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High-frequency files increasingly use a block size of over 4608B, which means we need larger buffers to decode them. However, larger buffers no longer fit in IRAM on less-capable devices, hurting performance for "normal" file playback. On our slowest devices (M68K and PP-based devices), this is not worth the tradeoff as they will likely not have enough CPU oomph to decode and downmix these files in realtime. S5L87xx-based devices have the raw performance to do this, so we decided to err on the side of wider file compatibility at the cost of some performance. All other devices are unaffected. Change-Id: I7344cf4c8c6b7b5c14f1ea67381160665d6ece5b
48 lines
1.9 KiB
Text
48 lines
1.9 KiB
Text
Library: libffmpegFLAC
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Imported: 2005-10-26 by Dave Chapman
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This directory contains a FLAC decoder based on the flac.c (and a small
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number of other files) from the ffmpeg project.
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LICENSING INFORMATION
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ffmpeg is licensed under the Lesser GNU General Public License and the
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file flac.c is Copyright 2003 Alex Beregszaszi.
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IMPORT DETAILS
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The decoder is based on ffmpeg-cvs from 26 October 2005.
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The file libavcodec/flac.c was renamed to decoder.c and all ffmpeg
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related code was removed. It was also heavily modified to remove
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debugging information and unused experiemental code.
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It was also changed to provide output as 32-bit integers (with the
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data left-shifted to 28 bit accuracy). The original flac.c provided
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16-bit output.
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In order to minimise memory usage, a small number of hard-coded limits
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are present in decoder.h - mainly limiting the supported blocksize
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(number of samples in a frame) to 4608. This is the default value
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used by the reference FLAC encoder at all standard compression
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settings. However, it is increasingly common for high-resolution/freq
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files to use 8K block sizes, so on devices that can play back such
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files natively and/or have sufficient processor oomph to downmix in
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realtime, we use that larger limit.
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TESTING
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A test program is contained in the file main.c and this should compile
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in any Unix-like environment using the command "make -f Makefile.test".
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Running "test file.flac" will decode the audio data to a WAV file
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called "test.wav" in the current directory. This has been tested with
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16-bit and 24-bit mono and stereo FLAC files.
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This can be used to test the accuracy of the decoder by comparing the
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md5sum of the WAV file produced by this decoder with a WAV file
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produced by the official libFLAC decoder.
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This test program could be extended to perform an internal md5sum
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calculation and comparing that against the md5sum stored in the FLAC
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file's header.
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