audio: move initial decode to generic code

This commit mainly moves the initial decoding of data (done to probe the
audio format) to generic code. This will make it easier to make audio
decoding non-blocking in a later commit.

This commit also changes how decoders return data: instead of having
them write the data into a prepared buffer, they return a reference to
an internal buffer (by setting dec_audio.decoded). This makes it
significantly easier to handle audio format changes, since the decoders
don't really need to care anymore.

1 files changed, 11 insertions, 16 deletions

diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 1042fdeb48..d655f91574 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -38,8 +38,7 @@ struct spdifContext {
     AVFormatContext *lavf_ctx;
     int              iec61937_packet_size;
     int              out_buffer_len;
-    int              out_buffer_size;
-    uint8_t         *out_buffer;
+    uint8_t          out_buffer[OUTBUF_SIZE];
     bool             need_close;
 };
 
@@ -47,7 +46,7 @@ static int write_packet(void *p, uint8_t *buf, int buf_size)
 {
     struct spdifContext *ctx = p;
 
-    int buffer_left = ctx->out_buffer_size - ctx->out_buffer_len;
+    int buffer_left = OUTBUF_SIZE - ctx->out_buffer_len;
     if (buf_size > buffer_left) {
         MP_ERR(ctx, "spdif packet too large.\n");
         buf_size = buffer_left;
@@ -183,24 +182,18 @@ fail:
     return 0;
 }
 
-static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
+static int decode_packet(struct dec_audio *da)
 {
     struct spdifContext *spdif_ctx = da->priv;
     AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
 
-    int sstride = 2 * da->decoded.channels.num;
-    assert(sstride == buffer->sstride);
-
-    if (maxlen * sstride < spdif_ctx->iec61937_packet_size)
-        return 0;
+    mp_audio_set_null_data(&da->decoded);
 
     spdif_ctx->out_buffer_len  = 0;
-    spdif_ctx->out_buffer_size = maxlen * sstride;
-    spdif_ctx->out_buffer      = buffer->planes[0];
 
     struct demux_packet *mpkt = demux_read_packet(da->header);
     if (!mpkt)
-        return AD_ERR;
+        return AD_EOF;
 
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
@@ -211,13 +204,15 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
         da->pts_offset = 0;
     }
     int ret = av_write_frame(lavf_ctx, &pkt);
-    avio_flush(lavf_ctx->pb);
-    buffer->samples = spdif_ctx->out_buffer_len / sstride;
-    da->pts_offset += buffer->samples;
     talloc_free(mpkt);
+    avio_flush(lavf_ctx->pb);
     if (ret < 0)
         return AD_ERR;
 
+    da->decoded.planes[0] = spdif_ctx->out_buffer;
+    da->decoded.samples = spdif_ctx->out_buffer_len / da->decoded.sstride;
+    da->pts_offset += da->decoded.samples;
+
     return 0;
 }
 
@@ -253,5 +248,5 @@ const struct ad_functions ad_spdif = {
     .init = init,
     .uninit = uninit,
     .control = control,
-    .decode_audio = decode_audio,
+    .decode_packet = decode_packet,
 };