audio/out: prepare for non-interleaved audio

This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.

1 files changed, 12 insertions, 11 deletions

diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index b849f9b2b4..9be3a7b6ae 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -296,8 +296,6 @@ static void fill_with_padding(void *buf, int cnt, int sz, const void *padding)
 }
 
 // close audio device
-static int encode(struct ao *ao, double apts, void *data);
-static int play(struct ao *ao, void *data, int len, int flags);
 static void uninit(struct ao *ao, bool cut_audio)
 {
     struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
@@ -315,7 +313,7 @@ static int get_space(struct ao *ao)
 {
     struct priv *ac = ao->priv;
 
-    return ac->aframesize * ac->sample_size * ao->channels.num * ac->framecount;
+    return ac->aframesize * ac->framecount;
 }
 
 // must get exactly ac->aframesize amount of data
@@ -444,10 +442,10 @@ static int encode(struct ao *ao, double apts, void *data)
     return packet.size;
 }
 
-// plays 'len' bytes of 'data'
+// plays 'samples' samples of 'ni_data[0]'
 // it should round it down to frame sizes
-// return: number of bytes played
-static int play(struct ao *ao, void *data, int len, int flags)
+// return: number of samples played
+static int play(struct ao *ao, void **ni_data, int samples, int flags)
 {
     struct priv *ac = ao->priv;
     struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
@@ -457,6 +455,8 @@ static int play(struct ao *ao, void *data, int len, int flags)
     double nextpts;
     double pts = ao->pts;
     double outpts;
+    void *data = ni_data[0];
+    int len = samples * ao->sstride;
     int bytelen = len;
 
     len /= ac->sample_size * ao->channels.num;
@@ -477,8 +477,9 @@ static int play(struct ao *ao, void *data, int len, int flags)
                               extralen / ac->sample_size,
                               ac->sample_size, ac->sample_padding);
             // No danger of recursion, because AOPLAY_FINAL_CHUNK not set
-            written = play(ao, paddingbuf, bytelen + extralen, 0);
-            if (written < bytelen) {
+            written =
+                play(ao, &paddingbuf, (bytelen + extralen) / ao->sstride, 0);
+            if (written * ao->sstride < bytelen) {
                 MP_ERR(ao, "did not write enough data at the end\n");
             }
             talloc_free(paddingbuf);
@@ -492,7 +493,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
 
         while (encode(ao, outpts, NULL) > 0) ;
 
-        return FFMIN(written, bytelen);
+        return (FFMIN(written, bytelen)) / ao->sstride;
     }
 
     if (pts == MP_NOPTS_VALUE) {
@@ -559,7 +560,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
             if (ac->offset_left <= -len) {
                 // skip whole frame
                 ac->offset_left += len;
-                return len * ac->sample_size * ao->channels.num;
+                return len;
             } else {
                 // skip part of this frame, buffer/encode the rest
                 bufpos -= ac->offset_left;
@@ -632,7 +633,7 @@ static int play(struct ao *ao, void *data, int len, int flags)
             ectx->next_in_pts = nextpts;
     }
 
-    return bufpos * ac->sample_size * ao->channels.num;
+    return bufpos;
 }
 
 const struct ao_driver audio_out_lavc = {