From 5fd8a1e04c725329435e3bead5f11ee3ffb9f1c1 Mon Sep 17 00:00:00 2001
From: wm4 <wm4@nowhere>
Date: Mon, 10 Nov 2014 22:01:23 +0100
Subject: audio: make decoders output refcounted frames

This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".

Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.

For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
---
 audio/decode/ad_mpg123.c | 19 ++++++++++---------
 1 file changed, 10 insertions(+), 9 deletions(-)

(limited to 'audio/decode/ad_mpg123.c')

diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 30a4790746..2fbbec6032 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -43,6 +43,7 @@ struct ad_mpg123_context {
     short delay;
     /* If the stream is actually VBR. */
     char vbr;
+    struct mp_audio frame;
 };
 
 static void uninit(struct dec_audio *da)
@@ -197,27 +198,25 @@ static int set_format(struct dec_audio *da)
     int encoding;
     ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
     if (ret == MPG123_OK) {
-        mp_audio_set_num_channels(&da->decoded, channels);
-        da->decoded.rate = rate;
+        mp_audio_set_num_channels(&con->frame, channels);
+        con->frame.rate = rate;
         int af = mpg123_format_to_af(encoding);
         if (!af) {
             /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
             MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
             return MPG123_ERR;
         }
-        mp_audio_set_format(&da->decoded, af);
+        mp_audio_set_format(&con->frame, af);
         con->sample_size = channels * af_fmt2bps(af);
     }
     return ret;
 }
 
-static int decode_packet(struct dec_audio *da)
+static int decode_packet(struct dec_audio *da, struct mp_audio **out)
 {
     struct ad_mpg123_context *con = da->priv;
     int ret;
 
-    mp_audio_set_null_data(&da->decoded);
-
     struct demux_packet *pkt;
     if (demux_read_packet_async(da->header, &pkt) == 0)
         return AD_WAIT;
@@ -257,8 +256,11 @@ static int decode_packet(struct dec_audio *da)
     }
 
     int got_samples = bytes / con->sample_size;
-    da->decoded.planes[0] = audio;
-    da->decoded.samples = got_samples;
+    *out = mp_audio_pool_get(da->pool, &con->frame, got_samples);
+    if (!*out)
+        return AD_ERR;
+
+    memcpy((*out)->planes[0], audio, bytes);
 
     update_info(da);
     return 0;
@@ -274,7 +276,6 @@ static int control(struct dec_audio *da, int cmd, void *arg)
 
     switch (cmd) {
     case ADCTRL_RESET:
-        mp_audio_set_null_data(&da->decoded);
         mpg123_close(con->handle);
 
         if (mpg123_open_feed(con->handle) != MPG123_OK) {
-- 
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