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

9 files changed, 36 insertions, 36 deletions

diff --git a/apps/menus/main_menu.c b/apps/menus/main_menu.c
index 902e14b819..6ee7ba4156 100644
--- a/apps/menus/main_menu.c
+++ b/apps/menus/main_menu.c
@@ -303,7 +303,7 @@ static int info_speak_item(int selected_item, void * data)
             {
                 talk_id(LANG_BATTERY_TIME, false);
                 talk_value(battery_level(), UNIT_PERCENT, true);
-                talk_value(battery_time() *60, UNIT_TIME_EXACT, true);
+                talk_value(battery_time() *60, UNIT_TIME, true);
             }
             else talk_id(VOICE_BLANK, false);
             break;
diff --git a/apps/metadata/id3tags.c b/apps/metadata/id3tags.c
index 9ee183a1b3..9143f8ad25 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+                 * content after this null termination is obsolete and can be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 e1ef9a0d62..4f001775f3 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 1ef3701e30..26ab7cc30a 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;
diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index 7afd97309f..4d7434d771 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -217,7 +217,7 @@ superdom.c
 
 
 
-#ifdef HAVE_TEST_PLUGINS /* enable in advanced build options */
+#if 1//#ifdef HAVE_TEST_PLUGINS /* enable in advanced build options */
 #ifdef HAVE_ADJUSTABLE_CPU_FREQ
 test_boost.c
 #endif
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index 9b00fdbb95..56aedda1c8 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -883,8 +883,7 @@ menu:
                         boost_settings, 2, NULL);
         goto menu;
     }
-    if(boost)
-        rb->cpu_boost(true);
+    rb->cpu_boost(boost ? true: false);
 #endif
 
     if (result == QUIT)
diff --git a/apps/screens.c b/apps/screens.c
index a11c4333ea..81472e6cc7 100644
--- a/apps/screens.c
+++ b/apps/screens.c
@@ -805,7 +805,7 @@ static int runtime_speak_data(int selected_item, void* data)
     talk_ids(false,
              (selected_item < 2) ? LANG_RUNNING_TIME : LANG_TOP_TIME,
              TALK_ID((selected_item < 2) ? global_status.runtime
-                     : global_status.topruntime, UNIT_TIME_EXACT));
+                     : global_status.topruntime, UNIT_TIME));
     return 0;
 }
 
diff --git a/apps/talk.c b/apps/talk.c
index 5fddf85986..8c0f1f3a07 100644
--- a/apps/talk.c
+++ b/apps/talk.c
@@ -1015,7 +1015,7 @@ int talk_number(long n, bool enqueue)
 
 /* Say time duration/interval. Input is time in seconds,
    say hours,minutes,seconds. */
-static int talk_time_unit(long secs, bool exact, bool enqueue)
+static int talk_time_unit(long secs, bool enqueue)
 {
     int hours, mins;
     if (!enqueue)
@@ -1026,11 +1026,9 @@ static int talk_time_unit(long secs, bool exact, bool enqueue)
     }
     if((mins = secs/60)) {
         secs %= 60;
-        if(exact || !hours)
-            talk_value(mins, UNIT_MIN, true);
-        else talk_number(mins, true); /* don't say "minutes" */
+        talk_value(mins, UNIT_MIN, true);
     }
-    if((exact && secs) || (!hours && !mins))
+    if((secs) || (!hours && !mins))
         talk_value(secs, UNIT_SEC, true);
     else if(!hours && secs)
         talk_number(secs, true);
@@ -1110,8 +1108,8 @@ int talk_value_decimal(long n, int unit, int decimals, bool enqueue)
 #endif
 
     /* special case for time duration */
-    if (unit == UNIT_TIME || unit == UNIT_TIME_EXACT)
-        return talk_time_unit(n, unit == UNIT_TIME_EXACT, enqueue);
+    if (unit == UNIT_TIME)
+        return talk_time_unit(n, enqueue);
 
     if (unit < 0 || unit >= UNIT_LAST)
         unit_id = -1;
diff --git a/apps/talk.h b/apps/talk.h
index fc1bfe5a2d..a2a9f44e4e 100644
--- a/apps/talk.h
+++ b/apps/talk.h
@@ -50,8 +50,7 @@ enum {
     UNIT_MB,      /* Megabytes */
     UNIT_KBIT,    /* kilobits per sec */
     UNIT_PM_TICK, /* peak meter units per tick */
-    UNIT_TIME_EXACT,/* time duration/interval in seconds, says hours,mins,secs*/
-    UNIT_TIME,    /* as above but less verbose */
+    UNIT_TIME,    /* time duration/interval in seconds, says hours,mins,secs */
     UNIT_LAST     /* END MARKER */
 };