mirror of https://github.com/mpv-player/mpv
ao/openal: Use only one source for audio output
Floating point audio not supported on this commit.
This commit is contained in:
LAGonauta
Jan Ekström
parent
9efb0278e7
commit
c59ebbe399
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@ -59,11 +59,11 @@
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#define MAX_CHANS MP_NUM_CHANNELS
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#define NUM_BUF 128
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#define CHUNK_SAMPLES 256
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static ALuint buffers[MAX_CHANS][NUM_BUF];
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static ALuint sources[MAX_CHANS];
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static ALuint buffers[NUM_BUF];
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static ALuint source;
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static int cur_buf[MAX_CHANS];
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static int unqueue_buf[MAX_CHANS];
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static int cur_buf;
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static int unqueue_buf;
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static struct ao *ao_data;
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@ -113,19 +113,124 @@ static const struct speaker speaker_pos[] = {
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{-1},
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};
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static ALenum get_al_format(int format)
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static enum af_format get_af_format(int format)
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{
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switch (format) {
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case AF_FORMAT_U8P: return AL_FORMAT_MONO8;
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case AF_FORMAT_S16P: return AL_FORMAT_MONO16;
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case AF_FORMAT_FLOATP:
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case AF_FORMAT_U8:
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if (alGetEnumValue("AL_FORMAT_MONO8"))
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return AL_TRUE;
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break;
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case AF_FORMAT_S16:
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if (alGetEnumValue("AL_FORMAT_MONO16"))
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return AL_TRUE;
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break;
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case AF_FORMAT_S32:
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if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
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return AL_TRUE;
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break;
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case AF_FORMAT_FLOAT:
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if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
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return AL_FORMAT_MONO_FLOAT32;
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return AL_TRUE;
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break;
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case AF_FORMAT_DOUBLEP:
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case AF_FORMAT_DOUBLE:
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if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE)
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return AL_FORMAT_MONO_DOUBLE_EXT;
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return AL_TRUE;
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break;
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}
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return AL_FALSE;
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}
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static ALenum get_al_format(struct ao *ao, int format)
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{
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switch (format) {
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case AF_FORMAT_U8:
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switch (ao->channels.num) {
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case 8:
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if (alGetEnumValue("AL_FORMAT_71CHN8")) {
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return alGetEnumValue("AL_FORMAT_71CHN8");
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}
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case 7:
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if (alGetEnumValue("AL_FORMAT_61CHN8")) {
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return alGetEnumValue("AL_FORMAT_61CHN8");
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}
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case 6:
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if (alGetEnumValue("AL_FORMAT_51CHN8")) {
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return alGetEnumValue("AL_FORMAT_51CHN8");
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}
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case 4:
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if (alGetEnumValue("AL_FORMAT_QUAD8")) {
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return alGetEnumValue("AL_FORMAT_QUAD8");
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}
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case 2:
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if (alGetEnumValue("AL_FORMAT_STEREO8")) {
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return alGetEnumValue("AL_FORMAT_STEREO8");
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}
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default:
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return alGetEnumValue("AL_FORMAT_MONO8");
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}
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case AF_FORMAT_S16:
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switch (ao->channels.num) {
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case 8:
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if (alGetEnumValue("AL_FORMAT_71CHN16")) {
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return alGetEnumValue("AL_FORMAT_71CHN16");
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}
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case 7:
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if (alGetEnumValue("AL_FORMAT_61CHN16")) {
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return alGetEnumValue("AL_FORMAT_61CHN16");
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}
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case 6:
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if (alGetEnumValue("AL_FORMAT_51CHN16")) {
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return alGetEnumValue("AL_FORMAT_51CHN16");
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}
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case 4:
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if (alGetEnumValue("AL_FORMAT_QUAD16")) {
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return alGetEnumValue("AL_FORMAT_QUAD16");
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}
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case 2:
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if (alGetEnumValue("AL_FORMAT_STEREO16")) {
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return alGetEnumValue("AL_FORMAT_STEREO16");
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}
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default:
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return alGetEnumValue("AL_FORMAT_MONO16");
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}
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case AF_FORMAT_S32:
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if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL) {
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switch (ao->channels.num) {
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case 8:
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if (alGetEnumValue("AL_FORMAT_71CHN32")) {
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return alGetEnumValue("AL_FORMAT_71CHN32");
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}
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break;
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case 7:
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if (alGetEnumValue("AL_FORMAT_61CHN32")) {
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return alGetEnumValue("AL_FORMAT_61CHN32");
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}
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break;
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case 6:
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if (alGetEnumValue("AL_FORMAT_51CHN32")) {
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return alGetEnumValue("AL_FORMAT_51CHN32");
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}
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break;
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case 4:
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if (alGetEnumValue("AL_FORMAT_QUAD32")) {
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return alGetEnumValue("AL_FORMAT_QUAD32");
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}
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break;
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case 2:
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if (alGetEnumValue("AL_FORMAT_STEREO32")) {
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return alGetEnumValue("AL_FORMAT_STEREO32");
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}
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default:
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return alGetEnumValue("AL_FORMAT_MONO32");
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}
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}
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}
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return AL_FALSE;
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}
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@ -133,9 +238,14 @@ static ALenum get_al_format(int format)
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// close audio device
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static void uninit(struct ao *ao)
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{
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alSourceStop(source);
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alSourcei(source, AL_BUFFER, 0);
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alDeleteBuffers(NUM_BUF, buffers);
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alDeleteSources(1, &source);
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ALCcontext *ctx = alcGetCurrentContext();
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ALCdevice *dev = alcGetContextsDevice(ctx);
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reset(ao);
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alcMakeContextCurrent(NULL);
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alcDestroyContext(ctx);
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alcCloseDevice(dev);
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@ -184,14 +294,12 @@ static int init(struct ao *ao)
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alcMakeContextCurrent(ctx);
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alListenerfv(AL_POSITION, position);
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alListenerfv(AL_ORIENTATION, direction);
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alGenSources(ao->channels.num, sources);
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for (i = 0; i < ao->channels.num; i++) {
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cur_buf[i] = 0;
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unqueue_buf[i] = 0;
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alGenBuffers(NUM_BUF, buffers[i]);
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alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
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alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
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}
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alGenSources(1, &source);
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cur_buf = 0;
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unqueue_buf = 0;
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alGenBuffers(NUM_BUF, buffers);
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alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
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if (alcGetError(dev) == ALC_NO_ERROR && freq)
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ao->samplerate = freq;
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@ -200,7 +308,7 @@ static int init(struct ao *ao)
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int try_formats[AF_FORMAT_COUNT + 1];
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af_get_best_sample_formats(ao->format, try_formats);
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for (int n = 0; try_formats[n]; n++) {
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p->al_format = get_al_format(try_formats[n]);
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p->al_format = get_al_format(ao, try_formats[n]);
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if (p->al_format != AL_FALSE) {
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ao->format = try_formats[n];
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break;
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@ -225,30 +333,26 @@ err_out:
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static void drain(struct ao *ao)
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{
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ALint state;
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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while (state == AL_PLAYING) {
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mp_sleep_us(10000);
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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}
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}
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static void unqueue_buffers(void)
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{
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ALint p;
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int s;
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for (s = 0; s < ao_data->channels.num; s++) {
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int till_wrap = NUM_BUF - unqueue_buf[s];
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alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
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if (p >= till_wrap) {
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alSourceUnqueueBuffers(sources[s], till_wrap,
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&buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] = 0;
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p -= till_wrap;
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}
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if (p) {
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alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] += p;
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}
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int till_wrap = NUM_BUF - unqueue_buf;
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alGetSourcei(source, AL_BUFFERS_PROCESSED, &p);
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if (p >= till_wrap) {
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alSourceUnqueueBuffers(source, till_wrap, &buffers[unqueue_buf]);
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unqueue_buf = 0;
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p -= till_wrap;
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}
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if (p) {
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alSourceUnqueueBuffers(source, p, &buffers[unqueue_buf]);
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unqueue_buf += p;
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}
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}
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@ -257,7 +361,7 @@ static void unqueue_buffers(void)
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*/
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static void reset(struct ao *ao)
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{
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alSourceStopv(ao->channels.num, sources);
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alSourceStop(source);
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unqueue_buffers();
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}
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@ -266,7 +370,7 @@ static void reset(struct ao *ao)
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*/
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static void audio_pause(struct ao *ao)
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{
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alSourcePausev(ao->channels.num, sources);
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alSourcePause(source);
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}
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/**
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@ -274,14 +378,14 @@ static void audio_pause(struct ao *ao)
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*/
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static void audio_resume(struct ao *ao)
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{
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alSourcePlayv(ao->channels.num, sources);
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alSourcePlay(source);
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}
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static int get_space(struct ao *ao)
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{
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
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queued = NUM_BUF - queued - 3;
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if (queued < 0)
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return 0;
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@ -297,18 +401,15 @@ static int play(struct ao *ao, void **data, int samples, int flags)
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ALint state;
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int num = samples / CHUNK_SAMPLES;
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for (int i = 0; i < num; i++) {
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for (int ch = 0; ch < ao->channels.num; ch++) {
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char *d = data[ch];
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d += i * p->chunk_size;
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alBufferData(buffers[ch][cur_buf[ch]], p->al_format, d,
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p->chunk_size, ao->samplerate);
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alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
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cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
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}
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char *d = data[0];
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d += i * p->chunk_size * ao->channels.num;
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alBufferData(buffers[cur_buf], p->al_format, d, p->chunk_size * ao->channels.num, ao->samplerate);
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alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
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cur_buf = (cur_buf + 1) % NUM_BUF;
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}
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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if (state != AL_PLAYING) // checked here in case of an underrun
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alSourcePlayv(ao->channels.num, sources);
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alSourcePlay(source);
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return num * CHUNK_SAMPLES;
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}
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@ -316,7 +417,7 @@ static double get_delay(struct ao *ao)
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{
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
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return queued * CHUNK_SAMPLES / (double)ao->samplerate;
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}
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