diff options
Diffstat (limited to 'examples/alstream.c')
| -rw-r--r-- | examples/alstream.c | 134 |
1 files changed, 76 insertions, 58 deletions
diff --git a/examples/alstream.c b/examples/alstream.c index 2972d375..68115e8d 100644 --- a/examples/alstream.c +++ b/examples/alstream.c @@ -30,17 +30,21 @@ #include <signal.h> #include <assert.h> +#include <SDL_sound.h> + #include "AL/al.h" #include "AL/alc.h" #include "AL/alext.h" #include "common/alhelpers.h" -#include "common/sdl_sound.h" - -static LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples; -static LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT; +#ifndef SDL_AUDIO_MASK_BITSIZE +#define SDL_AUDIO_MASK_BITSIZE (0xFF) +#endif +#ifndef SDL_AUDIO_BITSIZE +#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) +#endif /* Define the number of buffers and buffer size (in milliseconds) to use. 4 * buffers with 200ms each gives a nice per-chunk size, and lets the queue last @@ -54,13 +58,11 @@ typedef struct StreamPlayer { ALuint source; /* Handle for the audio file */ - FilePtr file; + Sound_Sample *sample; /* The format of the output stream */ ALenum format; - ALenum channels; - ALenum type; - ALuint rate; + ALsizei srate; } StreamPlayer; static StreamPlayer *NewPlayer(void); @@ -77,11 +79,9 @@ static StreamPlayer *NewPlayer(void) { StreamPlayer *player; - player = malloc(sizeof(*player)); + player = calloc(1, sizeof(*player)); assert(player != NULL); - memset(player, 0, sizeof(*player)); - /* Generate the buffers and source */ alGenBuffers(NUM_BUFFERS, player->buffers); assert(alGetError() == AL_NO_ERROR && "Could not create buffers"); @@ -119,37 +119,63 @@ static void DeletePlayer(StreamPlayer *player) * it will be closed first. */ static int OpenPlayerFile(StreamPlayer *player, const char *filename) { + Uint32 frame_size; + ClosePlayerFile(player); /* Open the file and get the first stream from it */ - player->file = openAudioFile(filename, BUFFER_TIME_MS); - if(!player->file) + player->sample = Sound_NewSampleFromFile(filename, NULL, 0); + if(!player->sample) { fprintf(stderr, "Could not open audio in %s\n", filename); goto error; } /* Get the stream format, and figure out the OpenAL format */ - if(getAudioInfo(player->file, &player->rate, &player->channels, &player->type) != 0) + if(player->sample->actual.channels == 1) { - fprintf(stderr, "Error getting audio info for %s\n", filename); - goto error; + if(player->sample->actual.format == AUDIO_U8) + player->format = AL_FORMAT_MONO8; + else if(player->sample->actual.format == AUDIO_S16SYS) + player->format = AL_FORMAT_MONO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format); + goto error; + } } - - player->format = GetFormat(player->channels, player->type, alIsBufferFormatSupportedSOFT); - if(player->format == 0) + else if(player->sample->actual.channels == 2) + { + if(player->sample->actual.format == AUDIO_U8) + player->format = AL_FORMAT_STEREO8; + else if(player->sample->actual.format == AUDIO_S16SYS) + player->format = AL_FORMAT_STEREO16; + else + { + fprintf(stderr, "Unsupported sample format: 0x%04x\n", player->sample->actual.format); + goto error; + } + } + else { - fprintf(stderr, "Unsupported format (%s, %s) for %s\n", - ChannelsName(player->channels), TypeName(player->type), - filename); + fprintf(stderr, "Unsupported channel count: %d\n", player->sample->actual.channels); goto error; } + player->srate = player->sample->actual.rate; + + frame_size = player->sample->actual.channels * + SDL_AUDIO_BITSIZE(player->sample->actual.format) / 8; + + /* Set the buffer size, given the desired millisecond length. */ + Sound_SetBufferSize(player->sample, (Uint32)((Uint64)player->srate*BUFFER_TIME_MS/1000) * + frame_size); return 1; error: - closeAudioFile(player->file); - player->file = NULL; + if(player->sample) + Sound_FreeSample(player->sample); + player->sample = NULL; return 0; } @@ -157,8 +183,9 @@ error: /* Closes the audio file stream */ static void ClosePlayerFile(StreamPlayer *player) { - closeAudioFile(player->file); - player->file = NULL; + if(player->sample) + Sound_FreeSample(player->sample); + player->sample = NULL; } @@ -174,16 +201,12 @@ static int StartPlayer(StreamPlayer *player) /* Fill the buffer queue */ for(i = 0;i < NUM_BUFFERS;i++) { - uint8_t *data; - size_t got; - /* Get some data to give it to the buffer */ - data = getAudioData(player->file, &got); - if(!data) break; + Uint32 slen = Sound_Decode(player->sample); + if(slen == 0) break; - alBufferSamplesSOFT(player->buffers[i], player->rate, player->format, - BytesToFrames(got, player->channels, player->type), - player->channels, player->type, data); + alBufferData(player->buffers[i], player->format, + player->sample->buffer, slen, player->srate); } if(alGetError() != AL_NO_ERROR) { @@ -220,20 +243,21 @@ static int UpdatePlayer(StreamPlayer *player) while(processed > 0) { ALuint bufid; - uint8_t *data; - size_t got; + Uint32 slen; alSourceUnqueueBuffers(player->source, 1, &bufid); processed--; + if((player->sample->flags&(SOUND_SAMPLEFLAG_EOF|SOUND_SAMPLEFLAG_ERROR))) + continue; + /* Read the next chunk of data, refill the buffer, and queue it * back on the source */ - data = getAudioData(player->file, &got); - if(data != NULL) + slen = Sound_Decode(player->sample); + if(slen > 0) { - alBufferSamplesSOFT(bufid, player->rate, player->format, - BytesToFrames(got, player->channels, player->type), - player->channels, player->type, data); + alBufferData(bufid, player->format, player->sample->buffer, slen, + player->srate); alSourceQueueBuffers(player->source, 1, &bufid); } if(alGetError() != AL_NO_ERROR) @@ -270,29 +294,23 @@ int main(int argc, char **argv) StreamPlayer *player; int i; - /* Print out usage if no file was specified */ + /* Print out usage if no arguments were specified */ if(argc < 2) { - fprintf(stderr, "Usage: %s <filenames...>\n", argv[0]); + fprintf(stderr, "Usage: %s [-device <name>] <filenames...>\n", argv[0]); return 1; } - if(InitAL() != 0) + argv++; argc--; + if(InitAL(&argv, &argc) != 0) return 1; - if(alIsExtensionPresent("AL_SOFT_buffer_samples")) - { - printf("AL_SOFT_buffer_samples supported!\n"); - alBufferSamplesSOFT = alGetProcAddress("alBufferSamplesSOFT"); - alIsBufferFormatSupportedSOFT = alGetProcAddress("alIsBufferFormatSupportedSOFT"); - } - else - printf("AL_SOFT_buffer_samples not supported\n"); + Sound_Init(); player = NewPlayer(); /* Play each file listed on the command line */ - for(i = 1;i < argc;i++) + for(i = 0;i < argc;i++) { const char *namepart; @@ -306,9 +324,8 @@ int main(int argc, char **argv) else namepart = argv[i]; - printf("Playing: %s (%s, %s, %dhz)\n", namepart, - TypeName(player->type), ChannelsName(player->channels), - player->rate); + printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format), + player->srate); fflush(stdout); if(!StartPlayer(player)) @@ -318,17 +335,18 @@ int main(int argc, char **argv) } while(UpdatePlayer(player)) - Sleep(10); + al_nssleep(10000000); /* All done with this file. Close it and go to the next */ ClosePlayerFile(player); } printf("Done.\n"); - /* All files done. Delete the player, and close OpenAL */ + /* All files done. Delete the player, and close down SDL_sound and OpenAL */ DeletePlayer(player); player = NULL; + Sound_Quit(); CloseAL(); return 0; |