demux: remove audio parser

The audio parser was needed only by the "old" demuxers, and
demux_rawaudio. All other demuxers output already parsed packets.

demux_rawaudio is usually for raw audio, so using a parser with it
doesn't usually make sense. But you can also force it to read
compressed formats with fixed packet sizes, in which case the parser
would have been used. This use case is probably broken now, but you
will be able to do the same thing with libavformat demuxers.

2 files changed, 5 insertions, 26 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index b5a4ee1ef8..2495012e47 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -344,7 +344,6 @@ static int control(sh_audio_t *sh, int cmd, void *arg)
     switch (cmd) {
     case ADCTRL_RESYNC_STREAM:
         avcodec_flush_buffers(ctx->avctx);
-        ds_clear_parser(sh->ds);
         ctx->previous_data_left = 0;
         ctx->output_left = 0;
         return CONTROL_TRUE;
@@ -384,10 +383,7 @@ static int decode_new_packet(struct sh_audio *sh)
     if (!mpkt) {
         assert(!priv->previous_data_left);
         start = NULL;
-        insize = 0;
-        ds_parse(sh->ds, &start, &insize, pts, 0);
-        if (insize <= 0)
-            return -1;  // error or EOF
+        return -1;  // error or EOF
     } else {
         assert(mpkt->len >= priv->previous_data_left);
         if (!priv->previous_data_left) {
@@ -396,8 +392,7 @@ static int decode_new_packet(struct sh_audio *sh)
         }
         insize = priv->previous_data_left;
         start = mpkt->buffer + mpkt->len - priv->previous_data_left;
-        int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
-        priv->previous_data_left -= consumed;
+        priv->previous_data_left -= insize;
         priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
     }
 
@@ -420,7 +415,7 @@ static int decode_new_packet(struct sh_audio *sh)
         return -1;
     }
     // The "insize >= ret" test is sanity check against decoder overreads
-    if (!sh->parser && insize >= ret)
+    if (insize >= ret)
         priv->previous_data_left = insize - ret;
     if (!got_frame)
         return 0;
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 1314110062..49b7d9a0d8 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -132,20 +132,8 @@ static int init(sh_audio_t *sh, const char *decoder)
         pts = MP_NOPTS_VALUE;
         x = 0;
     }
-    ds_parse(sh->ds, &start, &x, pts, 0);
     srate = 48000;    //fake value
     bps   = 768000/8; //fake value
-    if (x && sh->avctx) { // we have parser and large enough buffer
-        if (sh->avctx->sample_rate < 44100) {
-            mp_msg(MSGT_DECAUDIO,MSGL_INFO,
-                   "This stream sample_rate[%d Hz] may be broken. "
-                   "Force reset 48000Hz.\n",
-                   sh->avctx->sample_rate);
-            srate = 48000; //fake value
-        } else
-            srate = sh->avctx->sample_rate;
-        bps = sh->avctx->bit_rate/8;
-    }
     sh->ds->buffer_pos -= in_size;
 
     int num_channels = 0;
@@ -239,14 +227,10 @@ static int decode_audio(sh_audio_t *sh, unsigned char *buf,
             break;
         x = ds_get_packet_pts(sh->ds, &start, &pts);
         if (x <= 0) {
-            x = 0;
-            ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
-            if (x == 0)
-                continue; // END_NOT_FOUND
-            in_size = x;
+            continue; // END_NOT_FOUND
         } else {
             in_size = x;
-            consumed = ds_parse(sh->ds, &start, &x, pts, 0);
+            consumed = x;
             if (x == 0) {
                 mp_msg(MSGT_DECAUDIO,MSGL_V,
                        "start[%p] in_size[%d] consumed[%d] x[%d].\n",