1 files changed, 33 insertions, 24 deletions

diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index ad3865964c..33e82219b0 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -156,7 +156,8 @@ static int init(struct ao *ao)
 
     ao->untimed = true;
 
-    ao->period_size = ac->aframesize * ac->framecount;
+    ao->device_buffer = ac->aframesize * ac->framecount;
+    ao->period_size = ao->device_buffer;
 
     if (ao->channels.num > AV_NUM_DATA_POINTERS)
         goto fail;
@@ -188,14 +189,6 @@ static void uninit(struct ao *ao)
     }
 }
 
-// return: how many samples can be played without blocking
-static int get_space(struct ao *ao)
-{
-    struct priv *ac = ao->priv;
-
-    return ac->aframesize * ac->framecount;
-}
-
 // must get exactly ac->aframesize amount of data
 static void encode(struct ao *ao, double apts, void **data)
 {
@@ -249,9 +242,9 @@ static void encode(struct ao *ao, double apts, void **data)
     }
 }
 
-// this should round samples down to frame sizes
-// return: number of samples played
-static int play(struct ao *ao, void **data, int samples, int flags)
+// Note: currently relies on samples aligned to period sizes - will not work
+//       in the future.
+static bool audio_write(struct ao *ao, void **data, int samples)
 {
     struct priv *ac = ao->priv;
     struct encoder_context *enc = ac->enc;
@@ -271,7 +264,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     void *tempdata = NULL;
     void *padded[MP_NUM_CHANNELS];
 
-    if ((flags & AOPLAY_FINAL_CHUNK) && (samples % ac->aframesize)) {
+    if (samples % ac->aframesize) {
        tempdata = talloc_new(NULL);
        size_t bytelen = samples * ao->sstride;
        size_t extralen = (ac->aframesize - 1) * ao->sstride;
@@ -282,6 +275,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
        }
        data = padded;
        samples = (bytelen + extralen) / ao->sstride;
+       MP_VERBOSE(ao, "padding final frame with silence\n");
     }
 
     double outpts = pts;
@@ -334,15 +328,28 @@ static int play(struct ao *ao, void **data, int samples, int flags)
 
     pthread_mutex_unlock(&ectx->lock);
 
-    if (flags & AOPLAY_FINAL_CHUNK) {
-        if (bufpos < orig_samples)
-            MP_ERR(ao, "did not write enough data at the end\n");
-    } else {
-        if (bufpos > orig_samples)
-            MP_ERR(ao, "audio buffer overflow (should never happen)\n");
-    }
+    return true;
+}
+
+static void get_state(struct ao *ao, struct mp_pcm_state *state)
+{
+    state->free_samples = ao->device_buffer;
+    state->queued_samples = 0;
+    state->delay = 0;
+}
+
+static bool set_pause(struct ao *ao, bool paused)
+{
+    return true; // signal support so common code doesn't write silence
+}
 
-    return taken;
+static void start(struct ao *ao)
+{
+    // we use data immediately
+}
+
+static void reset(struct ao *ao)
+{
 }
 
 const struct ao_driver audio_out_lavc = {
@@ -350,12 +357,14 @@ const struct ao_driver audio_out_lavc = {
     .description = "audio encoding using libavcodec",
     .name      = "lavc",
     .initially_blocked = true,
-    .reports_underruns = true, // not a thing
     .priv_size = sizeof(struct priv),
     .init      = init,
     .uninit    = uninit,
-    .get_space = get_space,
-    .play      = play,
+    .get_state = get_state,
+    .set_pause = set_pause,
+    .write     = audio_write,
+    .start     = start,
+    .reset     = reset,
 };
 
 // vim: sw=4 ts=4 et tw=80