumeration_all_ext = qfalse;
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static qboolean ima4_ext = qfalse;
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static qboolean soft_block_align = qfalse;
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song_t music_songs[MAX_MUSIC_SONGS];
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openal_internal_t openal;
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static float s_fFadeStartTime;
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@ -113,7 +116,7 @@ static int
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S_OPENAL_SpatializeStereoSound(const vec3_t listener_origin, const vec3_t listener_left, const vec3_t origin);
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static void S_OPENAL_reverb(int iChannel, int iReverbType, float fReverbLevel);
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static bool S_OPENAL_LoadMP3_Codec(const char *_path, sfx_t *pSfx);
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static ALuint S_OPENAL_Format(int width, int channels);
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static ALuint S_OPENAL_Format(float width, int channels);
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#define alDieIfError() __alDieIfError(__FILE__, __LINE__)
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@ -449,7 +452,9 @@ S_OPENAL_InitExtensions
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*/
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static bool S_OPENAL_InitExtensions()
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{
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Com_Printf("AL extensions ignored\n");
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ima4_ext = qalIsExtensionPresent("AL_EXT_IMA4");
|
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soft_block_align = qalIsExtensionPresent("AL_SOFT_block_alignment");
|
||||
|
||||
return true;
|
||||
|
||||
extensions_table_t extensions_table[4] = {
|
||||
|
|
@ -612,7 +617,9 @@ qboolean S_OPENAL_Init()
|
|||
}
|
||||
|
||||
for (i = 0; i < MAX_SOUNDSYSTEM_CHANNELS_2D_STREAM; i++) {
|
||||
if (!S_OPENAL_InitChannel(i + MAX_SOUNDSYSTEM_CHANNELS_3D + MAX_SOUNDSYSTEM_CHANNELS_2D, &openal.chan_2D_stream[i])) {
|
||||
if (!S_OPENAL_InitChannel(
|
||||
i + MAX_SOUNDSYSTEM_CHANNELS_3D + MAX_SOUNDSYSTEM_CHANNELS_2D, &openal.chan_2D_stream[i]
|
||||
)) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
|
@ -827,7 +834,8 @@ void S_DumpInfo()
|
|||
S_DumpStatus(
|
||||
"Misc",
|
||||
i,
|
||||
openal.channel[MAX_SOUNDSYSTEM_CHANNELS_3D + MAX_SOUNDSYSTEM_CHANNELS_2D + MAX_SOUNDSYSTEM_CHANNELS_2D_STREAM + i]
|
||||
openal.channel
|
||||
[MAX_SOUNDSYSTEM_CHANNELS_3D + MAX_SOUNDSYSTEM_CHANNELS_2D + MAX_SOUNDSYSTEM_CHANNELS_2D_STREAM + i]
|
||||
);
|
||||
}
|
||||
}
|
||||
|
|
@ -1818,7 +1826,9 @@ static int S_OPENAL_Start2DLoopSound(
|
|||
pChannel->set_gain(fVolumeToPlay);
|
||||
pChannel->start_sample();
|
||||
if (s_show_sounds->integer > 0) {
|
||||
Com_DPrintf("OpenAL: %d (#%i) - %s (vol %f)\n", cl.serverTime, pLoopSound->iChannel, pLoopSound->pSfx->name, fVolume);
|
||||
Com_DPrintf(
|
||||
"OpenAL: %d (#%i) - %s (vol %f)\n", cl.serverTime, pLoopSound->iChannel, pLoopSound->pSfx->name, fVolume
|
||||
);
|
||||
}
|
||||
|
||||
return iChannel;
|
||||
|
|
@ -2138,7 +2148,9 @@ void S_OPENAL_AddLoopSounds(const vec3_t vTempAxis)
|
|||
}
|
||||
|
||||
if (s_show_sounds->integer > 0) {
|
||||
Com_DPrintf("OpenAL: %d (#%i) - started loop - %s\n", cl.serverTime, pLoopSound->iChannel, pLoopSound->pSfx->name);
|
||||
Com_DPrintf(
|
||||
"OpenAL: %d (#%i) - started loop - %s\n", cl.serverTime, pLoopSound->iChannel, pLoopSound->pSfx->name
|
||||
);
|
||||
}
|
||||
|
||||
if (pLoopSound->pSfx->iFlags & (SFX_FLAG_NO_OFFSET)
|
||||
|
|
@ -2991,9 +3003,21 @@ bool openal_channel::set_sfx(sfx_t *pSfx)
|
|||
return false;
|
||||
}
|
||||
|
||||
if (pSfx->info.dataalign > 1 && !soft_block_align) {
|
||||
Com_DPrintf(
|
||||
"OpenAL: Alignment specified but AL doesn't support block alignment (%d).", pSfx->info.dataalign
|
||||
);
|
||||
return false;
|
||||
}
|
||||
|
||||
qalGenBuffers(1, &pSfx->buffer);
|
||||
alDieIfError();
|
||||
|
||||
if (pSfx->info.dataalign > 1) {
|
||||
qalBufferi(pSfx->buffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, pSfx->info.dataalign);
|
||||
alDieIfError();
|
||||
}
|
||||
|
||||
qalBufferData(
|
||||
pSfx->buffer,
|
||||
fmt,
|
||||
|
|
@ -4216,7 +4240,7 @@ int S_CurrentMoviePosition()
|
|||
S_AL_Format
|
||||
=================
|
||||
*/
|
||||
static ALuint S_OPENAL_Format(int width, int channels)
|
||||
static ALuint S_OPENAL_Format(float width, int channels)
|
||||
{
|
||||
ALuint format = AL_FORMAT_MONO16;
|
||||
|
||||
|
|
@ -4233,6 +4257,17 @@ static ALuint S_OPENAL_Format(int width, int channels)
|
|||
} else if (channels == 2) {
|
||||
format = AL_FORMAT_STEREO16;
|
||||
}
|
||||
} else if (width == 0.5) {
|
||||
if (ima4_ext && soft_block_align) {
|
||||
if (channels == 1) {
|
||||
format = AL_FORMAT_MONO_IMA4;
|
||||
} else if (channels == 2) {
|
||||
format = AL_FORMAT_STEREO_IMA4;
|
||||
}
|
||||
} else {
|
||||
// unsupported
|
||||
format = 0;
|
||||
}
|
||||
}
|
||||
|
||||
return format;
|
||||
|
|
@ -4360,6 +4395,11 @@ bool openal_channel_two_d_stream::set_sfx(sfx_t *pSfx)
|
|||
return true;
|
||||
}
|
||||
|
||||
if (stream->info.dataalign > 1 && soft_block_align) {
|
||||
qalBufferi(buffers[currentBuf], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, stream->info.dataalign);
|
||||
alDieIfError();
|
||||
}
|
||||
|
||||
qalBufferData(buffers[currentBuf], pSfx->info.format, rawData, bytesRead, stream->info.rate);
|
||||
alDieIfError();
|
||||
|
||||
|
|
@ -4499,6 +4539,11 @@ void openal_channel_two_d_stream::update()
|
|||
}
|
||||
}
|
||||
|
||||
if (stream->info.dataalign > 1 && soft_block_align) {
|
||||
qalBufferi(buffers[currentBuf], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, stream->info.dataalign);
|
||||
alDieIfError();
|
||||
}
|
||||
|
||||
qalBufferData(buffers[currentBuf], format, rawData, bytesRead, stream->info.rate);
|
||||
alDieIfError();
|
||||
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue