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, 29 insertions, 12 deletions

diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 4b3e8149ec..0ca20e5485 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -40,8 +40,9 @@ struct spdifContext {
     uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
     bool             use_dts_hd;
-    struct mp_audio  fmt;
-    struct mp_audio_pool *pool;
+    struct mp_aframe *fmt;
+    int              sstride;
+    struct mp_aframe_pool *pool;
     bool             got_eof;
     struct demux_packet *queued_packet;
 };
@@ -84,7 +85,7 @@ static int init(struct dec_audio *da, const char *decoder)
     da->priv = spdif_ctx;
     spdif_ctx->log = da->log;
     spdif_ctx->use_dts_hd = da->opts->dtshd;
-    spdif_ctx->pool = mp_audio_pool_create(spdif_ctx);
+    spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
 
     if (strcmp(decoder, "spdif_dts_hd") == 0)
         spdif_ctx->use_dts_hd = true;
@@ -198,6 +199,9 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
 
     AVDictionary *format_opts = NULL;
 
+    spdif_ctx->fmt = mp_aframe_create();
+    talloc_steal(spdif_ctx, spdif_ctx->fmt);
+
     int num_channels = 0;
     int sample_format = 0;
     int samplerate = 0;
@@ -246,9 +250,14 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
     default:
         abort();
     }
-    mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
-    mp_audio_set_format(&spdif_ctx->fmt, sample_format);
-    spdif_ctx->fmt.rate = samplerate;
+
+    struct mp_chmap chmap;
+    mp_chmap_from_channels(&chmap, num_channels);
+    mp_aframe_set_chmap(spdif_ctx->fmt, &chmap);
+    mp_aframe_set_format(spdif_ctx->fmt, sample_format);
+    mp_aframe_set_rate(spdif_ctx->fmt, samplerate);
+
+    spdif_ctx->sstride = mp_aframe_get_sstride(spdif_ctx->fmt);
 
     if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
         MP_FATAL(da, "libavformat spdif initialization failed.\n");
@@ -279,7 +288,7 @@ 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 spdifContext *spdif_ctx = da->priv;
 
@@ -308,13 +317,21 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
         goto done;
     }
 
-    int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
-    *out = mp_audio_pool_get(spdif_ctx->pool, &spdif_ctx->fmt, samples);
-    if (!*out)
+    *out = mp_aframe_new_ref(spdif_ctx->fmt);
+    int samples = spdif_ctx->out_buffer_len / spdif_ctx->sstride;
+    if (mp_aframe_pool_allocate(spdif_ctx->pool, *out, samples) < 0) {
+        TA_FREEP(out);
         goto done;
+    }
+
+    uint8_t **data = mp_aframe_get_data_rw(*out);
+    if (!data) {
+        TA_FREEP(out);
+        goto done;
+    }
 
-    memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
-    (*out)->pts = pts;
+    memcpy(data[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
+    mp_aframe_set_pts(*out, pts);
 
 done:
     talloc_free(spdif_ctx->queued_packet);