mirror of
https://github.com/TorqueGameEngines/Torque3D.git
synced 2026-07-14 16:14:38 +00:00
Initial commit
added libraries: opus flac libsndfile updated: libvorbis libogg openal - Everything works as expected for now. Bare in mind libsndfile needed the check for whether or not it could find the xiph libraries removed in order for this to work.
This commit is contained in:
marauder2k7
parent
05a083ca6f
commit
a745fc3757
1954 changed files with 431332 additions and 21037 deletions
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@ -39,20 +39,31 @@
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/* Define the number of buffers and buffer size (in milliseconds) to use. 4
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* buffers with 8192 samples each gives a nice per-chunk size, and lets the
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* queue last for almost one second at 44.1khz. */
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* buffers at 200ms each gives a nice per-chunk size, and lets the queue last
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* for almost one second.
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*/
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#define NUM_BUFFERS 4
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#define BUFFER_SAMPLES 8192
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#define BUFFER_MILLISEC 200
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typedef enum SampleType {
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Int16, Float, IMA4, MSADPCM
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} SampleType;
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typedef struct StreamPlayer {
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/* These are the buffers and source to play out through OpenAL with */
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/* These are the buffers and source to play out through OpenAL with. */
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ALuint buffers[NUM_BUFFERS];
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ALuint source;
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/* Handle for the audio file */
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SNDFILE *sndfile;
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SF_INFO sfinfo;
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short *membuf;
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void *membuf;
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/* The sample type and block/frame size being read for the buffer. */
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SampleType sample_type;
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int byteblockalign;
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int sampleblockalign;
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int block_count;
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/* The format of the output stream (sample rate is in sfinfo) */
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ALenum format;
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@ -67,7 +78,8 @@ static int UpdatePlayer(StreamPlayer *player);
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/* Creates a new player object, and allocates the needed OpenAL source and
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* buffer objects. Error checking is simplified for the purposes of this
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* example, and will cause an abort if needed. */
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* example, and will cause an abort if needed.
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*/
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static StreamPlayer *NewPlayer(void)
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{
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StreamPlayer *player;
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@ -112,7 +124,7 @@ static void DeletePlayer(StreamPlayer *player)
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* it will be closed first. */
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static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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{
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size_t frame_size;
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int byteblockalign=0, splblockalign=0;
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ClosePlayerFile(player);
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@ -124,20 +136,151 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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return 0;
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}
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/* Get the sound format, and figure out the OpenAL format */
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/* Detect a suitable format to load. Formats like Vorbis and Opus use float
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* natively, so load as float to avoid clipping when possible. Formats
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* larger than 16-bit can also use float to preserve a bit more precision.
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*/
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switch((player->sfinfo.format&SF_FORMAT_SUBMASK))
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{
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case SF_FORMAT_PCM_24:
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case SF_FORMAT_PCM_32:
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case SF_FORMAT_FLOAT:
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case SF_FORMAT_DOUBLE:
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case SF_FORMAT_VORBIS:
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case SF_FORMAT_OPUS:
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case SF_FORMAT_ALAC_20:
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case SF_FORMAT_ALAC_24:
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case SF_FORMAT_ALAC_32:
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case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
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case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
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case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
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if(alIsExtensionPresent("AL_EXT_FLOAT32"))
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player->sample_type = Float;
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break;
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case SF_FORMAT_IMA_ADPCM:
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/* ADPCM formats require setting a block alignment as specified in the
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* file, which needs to be read from the wave 'fmt ' chunk manually
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* since libsndfile doesn't provide it in a format-agnostic way.
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*/
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if(player->sfinfo.channels <= 2
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&& (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
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&& alIsExtensionPresent("AL_EXT_IMA4")
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&& alIsExtensionPresent("AL_SOFT_block_alignment"))
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player->sample_type = IMA4;
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break;
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case SF_FORMAT_MS_ADPCM:
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if(player->sfinfo.channels <= 2
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&& (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
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&& alIsExtensionPresent("AL_SOFT_MSADPCM")
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&& alIsExtensionPresent("AL_SOFT_block_alignment"))
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player->sample_type = MSADPCM;
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break;
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}
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if(player->sample_type == IMA4 || player->sample_type == MSADPCM)
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{
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/* For ADPCM, lookup the wave file's "fmt " chunk, which is a
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* WAVEFORMATEX-based structure for the audio format.
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*/
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SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL };
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SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(player->sndfile, &inf);
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/* If there's an issue getting the chunk or block alignment, load as
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* 16-bit and have libsndfile do the conversion.
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*/
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if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
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player->sample_type = Int16;
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else
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{
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ALubyte *fmtbuf = calloc(inf.datalen, 1);
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inf.data = fmtbuf;
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if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
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player->sample_type = Int16;
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else
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{
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/* Read the nBlockAlign field, and convert from bytes- to
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* samples-per-block (verifying it's valid by converting back
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* and comparing to the original value).
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*/
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byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8);
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if(player->sample_type == IMA4)
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{
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splblockalign = (byteblockalign/player->sfinfo.channels - 4)/4*8 + 1;
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if(splblockalign < 1
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|| ((splblockalign-1)/2 + 4)*player->sfinfo.channels != byteblockalign)
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player->sample_type = Int16;
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}
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else
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{
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splblockalign = (byteblockalign/player->sfinfo.channels - 7)*2 + 2;
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if(splblockalign < 2
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|| ((splblockalign-2)/2 + 7)*player->sfinfo.channels != byteblockalign)
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player->sample_type = Int16;
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}
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}
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free(fmtbuf);
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}
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}
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if(player->sample_type == Int16)
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{
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player->sampleblockalign = 1;
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player->byteblockalign = player->sfinfo.channels * 2;
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}
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else if(player->sample_type == Float)
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{
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player->sampleblockalign = 1;
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player->byteblockalign = player->sfinfo.channels * 4;
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}
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else
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{
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player->sampleblockalign = splblockalign;
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player->byteblockalign = byteblockalign;
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}
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/* Figure out the OpenAL format from the file and desired sample type. */
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player->format = AL_NONE;
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if(player->sfinfo.channels == 1)
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player->format = AL_FORMAT_MONO16;
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_MONO16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_MONO_FLOAT32;
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else if(player->sample_type == IMA4)
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player->format = AL_FORMAT_MONO_IMA4;
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else if(player->sample_type == MSADPCM)
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player->format = AL_FORMAT_MONO_MSADPCM_SOFT;
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}
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else if(player->sfinfo.channels == 2)
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player->format = AL_FORMAT_STEREO16;
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_STEREO16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_STEREO_FLOAT32;
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else if(player->sample_type == IMA4)
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player->format = AL_FORMAT_STEREO_IMA4;
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else if(player->sample_type == MSADPCM)
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player->format = AL_FORMAT_STEREO_MSADPCM_SOFT;
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}
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else if(player->sfinfo.channels == 3)
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{
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if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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player->format = AL_FORMAT_BFORMAT2D_16;
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_BFORMAT2D_16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
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}
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}
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else if(player->sfinfo.channels == 4)
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{
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if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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player->format = AL_FORMAT_BFORMAT3D_16;
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_BFORMAT3D_16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
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}
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}
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if(!player->format)
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{
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@ -147,8 +290,9 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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return 0;
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}
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frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(short);
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player->membuf = malloc(frame_size);
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player->block_count = player->sfinfo.samplerate / player->sampleblockalign;
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player->block_count = player->block_count * BUFFER_MILLISEC / 1000;
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player->membuf = malloc((size_t)(player->block_count * player->byteblockalign));
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return 1;
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}
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@ -162,6 +306,15 @@ static void ClosePlayerFile(StreamPlayer *player)
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free(player->membuf);
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player->membuf = NULL;
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if(player->sampleblockalign > 1)
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{
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ALsizei i;
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for(i = 0;i < NUM_BUFFERS;i++)
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alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0);
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player->sampleblockalign = 0;
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player->byteblockalign = 0;
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}
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}
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@ -177,11 +330,35 @@ static int StartPlayer(StreamPlayer *player)
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/* Fill the buffer queue */
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for(i = 0;i < NUM_BUFFERS;i++)
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{
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/* Get some data to give it to the buffer */
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sf_count_t slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
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if(slen < 1) break;
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sf_count_t slen;
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/* Get some data to give it to the buffer */
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if(player->sample_type == Int16)
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{
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slen = sf_readf_short(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen < 1) break;
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slen *= player->byteblockalign;
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}
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else if(player->sample_type == Float)
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{
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slen = sf_readf_float(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen < 1) break;
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slen *= player->byteblockalign;
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}
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else
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{
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slen = sf_read_raw(player->sndfile, player->membuf,
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player->block_count * player->byteblockalign);
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if(slen > 0) slen -= slen%player->byteblockalign;
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if(slen < 1) break;
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}
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if(player->sampleblockalign > 1)
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alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
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player->sampleblockalign);
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slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
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alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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}
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@ -227,10 +404,27 @@ static int UpdatePlayer(StreamPlayer *player)
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/* Read the next chunk of data, refill the buffer, and queue it
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* back on the source */
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slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES);
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if(player->sample_type == Int16)
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{
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slen = sf_readf_short(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen > 0) slen *= player->byteblockalign;
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}
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else if(player->sample_type == Float)
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{
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slen = sf_readf_float(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen > 0) slen *= player->byteblockalign;
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}
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else
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{
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slen = sf_read_raw(player->sndfile, player->membuf,
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player->block_count * player->byteblockalign);
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if(slen > 0) slen -= slen%player->byteblockalign;
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}
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if(slen > 0)
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{
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slen *= player->sfinfo.channels * (sf_count_t)sizeof(short);
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alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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alSourceQueueBuffers(player->source, 1, &bufid);
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