[Ffmpeg-cvslog] r8715 - in trunk/libavformat: asf.c utils.c

[benoit]

Author: benoit
Date: Thu Apr 12 10:11:33 2007
New Revision: 8715

Modified:
   trunk/libavformat/asf.c
   trunk/libavformat/utils.c

Log:

DVR-MS probe audio format
patch by John Donaghy \ johnfdonaghy chez gmail dot com /
Original thread:
date: 03/13/2007 04:56 PM
subject: [Ffmpeg-devel] [PATCH] DVR-MS probe audio format - was DVR-MS bug (MP2 decoding)



Modified: trunk/libavformat/asf.c
==============================================================================
--- trunk/libavformat/asf.c	(original)
+++ trunk/libavformat/asf.c	Thu Apr 12 10:11:33 2007
@@ -195,7 +195,7 @@ static int asf_read_header(AVFormatConte
             uint64_t total_size;
             unsigned int tag1;
             int64_t pos1, pos2;
-            int test_for_ext_stream_audio;
+            int test_for_ext_stream_audio, is_dvr_ms_audio=0;
 
             pos1 = url_ftell(pb);
 
@@ -241,6 +241,7 @@ static int asf_read_header(AVFormatConte
                 get_guid(pb, &g);
                 if (!memcmp(&g, &ext_stream_audio_stream, sizeof(GUID))) {
                     type = CODEC_TYPE_AUDIO;
+                    is_dvr_ms_audio=1;
                     get_guid(pb, &g);
                     get_le32(pb);
                     get_le32(pb);
@@ -253,6 +254,12 @@ static int asf_read_header(AVFormatConte
             st->codec->codec_type = type;
             if (type == CODEC_TYPE_AUDIO) {
                 get_wav_header(pb, st->codec, type_specific_size);
+                if (is_dvr_ms_audio) {
+                    // codec_id and codec_tag are unreliable in dvr_ms
+                    // files. Set them later by probing stream.
+                    st->codec->codec_id = CODEC_ID_NONE;
+                    st->codec->codec_tag = 0;
+                }
                 st->need_parsing = 1;
                 /* We have to init the frame size at some point .... */
                 pos2 = url_ftell(pb);

Modified: trunk/libavformat/utils.c
==============================================================================
--- trunk/libavformat/utils.c	(original)
+++ trunk/libavformat/utils.c	Thu Apr 12 10:11:33 2007
@@ -1641,6 +1641,20 @@ static int try_decode_frame(AVStream *st
     return ret;
 }
 
+static int set_codec_from_probe_data(AVStream *st, AVProbeData *pd, int score)
+{
+    AVInputFormat *fmt;
+    fmt = av_probe_input_format2(pd, 1, &score);
+
+    if (fmt) {
+        if (strncmp(fmt->name, "mp3", 3) == 0)
+            st->codec->codec_id = CODEC_ID_MP3;
+        else if (strncmp(fmt->name, "ac3", 3) == 0)
+            st->codec->codec_id = CODEC_ID_AC3;
+    }
+    return fmt;
+}
+
 /* absolute maximum size we read until we abort */
 #define MAX_READ_SIZE        5000000
 
@@ -1662,6 +1676,8 @@ int av_find_stream_info(AVFormatContext 
     offset_t old_offset = url_ftell(&ic->pb);
     int64_t codec_info_duration[MAX_STREAMS]={0};
     int codec_info_nb_frames[MAX_STREAMS]={0};
+    AVProbeData probe_data[MAX_STREAMS];
+    int codec_identified[MAX_STREAMS]={0};
 
     duration_error = av_mallocz(MAX_STREAMS * sizeof(*duration_error));
     if (!duration_error) return AVERROR_NOMEM;
@@ -1687,6 +1703,7 @@ int av_find_stream_info(AVFormatContext 
         last_dts[i]= AV_NOPTS_VALUE;
     }
 
+    memset(probe_data, 0, sizeof(probe_data));
     count = 0;
     read_size = 0;
     ppktl = &ic->packet_buffer;
@@ -1702,6 +1719,9 @@ int av_find_stream_info(AVFormatContext 
                 break;
             if(st->parser && st->parser->parser->split && !st->codec->extradata)
                 break;
+            if (st->codec->codec_type == CODEC_TYPE_AUDIO &&
+                st->codec->codec_id == CODEC_ID_NONE)
+                break;
         }
         if (i == ic->nb_streams) {
             /* NOTE: if the format has no header, then we need to read
@@ -1787,6 +1807,13 @@ int av_find_stream_info(AVFormatContext 
             }
             if(last == AV_NOPTS_VALUE || duration_count[index]<=1)
                 last_dts[pkt->stream_index]= pkt->dts;
+
+            if (st->codec->codec_id == CODEC_ID_NONE) {
+                AVProbeData *pd = &(probe_data[st->index]);
+                pd->buf = av_realloc(pd->buf, pd->buf_size+pkt->size);
+                memcpy(pd->buf+pd->buf_size, pkt->data, pkt->size);
+                pd->buf_size += pkt->size;
+            }
         }
         if(st->parser && st->parser->parser->split && !st->codec->extradata){
             int i= st->parser->parser->split(st->codec, pkt->data, pkt->size);
@@ -1866,12 +1893,28 @@ int av_find_stream_info(AVFormatContext 
                 }
             }
         }else if(st->codec->codec_type == CODEC_TYPE_AUDIO) {
+            if (st->codec->codec_id == CODEC_ID_NONE) {
+                codec_identified[st->index] = set_codec_from_probe_data(st, &(probe_data[st->index]), 0);
+                if (codec_identified[st->index]) {
+                    st->need_parsing = 1;
+                }
+            }
             if(!st->codec->bits_per_sample)
                 st->codec->bits_per_sample= av_get_bits_per_sample(st->codec->codec_id);
         }
     }
 
     av_estimate_timings(ic, old_offset);
+
+    for(i=0;i<ic->nb_streams;i++) {
+        st = ic->streams[i];
+        if (codec_identified[st->index]) {
+            av_read_frame_flush(ic);
+            av_seek_frame(ic, st->index, 0.0, 0);
+            url_fseek(&ic->pb, ic->data_offset, SEEK_SET);
+        }
+    }
+
 #if 0
     /* correct DTS for b frame streams with no timestamps */
     for(i=0;i<ic->nb_streams;i++) {
@@ -1898,6 +1941,9 @@ int av_find_stream_info(AVFormatContext 
 #endif
 
     av_free(duration_error);
+    for(i=0;i<MAX_STREAMS;i++){
+        av_freep(&(probe_data[i].buf));
+    }
 
     return ret;
 }