audio: introduce a new type to hold audio frames

This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).

The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.

Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.

For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.

Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.

1 files changed, 12 insertions, 21 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index d701630fc6..fb429d567b 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -40,7 +40,6 @@
 struct priv {
     AVCodecContext *avctx;
     AVFrame *avframe;
-    struct mp_audio frame;
     bool force_channel_map;
     uint32_t skip_samples, trim_samples;
     bool preroll_done;
@@ -191,7 +190,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
     return true;
 }
 
-static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
+static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
 {
     struct priv *priv = da->priv;
     AVCodecContext *avctx = priv->avctx;
@@ -217,25 +216,18 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
 
     double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
 
-    struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
+    struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
     if (!mpframe)
         return true;
 
-    struct mp_chmap lavc_chmap = mpframe->channels;
-    if (lavc_chmap.num != avctx->channels)
-        mp_chmap_from_channels(&lavc_chmap, avctx->channels);
-    if (priv->force_channel_map) {
-        if (lavc_chmap.num == da->codec->channels.num)
-            lavc_chmap = da->codec->channels;
-    }
-    mp_audio_set_channels(mpframe, &lavc_chmap);
+    if (priv->force_channel_map)
+        mp_aframe_set_chmap(mpframe, &da->codec->channels);
 
-    mpframe->pts = out_pts;
+    if (out_pts == MP_NOPTS_VALUE)
+        out_pts = priv->next_pts;
+    mp_aframe_set_pts(mpframe, out_pts);
 
-    if (mpframe->pts == MP_NOPTS_VALUE)
-        mpframe->pts = priv->next_pts;
-    if (mpframe->pts != MP_NOPTS_VALUE)
-        priv->next_pts = mpframe->pts + mpframe->samples / (double)mpframe->rate;
+    priv->next_pts = mp_aframe_end_pts(mpframe);
 
 #if LIBAVCODEC_VERSION_MICRO >= 100
     AVFrameSideData *sd =
@@ -254,14 +246,14 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
         priv->preroll_done = true;
     }
 
-    uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
+    uint32_t skip = MPMIN(priv->skip_samples, mp_aframe_get_size(mpframe));
     if (skip) {
-        mp_audio_skip_samples(mpframe, skip);
+        mp_aframe_skip_samples(mpframe, skip);
         priv->skip_samples -= skip;
     }
-    uint32_t trim = MPMIN(priv->trim_samples, mpframe->samples);
+    uint32_t trim = MPMIN(priv->trim_samples, mp_aframe_get_size(mpframe));
     if (trim) {
-        mpframe->samples -= trim;
+        mp_aframe_set_size(mpframe, mp_aframe_get_size(mpframe) - trim);
         priv->trim_samples -= trim;
     }
 
@@ -269,7 +261,6 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
 
     av_frame_unref(priv->avframe);
 
-    MP_DBG(da, "Decoded %d samples\n", mpframe->samples);
     return true;
 }