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
This commit is contained in:
Andree Buschmann 2011-02-15 20:00:28 +00:00
parent 258626f455
commit 237ca504e1
9 changed files with 36 additions and 36 deletions

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@ -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;

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@ -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* content after this null termination is obsolete and can be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)

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@ -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);

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@ -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;

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@ -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

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@ -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)

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@ -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;
}

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@ -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;

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@ -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 */
};