Dan Everton c22e29f1c3 Fix a warning in Speex codec_main declaration. ... git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12243 a1c6a512-1295-4272-9138-f99709370657		2007-02-09 10:20:27 +00:00
..
lib	FS#6357, patch 1: let iramcopy and bss share the same space in codecs and	2006-11-26 18:31:41 +00:00
liba52	Update several codec Makefiles so that the codec libs build again on Coldfire targets, after the recent move of system-related things to the target tree. (Note to admins: make errors aren't detected on the CVS build page. :))	2006-10-30 18:14:12 +00:00
libalac	Added macros controlling what goes to IRAM on different targets.	2006-11-09 21:59:27 +00:00
libfaad	Next step of Makefile tuning: * Use 'make' internal commands for printing messages. Saves build time especially on cygwin. * SILENT variable used in more places. * Bitmap build system uses one Makefille less.	2006-10-27 21:48:06 +00:00
libffmpegFLAC	fix previous commit and use just .text	2006-12-31 01:04:23 +00:00
libm4a	Fix a couple of MP4 demuxing problems, preventing playback in a few cases. All my test files now play properly.	2007-01-30 21:42:36 +00:00
libmad	Added macros controlling what goes to IRAM on different targets.	2006-11-09 21:59:27 +00:00
libmusepack	Added macros controlling what goes to IRAM on different targets.	2006-11-09 21:59:27 +00:00
libspeex	Add Speex playback support. Patch from FS #5607 thanks to Frederik Vestre.	2007-02-09 10:06:53 +00:00
libwavpack	Update libwavpack with latest changes from the tiny_encoder. This allows	2007-01-08 04:24:32 +00:00
Tremor	FS#6357, patch 3: implemented simple temporary malloc for the Vorbis decoder.	2006-11-26 20:56:26 +00:00
a52.c	Woops. Upon examining the diffs again I find I shouldn't have deleted that one yield() from the a52 codec.	2007-02-07 01:30:05 +00:00
aac.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
adx.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
aiff.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
aiff_enc.c	FS#6096. Recording on PortalPlayer targets (H10, iPod Video, iPod 4g, iPod Color, iPod Nano).	2006-12-18 01:52:21 +00:00
alac.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
codec.h	FS#6357, patch 1: let iramcopy and bss share the same space in codecs and	2006-11-26 18:31:41 +00:00
codec_crt0.c	FS#6357, patch 1: let iramcopy and bss share the same space in codecs and	2006-11-26 18:31:41 +00:00
flac.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
Makefile	Add Speex playback support. Patch from FS #5607 thanks to Frederik Vestre.	2007-02-09 10:06:53 +00:00
mp3_enc.c	Fix last build warning from PP recording changes.	2006-12-18 13:20:27 +00:00
mpa.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
mpc.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
nsf.c	VRC6 speedup/bugfix from Takashi Obara, FS #6635	2007-02-09 08:41:43 +00:00
shorten.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
sid.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
SOURCES	Add Speex playback support. Patch from FS #5607 thanks to Frederik Vestre.	2007-02-09 10:06:53 +00:00
speex.c	Fix a warning in Speex codec_main declaration.	2007-02-09 10:20:27 +00:00
vorbis.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
wav.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
wav_enc.c	Encoders: Mixdown to mono should round towards zero not -infinity.	2006-12-01 00:39:37 +00:00
wavpack.c	Fix resampling clicking as much as possible at the moment. 1) Upsampling clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.	2007-02-07 00:51:50 +00:00
wavpack_enc.c	Encoders: Mixdown to mono should round towards zero not -infinity.	2006-12-01 00:39:37 +00:00