Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavformat / pmpdec.c

diff --git a/libavformat/pmpdec.c b/libavformat/pmpdec.c
index 9df39be0a19cd911e20ba1fdd3f5670ee2c563f2..4ef8ca34379810668536701e31d6dfb56414cf26 100644 (file)
--- a/libavformat/pmpdec.c
+++ b/libavformat/pmpdec.c
@@ -1,21 +1,21 @@
 /*
- * PMP demuxer
+ * PMP demuxer.
  * Copyright (c) 2011 Reimar Döffinger
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -23,18 +23,18 @@
 #include "avformat.h"
 #include "internal.h"
 
-typedef struct PMPContext {
-    int       cur_stream;
-    int       num_streams;
-    int       audio_packets;
-    int       current_packet;
+typedef struct {
+    int cur_stream;
+    int num_streams;
+    int audio_packets;
+    int current_packet;
     uint32_t *packet_sizes;
-    int       packet_sizes_alloc;
+    int packet_sizes_alloc;
 } PMPContext;
 
-static int pmp_probe(AVProbeData *p)
-{
-    if (!memcmp(p->buf, "pmpm\1\0\0\0", 8))
+static int pmp_probe(AVProbeData *p) {
+    if (AV_RN32(p->buf) == AV_RN32("pmpm") &&
+        AV_RL32(p->buf + 4) == 1)
         return AVPROBE_SCORE_MAX;
     return 0;
 }
@@ -65,7 +65,7 @@ static int pmp_header(AVFormatContext *s)
         av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
         break;
     }
-    index_cnt          = avio_rl32(pb);
+    index_cnt = avio_rl32(pb);
     vst->codec->width  = avio_rl32(pb);
     vst->codec->height = avio_rl32(pb);
 
@@ -73,14 +73,14 @@ static int pmp_header(AVFormatContext *s)
     tb_den = avio_rl32(pb);
     avpriv_set_pts_info(vst, 32, tb_num, tb_den);
     vst->nb_frames = index_cnt;
-    vst->duration  = index_cnt;
+    vst->duration = index_cnt;
 
     switch (avio_rl32(pb)) {
     case 0:
         audio_codec_id = CODEC_ID_MP3;
         break;
     case 1:
-        av_log(s, AV_LOG_WARNING, "AAC is not yet correctly supported\n");
+        av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
         audio_codec_id = CODEC_ID_AAC;
         break;
     default:
@@ -89,21 +89,21 @@ static int pmp_header(AVFormatContext *s)
     }
     pmp->num_streams = avio_rl16(pb) + 1;
     avio_skip(pb, 10);
-    srate    = avio_rl32(pb);
+    srate = avio_rl32(pb);
     channels = avio_rl32(pb) + 1;
     for (i = 1; i < pmp->num_streams; i++) {
         AVStream *ast = avformat_new_stream(s, NULL);
         if (!ast)
             return AVERROR(ENOMEM);
-        ast->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
-        ast->codec->codec_id    = audio_codec_id;
-        ast->codec->channels    = channels;
+        ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+        ast->codec->codec_id = audio_codec_id;
+        ast->codec->channels = channels;
         ast->codec->sample_rate = srate;
         avpriv_set_pts_info(ast, 32, 1, srate);
     }
-    pos = avio_tell(pb) + 4 * index_cnt;
+    pos = avio_tell(pb) + 4*index_cnt;
     for (i = 0; i < index_cnt; i++) {
-        int size  = avio_rl32(pb);
+        int size = avio_rl32(pb);
         int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
         size >>= 1;
         av_add_index_entry(vst, pos, i, size, 0, flags);
@@ -119,7 +119,7 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
     int ret = 0;
     int i;
 
-    if (pb->eof_reached)
+    if (url_feof(pb))
         return AVERROR_EOF;
     if (pmp->cur_stream == 0) {
         int num_packets;
@@ -138,7 +138,7 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
             pmp->packet_sizes[i] = avio_rl32(pb);
     }
     ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
-    if (ret > 0) {
+    if (ret >= 0) {
         ret = 0;
         // FIXME: this is a hack that should be removed once
         // compute_pkt_fields() can handle timestamps properly
@@ -146,14 +146,13 @@ static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
             pkt->dts = s->streams[0]->cur_dts++;
         pkt->stream_index = pmp->cur_stream;
     }
-    pmp->current_packet++;
-    if (pmp->current_packet == 1 || pmp->current_packet > pmp->audio_packets)
+    if (pmp->current_packet % pmp->audio_packets == 0)
         pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
-
+    pmp->current_packet++;
     return ret;
 }
 
-static int pmp_seek(AVFormatContext *s, int stream_idx, int64_t ts, int flags)
+static int pmp_seek(AVFormatContext *s, int stream_index, int64_t ts, int flags)
 {
     PMPContext *pmp = s->priv_data;
     pmp->cur_stream = 0;