If a driver can definitely see available devices, it is chosen. Otherwise,
we'll take the first driver that initializes but saw no devices...this might
be because it can't enumerate them, or there really aren't any available.
This prevents the dsp driver from hogging control when there are no /dev/dsp*
nodes (for example, on a Linux box with ALSA and no OSS emulation).
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The API specifies that SDL_OpenAudio() will fill out the 'desired' audio spec
with the correct samples and size set by the driver. This value is important
since it may be used by applications that size audio buffers, etc.
However, we want to allow advanced applications to call SDL_OpenAudioDevice()
which gets passed a const 'desired' parameter, and have the correct data filled
into the 'obtained' parameter, possibly allowing or not allowing format changes.
So... 'obtained' becomes the audio format the user callback is expected to use,
and we add flags to allow the application to specify which format changes are
allowed.
Note: We really need to add a way to query the 'obtained' audio spec.
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Bring SDL to iPhone and iPod Touch
by Holmes Futrell, mentored by Sam Lantinga
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Port SDL 1.3 to the Nintendo DS
by Darren Alton, mentored by Sam Lantinga
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Audio Ideas - Resampling and Pitch Shifting
by Aaron Wishnick, mentored by Ryan C. Gordon
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(This already had a concession for devices opened via the 1.2 entry points,
I've changed it to respect the environment variable and do it for all devices
now.)
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Notable changes:
- Converters between types are autogenerated. Instead of making multiple
passes over the data with seperate filters for endianess, size, signedness,
etc, converting between data types is always one specialized filter. This
simplifies SDL_BuildAudioCVT(), which otherwise had a million edge cases
with the new types, and makes the actually conversions more CPU cache
friendly. Left a stub for adding specific optimized versions of these
routines (SSE/MMX/Altivec, assembler, etc)
- Autogenerated converters are built by SDL/src/audio/sdlgenaudiocvt.pl. This
does not need to be run unless tweaking the code, and thus doesn't need
integration into the build system.
- Went through all the drivers and tried to weed out all the "Uint16"
references that are better specified with the new SDL_AudioFormat typedef.
- Cleaned out a bunch of hardcoded bitwise magic numbers and replaced them
with new SDL_AUDIO_* macros.
- Added initial float32 and int32 support code. Theoretically, existing
drivers will push these through converters to get the data they want to
feed to the hardware.
Still TODO:
- Optimize and debug new converters.
- Update the CoreAudio backend to accept float32 data directly.
- Other backends, too?
- SDL_LoadWAV() needs to be updated to support int32 and float32 .wav files
(both of which exist and can be generated by 'sox' for testing purposes).
- Update the mixer to handle new datatypes.
- Optionally update SDL_sound and SDL_mixer, etc.
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Fixed performance problem with testsprite2 on the D3D driver.
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If SDL_OpenAudio() is passed zero for the desired format
fields, the following environment variables will be used
to fill them in:
SDL_AUDIO_FREQUENCY
SDL_AUDIO_FORMAT
SDL_AUDIO_CHANNELS
SDL_AUDIO_SAMPLES
If an environment variable is not specified, it will be set
to a reasonable default value.
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The current SVN trunk is missing the SDLCALL specifier at numerous locations.
It has to be added for all (possibly user provided) callbacks.
I stumbled over this while creating a makefile for the OpenWatcom compiler for
Win32.
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Make sure every source file includes SDL_config.h, so the proper system
headers are chosen.
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I'm still wrestling with autoheader, but this should work for now...
Fixed lots of build problems with C library support disabled
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FIXME:
Change #include <stdlib.h> to #include "SDL_stdlib.h"
Change #include <string.h> to #include "SDL_string.h"
Make sure nothing else broke because of this...
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I batch edited these files, so please let me know if I've accidentally removed anybody's
credit here.
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From: "alan buckley"
Subject: Patch to fix audio locking on RISC OS
When threads were not disabled on a RISC OS build
the audio mixer mutex was not created.
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This was mostly, if not entirely, written by "Doodle" and "Caetano":
doodle@scenergy.dfmk.hudaniel@caetano.eng.br
--ryan.
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From: "alan buckley"
Subject: Modification for RISC OS version of SDL
Ive attached a zip file with the changes to this email, it contains the
following:
The file sdldiff.txt is the output from cvs diff u. .
The directory thread/riscos contains all the new files to support threading.
Readme.riscos is a new readme file to add.
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surround sound on Linux using the Alsa driver. To use them, naturally
you need a sound card that will do 4 or 6 channels and probably also a
recent version of the Alsa drivers and library. Since the only SDL
output driver that knows about surround sound is the Alsa driver,
you���ll want to choose it, using:
export SDL_AUDIODRIVER=alsa
There are no syntactic changes to the programming API. No new
library calls, no differences in arguments.
There are two semantic changes:
(1) For library calls with number of channels as an argument, formerly
you could use only 1 or 2 for the number of channels. Now you
can also use 4 or 6.
(2) The two "left" and "right" arguments to Mix_SetPanning, for the
case of 4 or 6 channels, no longer simply control the volumes of
the left and right channels. Now the "left" argument is converted
to an angle and Mix_SetPosition is called, and the "right" argu-
ment is ignored.
With two exceptions, so far as I know, the modified SDL12 and
SDL_mixer work the same way as the original versions, when opened for
1 or 2 channel output. The two exceptions are bugs which I fixed.
Well, the first, anyway, is a bug for sure. When rate conversions up
or down by a factor of two are applied (in src/audio/SDL_audiocvt.c),
streams with different numbers of channels (that is, mono and stereo)
are treated the same way: either each sample is copied or every other
sample is omitted. This is ok for mono, but for stereo, it is frames
that should be copied or omitted, where by "frame" I mean a portion of
the stream containing one sample for each channel. (In the SDL source,
confusingly, sometimes frames are called "samples".) So for these
rate conversions, stereo streams have to be treated differently, and
they are, in my modified version.
The other problem that might be characterized as a bug arises
when SDL_mixer is passed a multichannel chunk which does not have an
integral number of frames. Due to the way the effect_position code
loops over frames, when the chunk ends with a partial frame, memory
outside the chunk buffer will be accessed. In the case of stereo,
it���s possible that because malloc may give more memory than requested,
this potential problem never actually causes a segment fault. I don���t
know. For 6 channel chunks, I do know, and it does cause segment
faults.
If SDL_mixer is passed defective chunks and this causes a segment
fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not
it counts as a bug, it���s easy to protect against, so why not? I added
code in mixer.c to discard any partial frame at the end of a chunk.
Then what about when SDL or SDL_mixer is opened for 4 or 6 chan-
nel output? What happens with the parts of the current library
designed for stereo? I don���t know whether I���ve covered all the bases,
but I���ve tried:
(1) For playing 2 channel waves, or other cases where SDL knows it has
to match up a 2 channel source with a 4 or 6 channel output, I���ve
added code in SDL_audiocvt.c to make the necessary conversions.
(2) For playing midis using timidity, I���ve converted timidity to do 4
or 6 channel output, upon request.
(3) For playing mods using mikmod, I put ad hoc code in music.c to
convert the stereo output that mikmod produces to 4 or 6 chan-
nels. Obviously it would be better to change the mikmod code to
mix down into 4 or 6 channels, but I have a hard time following
the code in mikmod, so I didn���t do that.
(4) For playing mp3s, I put ad hoc code in smpeg to copy channels in
the case when 4 or 6 channel output is needed.
(5) There seems to be no problem with .ogg files - stereo .oggs can be
up converted as .wavs are.
(6) The effect_position code in SDL_mixer is now generalized to in-
clude the cases of 4 and 6 channel streams.
I���ve done a very limited amount of compatibility testing for some
of the games using SDL I happen to have. For details, see the file
TESTS.
I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a
couple of 6 channel wave files for testing and a 6 channel ogg file.
If you have the right hardware and version of Alsa, you should be able
to play the wave files with the Alsa utility aplay (and hear all
channels, except maybe lfe, for chan-id.wav, since it���s rather faint).
Don���t expect aplay to give good sound, though. There���s something
wrong with the current version of aplay.
The canyon.ogg file is to test loading of 6 channel oggs. After
patching and compiling, you can play it with playmus. (My version of
ogg123 will not play it, and I had to patch mplayer to get it to play
6 channel oggs.)
Greg Lee <greg@ling.lll.hawaii.edu>
Thus, July 1, 2004
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