All AAC-HE files will double the frame sample count, not only AAC-HE files with SBR upsampling. This change fixes issues with some m4a files reported in the forums.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29310 a1c6a512-1295-4272-9138-f99709370657

3 files changed, 26 insertions, 22 deletions

diff --git a/apps/metadata/id3tags.c b/apps/metadata/id3tags.c
index 9ee183a..9143f8a 100644
--- a/apps/metadata/id3tags.c
+++ b/apps/metadata/id3tags.c
@@ -358,8 +358,7 @@ static int parseuser( struct mp3entry* entry, char* tag, int bufferpos )
 
     if ((tag - entry->id3v2buf + desc_len + 2) < bufferpos) {
         /* At least part of the value was read, so we can safely try to
-         * parse it
-         */
+         * parse it */
         value = tag + desc_len + 1;
         value_len = bufferpos - (tag - entry->id3v2buf);
         
@@ -368,8 +367,7 @@ static int parseuser( struct mp3entry* entry, char* tag, int bufferpos )
             entry->albumartist = tag;
 #if CONFIG_CODEC == SWCODEC
         } else {
-            value_len = parse_replaygain(tag, value, entry, tag,
-                value_len);
+            value_len = parse_replaygain(tag, value, entry, tag, value_len);
 #endif
         }
     }
@@ -1040,6 +1038,12 @@ void setid3v2title(int fd, struct mp3entry *entry)
 #endif
                 if( tr->ppFunc )
                     bufferpos = tr->ppFunc(entry, tag, bufferpos);
+                    
+                /* Trim. Take into account that multiple string contents will
+                 * only be displayed up to their first null termination. All
+                 * content after this null termination is obsolete and can be
+                 * overwritten. */
+                bufferpos -= (bytesread - strlen(tag));
 
                 /* Seek to the next frame */
                 if(framelen < totframelen)
diff --git a/apps/metadata/metadata_common.c b/apps/metadata/metadata_common.c
index e1ef9a0..4f00177 100644
--- a/apps/metadata/metadata_common.c
+++ b/apps/metadata/metadata_common.c
@@ -337,7 +337,10 @@ long parse_tag(const char* name, char* value, struct mp3entry* id3,
         p = NULL;
     }
     
-    if (p)
+    /* Do not overwrite already available metadata. Especially when reading
+     * tags with e.g. multiple genres / artists. This way only the first 
+     * of multiple entries is used, all following are dropped. */
+    if (p!=NULL && *p==NULL)
     {
         len = strlen(value);
         len = MIN(len, buf_remaining - 1);
diff --git a/apps/metadata/mp4.c b/apps/metadata/mp4.c
index 1ef3701..26ab7cc 100644
--- a/apps/metadata/mp4.c
+++ b/apps/metadata/mp4.c
@@ -73,9 +73,6 @@
 #define MP4_udta FOURCC('u', 'd', 't', 'a')
 #define MP4_extra FOURCC('-', '-', '-', '-')
 
-/* Used to correct id3->samples, if SBR upsampling was detected in esds atom. */
-static bool SBR_upsampling_used = false;
-
 /* Read the tag data from an MP4 file, storing up to buffer_size bytes in
  * buffer.
  */
@@ -120,11 +117,17 @@ static unsigned int read_mp4_tag_string(int fd, int size_left, char** buffer,
 
     if (bytes_read)
     {
-        (*buffer)[bytes_read] = 0;
-        *dest = *buffer;
-        length = strlen(*buffer) + 1;
-        *buffer_left -= length;
-        *buffer += length;
+        /* Do not overwrite already available metadata. Especially when reading
+         * tags with e.g. multiple genres / artists. This way only the first 
+         * of multiple entries is used, all following are dropped. */
+        if (*dest == NULL)
+        {
+            (*buffer)[bytes_read] = 0;
+            *dest = *buffer;
+            length = strlen(*buffer) + 1;
+            *buffer_left -= length;
+            *buffer += length;
+        }
     }
     else
     {
@@ -343,11 +346,6 @@ static bool read_mp4_esds(int fd, struct mp3entry* id3, uint32_t* size)
              * decoding (parts of) the file.
              */
             id3->frequency *= 2;
-            
-            /* Set this to true to be able to calculate the correct runtime 
-             * and bitrate. */
-            SBR_upsampling_used = true;
-
             sbr = true;
         }
     }
@@ -640,7 +638,7 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
                 unsigned int i;
 
                 /* Reset to false. */
-                id3->needs_upsampling_correction = true;
+                id3->needs_upsampling_correction = false;
 
                 lseek(fd, 4, SEEK_CUR);
                 read_uint32be(fd, &entries);
@@ -654,12 +652,12 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
                     read_uint32be(fd, &n);
                     read_uint32be(fd, &l);
 
-                    /* Some SBR files use upsampling. In this case the number 
+                    /* Some AAC file use HE profile. In this case the number 
                      * of output samples is doubled to a maximum of 2048 
                      * samples per frame. This means that files which already 
                      * report a frame size of 2048 in their header will not 
                      * need any further special handling. */
-                    if (SBR_upsampling_used && l<=1024)
+                    if (id3->codectype==AFMT_MP4_AAC_HE && l<=1024)
                     {
                         id3->samples += n * l * 2;
                         id3->needs_upsampling_correction = true;
@@ -774,7 +772,6 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
 
 bool get_mp4_metadata(int fd, struct mp3entry* id3)
 {
-    SBR_upsampling_used = false;
     id3->codectype = AFMT_UNKNOWN;
     id3->filesize = 0;
     errno = 0;