summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
authorLAGonauta <lagonauta@gmail.com>2018-03-25 21:39:59 -0300
committerJan Ekström <jeebjp@gmail.com>2018-04-15 00:57:01 +0300
commitc59ebbe399a1e575f2bb8125509b0ad69eafe80a (patch)
tree2a18d7bc606410fa0829f3b8dc695a705f4930ef
parent9efb0278e7cdb851fc6696e94e8f7baa9b95235a (diff)
downloadmpv-c59ebbe399a1e575f2bb8125509b0ad69eafe80a.tar.bz2
mpv-c59ebbe399a1e575f2bb8125509b0ad69eafe80a.tar.xz
ao/openal: Use only one source for audio output
Floating point audio not supported on this commit.
-rw-r--r--audio/out/ao_openal.c205
1 files changed, 153 insertions, 52 deletions
diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
index 6ae91162d3..7ca10edbce 100644
--- a/audio/out/ao_openal.c
+++ b/audio/out/ao_openal.c
@@ -59,11 +59,11 @@
#define MAX_CHANS MP_NUM_CHANNELS
#define NUM_BUF 128
#define CHUNK_SAMPLES 256
-static ALuint buffers[MAX_CHANS][NUM_BUF];
-static ALuint sources[MAX_CHANS];
+static ALuint buffers[NUM_BUF];
+static ALuint source;
-static int cur_buf[MAX_CHANS];
-static int unqueue_buf[MAX_CHANS];
+static int cur_buf;
+static int unqueue_buf;
static struct ao *ao_data;
@@ -113,19 +113,124 @@ static const struct speaker speaker_pos[] = {
{-1},
};
-static ALenum get_al_format(int format)
+static enum af_format get_af_format(int format)
{
switch (format) {
- case AF_FORMAT_U8P: return AL_FORMAT_MONO8;
- case AF_FORMAT_S16P: return AL_FORMAT_MONO16;
- case AF_FORMAT_FLOATP:
+ case AF_FORMAT_U8:
+ if (alGetEnumValue("AL_FORMAT_MONO8"))
+ return AL_TRUE;
+ break;
+
+ case AF_FORMAT_S16:
+ if (alGetEnumValue("AL_FORMAT_MONO16"))
+ return AL_TRUE;
+ break;
+
+ case AF_FORMAT_S32:
+ if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL)
+ return AL_TRUE;
+ break;
+
+ case AF_FORMAT_FLOAT:
if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
- return AL_FORMAT_MONO_FLOAT32;
+ return AL_TRUE;
break;
- case AF_FORMAT_DOUBLEP:
+
+ case AF_FORMAT_DOUBLE:
if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE)
- return AL_FORMAT_MONO_DOUBLE_EXT;
+ return AL_TRUE;
break;
+
+ }
+ return AL_FALSE;
+}
+
+static ALenum get_al_format(struct ao *ao, int format)
+{
+ switch (format) {
+ case AF_FORMAT_U8:
+ switch (ao->channels.num) {
+ case 8:
+ if (alGetEnumValue("AL_FORMAT_71CHN8")) {
+ return alGetEnumValue("AL_FORMAT_71CHN8");
+ }
+ case 7:
+ if (alGetEnumValue("AL_FORMAT_61CHN8")) {
+ return alGetEnumValue("AL_FORMAT_61CHN8");
+ }
+ case 6:
+ if (alGetEnumValue("AL_FORMAT_51CHN8")) {
+ return alGetEnumValue("AL_FORMAT_51CHN8");
+ }
+ case 4:
+ if (alGetEnumValue("AL_FORMAT_QUAD8")) {
+ return alGetEnumValue("AL_FORMAT_QUAD8");
+ }
+ case 2:
+ if (alGetEnumValue("AL_FORMAT_STEREO8")) {
+ return alGetEnumValue("AL_FORMAT_STEREO8");
+ }
+ default:
+ return alGetEnumValue("AL_FORMAT_MONO8");
+ }
+
+ case AF_FORMAT_S16:
+ switch (ao->channels.num) {
+ case 8:
+ if (alGetEnumValue("AL_FORMAT_71CHN16")) {
+ return alGetEnumValue("AL_FORMAT_71CHN16");
+ }
+ case 7:
+ if (alGetEnumValue("AL_FORMAT_61CHN16")) {
+ return alGetEnumValue("AL_FORMAT_61CHN16");
+ }
+ case 6:
+ if (alGetEnumValue("AL_FORMAT_51CHN16")) {
+ return alGetEnumValue("AL_FORMAT_51CHN16");
+ }
+ case 4:
+ if (alGetEnumValue("AL_FORMAT_QUAD16")) {
+ return alGetEnumValue("AL_FORMAT_QUAD16");
+ }
+ case 2:
+ if (alGetEnumValue("AL_FORMAT_STEREO16")) {
+ return alGetEnumValue("AL_FORMAT_STEREO16");
+ }
+ default:
+ return alGetEnumValue("AL_FORMAT_MONO16");
+ }
+
+ case AF_FORMAT_S32:
+ if (strstr(alGetString(AL_RENDERER), "X-Fi") != NULL) {
+ switch (ao->channels.num) {
+ case 8:
+ if (alGetEnumValue("AL_FORMAT_71CHN32")) {
+ return alGetEnumValue("AL_FORMAT_71CHN32");
+ }
+ break;
+ case 7:
+ if (alGetEnumValue("AL_FORMAT_61CHN32")) {
+ return alGetEnumValue("AL_FORMAT_61CHN32");
+ }
+ break;
+ case 6:
+ if (alGetEnumValue("AL_FORMAT_51CHN32")) {
+ return alGetEnumValue("AL_FORMAT_51CHN32");
+ }
+ break;
+ case 4:
+ if (alGetEnumValue("AL_FORMAT_QUAD32")) {
+ return alGetEnumValue("AL_FORMAT_QUAD32");
+ }
+ break;
+ case 2:
+ if (alGetEnumValue("AL_FORMAT_STEREO32")) {
+ return alGetEnumValue("AL_FORMAT_STEREO32");
+ }
+ default:
+ return alGetEnumValue("AL_FORMAT_MONO32");
+ }
+ }
}
return AL_FALSE;
}
@@ -133,9 +238,14 @@ static ALenum get_al_format(int format)
// close audio device
static void uninit(struct ao *ao)
{
+ alSourceStop(source);
+ alSourcei(source, AL_BUFFER, 0);
+
+ alDeleteBuffers(NUM_BUF, buffers);
+ alDeleteSources(1, &source);
+
ALCcontext *ctx = alcGetCurrentContext();
ALCdevice *dev = alcGetContextsDevice(ctx);
- reset(ao);
alcMakeContextCurrent(NULL);
alcDestroyContext(ctx);
alcCloseDevice(dev);
@@ -184,14 +294,12 @@ static int init(struct ao *ao)
alcMakeContextCurrent(ctx);
alListenerfv(AL_POSITION, position);
alListenerfv(AL_ORIENTATION, direction);
- alGenSources(ao->channels.num, sources);
- for (i = 0; i < ao->channels.num; i++) {
- cur_buf[i] = 0;
- unqueue_buf[i] = 0;
- alGenBuffers(NUM_BUF, buffers[i]);
- alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
- alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
- }
+
+ alGenSources(1, &source);
+ cur_buf = 0;
+ unqueue_buf = 0;
+ alGenBuffers(NUM_BUF, buffers);
+
alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
if (alcGetError(dev) == ALC_NO_ERROR && freq)
ao->samplerate = freq;
@@ -200,7 +308,7 @@ static int init(struct ao *ao)
int try_formats[AF_FORMAT_COUNT + 1];
af_get_best_sample_formats(ao->format, try_formats);
for (int n = 0; try_formats[n]; n++) {
- p->al_format = get_al_format(try_formats[n]);
+ p->al_format = get_al_format(ao, try_formats[n]);
if (p->al_format != AL_FALSE) {
ao->format = try_formats[n];
break;
@@ -225,30 +333,26 @@ err_out:
static void drain(struct ao *ao)
{
ALint state;
- alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ alGetSourcei(source, AL_SOURCE_STATE, &state);
while (state == AL_PLAYING) {
mp_sleep_us(10000);
- alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ alGetSourcei(source, AL_SOURCE_STATE, &state);
}
}
static void unqueue_buffers(void)
{
ALint p;
- int s;
- for (s = 0; s < ao_data->channels.num; s++) {
- int till_wrap = NUM_BUF - unqueue_buf[s];
- alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
- if (p >= till_wrap) {
- alSourceUnqueueBuffers(sources[s], till_wrap,
- &buffers[s][unqueue_buf[s]]);
- unqueue_buf[s] = 0;
- p -= till_wrap;
- }
- if (p) {
- alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
- unqueue_buf[s] += p;
- }
+ int till_wrap = NUM_BUF - unqueue_buf;
+ alGetSourcei(source, AL_BUFFERS_PROCESSED, &p);
+ if (p >= till_wrap) {
+ alSourceUnqueueBuffers(source, till_wrap, &buffers[unqueue_buf]);
+ unqueue_buf = 0;
+ p -= till_wrap;
+ }
+ if (p) {
+ alSourceUnqueueBuffers(source, p, &buffers[unqueue_buf]);
+ unqueue_buf += p;
}
}
@@ -257,7 +361,7 @@ static void unqueue_buffers(void)
*/
static void reset(struct ao *ao)
{
- alSourceStopv(ao->channels.num, sources);
+ alSourceStop(source);
unqueue_buffers();
}
@@ -266,7 +370,7 @@ static void reset(struct ao *ao)
*/
static void audio_pause(struct ao *ao)
{
- alSourcePausev(ao->channels.num, sources);
+ alSourcePause(source);
}
/**
@@ -274,14 +378,14 @@ static void audio_pause(struct ao *ao)
*/
static void audio_resume(struct ao *ao)
{
- alSourcePlayv(ao->channels.num, sources);
+ alSourcePlay(source);
}
static int get_space(struct ao *ao)
{
ALint queued;
unqueue_buffers();
- alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
+ alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
queued = NUM_BUF - queued - 3;
if (queued < 0)
return 0;
@@ -297,18 +401,15 @@ static int play(struct ao *ao, void **data, int samples, int flags)
ALint state;
int num = samples / CHUNK_SAMPLES;
for (int i = 0; i < num; i++) {
- for (int ch = 0; ch < ao->channels.num; ch++) {
- char *d = data[ch];
- d += i * p->chunk_size;
- alBufferData(buffers[ch][cur_buf[ch]], p->al_format, d,
- p->chunk_size, ao->samplerate);
- alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
- cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
- }
+ char *d = data[0];
+ d += i * p->chunk_size * ao->channels.num;
+ alBufferData(buffers[cur_buf], p->al_format, d, p->chunk_size * ao->channels.num, ao->samplerate);
+ alSourceQueueBuffers(source, 1, &buffers[cur_buf]);
+ cur_buf = (cur_buf + 1) % NUM_BUF;
}
- alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ alGetSourcei(source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING) // checked here in case of an underrun
- alSourcePlayv(ao->channels.num, sources);
+ alSourcePlay(source);
return num * CHUNK_SAMPLES;
}
@@ -316,7 +417,7 @@ static double get_delay(struct ao *ao)
{
ALint queued;
unqueue_buffers();
- alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
+ alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
return queued * CHUNK_SAMPLES / (double)ao->samplerate;
}