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Refactor aac decoder as preparation for upcoming m4a changes. The aac decoder does not need to use get_sample_info() to gather frame size or the number of consumed bytes.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29727 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Andree Buschmann 2011-04-16 19:39:01 +00:00
parent a96a72b7b0
commit 5775159462
1 changed files with 6 additions and 33 deletions
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35
apps/codecs/aac.c
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@ -47,8 +47,6 @@ enum codec_status codec_main(void)
stream_t input_stream;
uint32_t sound_samples_done;
uint32_t elapsed_time;
uint32_t sample_duration;
uint32_t sample_byte_size;
int file_offset;
int framelength;
int lead_trim = 0;
@ -207,14 +205,6 @@ next_track:
ci->seek_complete();
}
/* Lookup the length (in samples and bytes) of block i */
if (!get_sample_info(&demux_res, i, &sample_duration,
&sample_byte_size)) {
LOGF("AAC: get_sample_info error\n");
err = CODEC_ERROR;
goto done;
}
/* There can be gaps between chunks, so skip ahead if needed. It
* doesn't seem to happen much, but it probably means that a
* "proper" file can have chunks out of order. Why one would want
@ -235,7 +225,7 @@ next_track:
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n,sample_byte_size);
buffer=ci->request_buffer(&n, demux_res.sample_byte_size[i]);
/* Decode one block - returned samples will be host-endian */
ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
@ -248,34 +238,17 @@ next_track:
}
/* Advance codec buffer (no need to call set_offset because of this) */
ci->advance_buffer(n);
ci->advance_buffer(frame_info.bytesconsumed);
/* Output the audio */
ci->yield();
/* Ensure correct sample_duration is used. For SBR upsampling files
* sample_duration is only half the size of real output frame size. */
sample_duration *= sbr_fac;
/* Gather number of samples for the decoded frame. */
framelength = (frame_info.samples >> 1) - lead_trim;
if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
{
/* Currently limited to at most one frame of tail_trim.
* Seems to be enough.
*/
if (ci->id3->tail_trim == 0
&& sample_duration < (frame_info.samples >> 1))
{
/* Subtract lead_trim just in case we decode a file with
* only one audio frame with actual data.
*/
framelength = sample_duration - lead_trim;
}
else
{
framelength -= ci->id3->tail_trim;
}
framelength -= ci->id3->tail_trim;
}
if (framelength > 0)