1 files changed, 24 insertions, 19 deletions

diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
index 64d957e893..7172908e70 100644
--- a/audio/out/ao_openal.c
+++ b/audio/out/ao_openal.c
@@ -19,8 +19,6 @@
  * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
-#include "config.h"
-
 #include <stdlib.h>
 #include <stdio.h>
 #include <inttypes.h>
@@ -58,7 +56,7 @@ struct priv {
     ALenum al_format;
     int num_buffers;
     int num_samples;
-    int direct_channels;
+    bool direct_channels;
 };
 
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
@@ -67,13 +65,13 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
     case AOCONTROL_GET_VOLUME:
     case AOCONTROL_SET_VOLUME: {
         ALfloat volume;
-        ao_control_vol_t *vol = (ao_control_vol_t *)arg;
+        float *vol = arg;
         if (cmd == AOCONTROL_SET_VOLUME) {
-            volume = (vol->left + vol->right) / 200.0;
+            volume = *vol / 100.0;
             alListenerf(AL_GAIN, volume);
         }
         alGetListenerf(AL_GAIN, &volume);
-        vol->left = vol->right = volume * 100;
+        *vol = volume * 100;
         return CONTROL_TRUE;
     }
     case AOCONTROL_GET_MUTE:
@@ -175,9 +173,6 @@ static void uninit(struct ao *ao)
 
 static int init(struct ao *ao)
 {
-    MP_FATAL(ao, "broken.\n");
-    return -1;
-
     float position[3] = {0, 0, 0};
     float direction[6] = {0, 0, -1, 0, 1, 0};
     ALCdevice *dev = NULL;
@@ -202,8 +197,14 @@ static int init(struct ao *ao)
     alListenerfv(AL_ORIENTATION, direction);
 
     alGenSources(1, &source);
-    if (p->direct_channels && alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT")) {
-        alSourcei(source, alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"), AL_TRUE);
+    if (p->direct_channels) {
+        if (alIsExtensionPresent("AL_SOFT_direct_channels_remix")) {
+            alSourcei(source,
+                alGetEnumValue((ALchar*)"AL_DIRECT_CHANNELS_SOFT"),
+                alcGetEnumValue(dev, "AL_REMIX_UNMATCHED_SOFT"));
+        } else {
+            MP_WARN(ao, "Direct channels aren't supported by this version of OpenAL\n");
+        }
     }
 
     cur_buf = 0;
@@ -267,7 +268,7 @@ static int init(struct ao *ao)
         goto err_out;
     }
 
-    ao->period_size = p->num_samples;
+    ao->device_buffer = p->num_buffers * p->num_samples;
     return 0;
 
 err_out:
@@ -317,7 +318,7 @@ static bool audio_write(struct ao *ao, void **data, int samples)
     for (int i = 0; i < num; i++) {
         char *d = *data;
         buffer_size[cur_buf] =
-            MPMIN(samples - num * p->num_samples, p->num_samples);
+            MPMIN(samples - i * p->num_samples, p->num_samples);
         d += i * buffer_size[cur_buf] * ao->sstride;
         alBufferData(buffers[cur_buf], p->al_format, d,
             buffer_size[cur_buf] * ao->sstride, ao->samplerate);
@@ -341,18 +342,18 @@ static void get_state(struct ao *ao, struct mp_pcm_state *state)
     unqueue_buffers(ao);
     alGetSourcei(source, AL_BUFFERS_QUEUED, &queued);
 
-    double soft_source_latency = 0;
+    double source_offset = 0;
     if(alIsExtensionPresent("AL_SOFT_source_latency")) {
         ALdouble offsets[2];
         LPALGETSOURCEDVSOFT alGetSourcedvSOFT = alGetProcAddress("alGetSourcedvSOFT");
         alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets);
         // Additional latency to the play buffer, the remaining seconds to be
         // played minus the offset (seconds already played)
-        soft_source_latency = offsets[1] - offsets[0];
+        source_offset = offsets[1] - offsets[0];
     } else {
         float offset = 0;
         alGetSourcef(source, AL_SEC_OFFSET, &offset);
-        soft_source_latency = -offset;
+        source_offset = -offset;
     }
 
     int queued_samples = 0;
@@ -361,10 +362,14 @@ static void get_state(struct ao *ao, struct mp_pcm_state *state)
         index = (index + 1) % p->num_buffers;
     }
 
-    state->delay = queued_samples / (double)ao->samplerate + soft_source_latency;
+    state->delay = queued_samples / (double)ao->samplerate + source_offset;
 
     state->queued_samples = queued_samples;
     state->free_samples = MPMAX(p->num_buffers - queued, 0) * p->num_samples;
+
+    ALint source_state = 0;
+    alGetSourcei(source, AL_SOURCE_STATE, &source_state);
+    state->playing = source_state == AL_PLAYING;
 }
 
 #define OPT_BASE_STRUCT struct priv
@@ -384,12 +389,12 @@ const struct ao_driver audio_out_openal = {
     .priv_defaults = &(const struct priv) {
         .num_buffers = 4,
         .num_samples = 8192,
-        .direct_channels = 0,
+        .direct_channels = true,
     },
     .options = (const struct m_option[]) {
         {"num-buffers", OPT_INT(num_buffers), M_RANGE(2, MAX_BUF)},
         {"num-samples", OPT_INT(num_samples), M_RANGE(256, MAX_SAMPLES)},
-        {"direct-channels", OPT_FLAG(direct_channels)},
+        {"direct-channels", OPT_BOOL(direct_channels)},
         {0}
     },
     .options_prefix = "openal",