diff options
author | Chris Robinson <[email protected]> | 2023-02-17 13:50:12 -0800 |
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committer | Chris Robinson <[email protected]> | 2023-02-17 13:50:12 -0800 |
commit | 7e00d4a00243e417ff59d3be001e1180ff55b72b (patch) | |
tree | 454ed450839181ee1ef42527c4b62f881ada32a1 | |
parent | 08bfbaa5336d183199c83a49365976da088b4bbf (diff) |
Support float and ADPCM formats in alstream
-rw-r--r-- | examples/alstream.c | 229 |
1 files changed, 210 insertions, 19 deletions
diff --git a/examples/alstream.c b/examples/alstream.c index 2e427a32..b561406c 100644 --- a/examples/alstream.c +++ b/examples/alstream.c @@ -39,20 +39,31 @@ /* Define the number of buffers and buffer size (in milliseconds) to use. 4 - * buffers with 8192 samples each gives a nice per-chunk size, and lets the - * queue last for almost one second at 44.1khz. */ + * buffers at 200ms each gives a nice per-chunk size, and lets the queue last + * for almost one second. + */ #define NUM_BUFFERS 4 -#define BUFFER_SAMPLES 8192 +#define BUFFER_MILLISEC 200 + +typedef enum SampleType { + Int16, Float, IMA4, MSADPCM +} SampleType; typedef struct StreamPlayer { - /* These are the buffers and source to play out through OpenAL with */ + /* These are the buffers and source to play out through OpenAL with. */ ALuint buffers[NUM_BUFFERS]; ALuint source; /* Handle for the audio file */ SNDFILE *sndfile; SF_INFO sfinfo; - short *membuf; + void *membuf; + + /* The sample type and block/frame size being read for the buffer. */ + SampleType sample_type; + int byteblockalign; + int sampleblockalign; + int block_count; /* The format of the output stream (sample rate is in sfinfo) */ ALenum format; @@ -67,7 +78,8 @@ static int UpdatePlayer(StreamPlayer *player); /* Creates a new player object, and allocates the needed OpenAL source and * buffer objects. Error checking is simplified for the purposes of this - * example, and will cause an abort if needed. */ + * example, and will cause an abort if needed. + */ static StreamPlayer *NewPlayer(void) { StreamPlayer *player; @@ -112,7 +124,7 @@ static void DeletePlayer(StreamPlayer *player) * it will be closed first. */ static int OpenPlayerFile(StreamPlayer *player, const char *filename) { - size_t frame_size; + int byteblockalign=0, splblockalign=0; ClosePlayerFile(player); @@ -124,20 +136,148 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename) return 0; } - /* Get the sound format, and figure out the OpenAL format */ + /* Detect a suitable format to load. Formats like Vorbis and Opus use float + * natively, so load as float to avoid clipping when possible. Formats + * larger than 16-bit can also use float to preserve a bit more precision. + */ + switch((player->sfinfo.format&SF_FORMAT_SUBMASK)) + { + case SF_FORMAT_PCM_24: + case SF_FORMAT_PCM_32: + case SF_FORMAT_FLOAT: + case SF_FORMAT_DOUBLE: + case SF_FORMAT_VORBIS: + case SF_FORMAT_OPUS: + case SF_FORMAT_MPEG_LAYER_I: + case SF_FORMAT_MPEG_LAYER_II: + case SF_FORMAT_MPEG_LAYER_III: + if(alIsExtensionPresent("AL_EXT_FLOAT32")) + player->sample_type = Float; + break; + case SF_FORMAT_IMA_ADPCM: + /* ADPCM formats require setting a block alignment as specified in the + * file, which needs to be read from the wave 'fmt ' chunk manually + * since libsndfile doesn't provide it in a format-agnostic way. + */ + if(player->sfinfo.channels <= 2 + && (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV + && alIsExtensionPresent("AL_EXT_IMA4") + && alIsExtensionPresent("AL_SOFT_block_alignment")) + player->sample_type = IMA4; + break; + case SF_FORMAT_MS_ADPCM: + if(player->sfinfo.channels <= 2 + && (player->sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV + && alIsExtensionPresent("AL_SOFT_MSADPCM") + && alIsExtensionPresent("AL_SOFT_block_alignment")) + player->sample_type = MSADPCM; + break; + } + + if(player->sample_type == IMA4 || player->sample_type == MSADPCM) + { + /* For ADPCM, lookup the wave file's "fmt " chunk, which is a + * WAVEFORMATEX-based structure for the audio format. + */ + SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL }; + SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(player->sndfile, &inf); + + /* If there's an issue getting the chunk or block alignment, load as + * 16-bit and have libsndfile do the conversion. + */ + if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR) + player->sample_type = Int16; + else + { + ALubyte *fmtbuf = calloc(inf.datalen, 1); + inf.data = fmtbuf; + if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR) + player->sample_type = Int16; + else + { + /* Read the nBlockAlign field, and convert from bytes- to + * samples-per-block (verifying it's valid by converting back + * and comparing to the original value). + */ + byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8); + if(player->sample_type == IMA4) + { + splblockalign = (byteblockalign/player->sfinfo.channels - 4)/4*8 + 1; + if(splblockalign < 1 + || ((splblockalign-1)/2 + 4)*player->sfinfo.channels != byteblockalign) + player->sample_type = Int16; + } + else + { + splblockalign = (byteblockalign/player->sfinfo.channels - 7)*2 + 2; + if(splblockalign < 2 + || ((splblockalign-2)/2 + 7)*player->sfinfo.channels != byteblockalign) + player->sample_type = Int16; + } + } + free(fmtbuf); + } + } + + if(player->sample_type == Int16) + { + player->sampleblockalign = 1; + player->byteblockalign = player->sfinfo.channels * 2; + } + else if(player->sample_type == Float) + { + player->sampleblockalign = 1; + player->byteblockalign = player->sfinfo.channels * 4; + } + else + { + player->sampleblockalign = splblockalign; + player->byteblockalign = byteblockalign; + } + + /* Figure out the OpenAL format from the file and desired sample type. */ + player->format = AL_NONE; if(player->sfinfo.channels == 1) - player->format = AL_FORMAT_MONO16; + { + if(player->sample_type == Int16) + player->format = AL_FORMAT_MONO16; + else if(player->sample_type == Float) + player->format = AL_FORMAT_MONO_FLOAT32; + else if(player->sample_type == IMA4) + player->format = AL_FORMAT_MONO_IMA4; + else if(player->sample_type == MSADPCM) + player->format = AL_FORMAT_MONO_MSADPCM_SOFT; + } else if(player->sfinfo.channels == 2) - player->format = AL_FORMAT_STEREO16; + { + if(player->sample_type == Int16) + player->format = AL_FORMAT_STEREO16; + else if(player->sample_type == Float) + player->format = AL_FORMAT_STEREO_FLOAT32; + else if(player->sample_type == IMA4) + player->format = AL_FORMAT_STEREO_IMA4; + else if(player->sample_type == MSADPCM) + player->format = AL_FORMAT_STEREO_MSADPCM_SOFT; + } else if(player->sfinfo.channels == 3) { if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT) - player->format = AL_FORMAT_BFORMAT2D_16; + { + if(player->sample_type == Int16) + player->format = AL_FORMAT_BFORMAT2D_16; + else if(player->sample_type == Float) + player->format = AL_FORMAT_BFORMAT2D_FLOAT32; + } } else if(player->sfinfo.channels == 4) { if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT) - player->format = AL_FORMAT_BFORMAT3D_16; + { + if(player->sample_type == Int16) + player->format = AL_FORMAT_BFORMAT3D_16; + else if(player->sample_type == Float) + player->format = AL_FORMAT_BFORMAT3D_FLOAT32; + } } if(!player->format) { @@ -147,8 +287,9 @@ static int OpenPlayerFile(StreamPlayer *player, const char *filename) return 0; } - frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(short); - player->membuf = malloc(frame_size); + player->block_count = player->sfinfo.samplerate / player->sampleblockalign; + player->block_count = player->block_count * BUFFER_MILLISEC / 1000; + player->membuf = malloc((size_t)(player->block_count * player->byteblockalign)); return 1; } @@ -162,6 +303,15 @@ static void ClosePlayerFile(StreamPlayer *player) free(player->membuf); player->membuf = NULL; + + if(player->sampleblockalign > 1) + { + ALsizei i; + for(i = 0;i < NUM_BUFFERS;i++) + alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0); + player->sampleblockalign = 0; + player->byteblockalign = 0; + } } @@ -177,11 +327,35 @@ static int StartPlayer(StreamPlayer *player) /* Fill the buffer queue */ for(i = 0;i < NUM_BUFFERS;i++) { + sf_count_t slen; + /* Get some data to give it to the buffer */ - sf_count_t slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES); - if(slen < 1) break; + if(player->sample_type == Int16) + { + slen = sf_readf_short(player->sndfile, player->membuf, + player->block_count * player->sampleblockalign); + if(slen < 1) break; + slen *= player->byteblockalign; + } + else if(player->sample_type == Float) + { + slen = sf_readf_float(player->sndfile, player->membuf, + player->block_count * player->sampleblockalign); + if(slen < 1) break; + slen *= player->byteblockalign; + } + else + { + slen = sf_read_raw(player->sndfile, player->membuf, + player->block_count * player->byteblockalign); + if(slen > 0) slen -= slen%player->byteblockalign; + if(slen < 1) break; + } + + if(player->sampleblockalign > 1) + alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, + player->sampleblockalign); - slen *= player->sfinfo.channels * (sf_count_t)sizeof(short); alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen, player->sfinfo.samplerate); } @@ -227,10 +401,27 @@ static int UpdatePlayer(StreamPlayer *player) /* Read the next chunk of data, refill the buffer, and queue it * back on the source */ - slen = sf_readf_short(player->sndfile, player->membuf, BUFFER_SAMPLES); + if(player->sample_type == Int16) + { + slen = sf_readf_short(player->sndfile, player->membuf, + player->block_count * player->sampleblockalign); + if(slen > 0) slen *= player->byteblockalign; + } + else if(player->sample_type == Float) + { + slen = sf_readf_float(player->sndfile, player->membuf, + player->block_count * player->sampleblockalign); + if(slen > 0) slen *= player->byteblockalign; + } + else + { + slen = sf_read_raw(player->sndfile, player->membuf, + player->block_count * player->byteblockalign); + if(slen > 0) slen -= slen%player->byteblockalign; + } + if(slen > 0) { - slen *= player->sfinfo.channels * (sf_count_t)sizeof(short); alBufferData(bufid, player->format, player->membuf, (ALsizei)slen, player->sfinfo.samplerate); alSourceQueueBuffers(player->source, 1, &bufid); 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