audio: use symbolic constants instead of magic integers

Similar to commit 26468743.

audio/decode/ad_lavc.c

@@ -398,7 +398,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
 
     if (!priv->frame.samples) {
         if (decode_new_packet(da) < 0)
-            return -1;
+            return AD_ERR;
     }
 
     if (!mp_audio_config_equals(buffer, &priv->frame))

audio/decode/ad_mpg123.c

@@ -309,7 +309,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
 
     if (con->need_data) {
         if (feed_new_packet(da) < 0)
-            return -1;
+            return AD_ERR;
     }
 
     if (!mp_audio_config_equals(&da->decoded, buffer))
@@ -338,9 +338,8 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
     return 0;
 
 mpg123_fail:
-    MP_ERR(da, "mpg123 decoding error: %s\n",
-           mpg123_strerror(con->handle));
-    return -1;
+    MP_ERR(da, "mpg123 decoding error: %s\n", mpg123_strerror(con->handle));
+    return AD_ERR;
 }
 
 static int control(struct dec_audio *da, int cmd, void *arg)

audio/decode/ad_spdif.c

@@ -200,7 +200,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
 
     struct demux_packet *mpkt = demux_read_packet(da->header);
     if (!mpkt)
-        return -1;
+        return AD_ERR;
 
     AVPacket pkt;
     mp_set_av_packet(&pkt, mpkt, NULL);
@@ -216,7 +216,7 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
     da->pts_offset += buffer->samples;
     talloc_free(mpkt);
     if (ret < 0)
-        return -1;
+        return AD_ERR;
 
     return 0;
 }

audio/decode/dec_audio.c

@@ -260,7 +260,7 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
             // first, and don't signal a format change to the caller yet.
             if (mp_audio_buffer_samples(da->decode_buffer) > 0)
                 break;
-            error = -2;
+            error = AD_NEW_FMT;
             break;
         }
     }
@@ -274,7 +274,7 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
     bool eof = filter_data.samples == 0 && error < 0;
 
     if (af_filter(da->afilter, &filter_data, eof ? AF_FILTER_FLAG_EOF : 0) < 0)
-        return -1;
+        return AD_ERR;
 
     mp_audio_buffer_append(outbuf, &filter_data);
     if (eof && filter_data.samples > 0)
@@ -285,9 +285,9 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
 
     // Assume the filter chain is drained from old data at this point.
     // (If not, the remaining old data is discarded.)
-    if (error == -2) {
+    if (error == AD_NEW_FMT) {
         if (!reinit_audio_buffer(da))
-            error = -1; // switch to invalid format
+            error = AD_ERR; // switch to invalid format
     }
 
     return error;
@@ -295,7 +295,7 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
 
 /* Try to get at least minsamples decoded+filtered samples in outbuf
  * (total length including possible existing data).
- * Return 0 on success, -1 on error/EOF (not distinguidaed).
+ * Return 0 on success, or negative AD_* error code.
  * In the former case outbuf has at least minsamples buffered on return.
  * In case of EOF/error it might or might not be. */
 int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,

audio/decode/dec_audio.h

@@ -51,6 +51,13 @@ struct dec_audio {
     void *priv;
 };
 
+enum {
+    AD_OK = 0,
+    AD_ERR = -1,
+    AD_NEW_FMT = -2,
+    AD_ASYNC_PLAY_DONE = -3,
+};
+
 struct mp_decoder_list *audio_decoder_list(void);
 int audio_init_best_codec(struct dec_audio *d_audio, char *audio_decoders);
 int audio_decode(struct dec_audio *d_audio, struct mp_audio_buffer *outbuf,

player/audio.c

@@ -297,7 +297,6 @@ static int write_silence_to_ao(struct MPContext *mpctx, int samples, int flags,
     return r;
 }
 
-#define ASYNC_PLAY_DONE -3
 static int audio_start_sync(struct MPContext *mpctx, int playsize)
 {
     struct ao *ao = mpctx->ao;
@@ -374,7 +373,7 @@ static int audio_start_sync(struct MPContext *mpctx, int playsize)
          * in playsize. */
         write_silence_to_ao(mpctx, playsize, 0,
                             written_pts - samples / real_samplerate);
-        return ASYNC_PLAY_DONE;
+        return AD_ASYNC_PLAY_DONE;
     }
     mpctx->syncing_audio = false;
     mp_audio_buffer_prepend_silence(mpctx->ao_buffer, samples);
@@ -418,7 +417,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
         res = audio_decode(d_audio, mpctx->ao_buffer, playsize);
 
     if (res < 0) {  // EOF, error or format change
-        if (res == -2) {
+        if (res == AD_NEW_FMT) {
             /* The format change isn't handled too gracefully. A more precise
              * implementation would require draining buffered old-format audio
              * while displaying video, then doing the output format switch.
@@ -427,7 +426,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
                 uninit_player(mpctx, INITIALIZED_AO);
             reinit_audio_chain(mpctx);
             return -1;
-        } else if (res == ASYNC_PLAY_DONE)
+        } else if (res == AD_ASYNC_PLAY_DONE)
             return 0;
         else if (demux_stream_eof(d_audio->header))
             audio_eof = true;