Additional status callback is added to pcm_play/rec_data instead of
using a special function to set it. Status includes DMA error
reporting to the status callback. Playback and recording callback
become more alike except playback uses "const void **addr" (because
the data should not be altered) and recording uses "void **addr".
"const" is put in place throughout where appropriate.
Most changes are fairly trivial. One that should be checked in
particular because it isn't so much is telechips, if anyone cares to
bother. PP5002 is not so trivial either but that tested as working.
Change-Id: I4928d69b3b3be7fb93e259f81635232df9bd1df2
Reviewed-on: http://gerrit.rockbox.org/166
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
When a global pointer is not declared as constant, gcc will put it in
memory. Getting the address of the string it points to requires loading
the address of the pointer and then loading the pointer. When the pointer
is declared constant, the address of the string is loaded directly.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31345 a1c6a512-1295-4272-9138-f99709370657
Fix problems with volume of recorded material by converting 14-bit samples to
16-bit. Remove duplicate samples from recorded data and support proper
samplerate since ADC runs 1/2 the codec clock. Support monitoring mono on both
output channels by feeding data manually to I2SOUT under the right conditions.
DMA is no longer used for recording since frames must be processed as described
above but it does allow full-duplex audio.
Miscellaneous change includes a proper constant (HW_SAMPR_DEFAULT) to reset the
hardware samplerate when recording is closed. PP5024 and AS3525 have different
default recording rates (22kHz and 44kHz respectively) but both have half-speed
ADC.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31180 a1c6a512-1295-4272-9138-f99709370657
Massive thanks to Michael Chicoine and other testers for finding the early bugs.
This removes all skin memory limitations
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30991 a1c6a512-1295-4272-9138-f99709370657
This is what gimp does when opening such a file.
Tt saves the alpha channel with all-0xff, but other programs might use 0x00.
As a fully transparent image doesn't make sense this should be OK.
Also split the 32bit and 24bit case in the bmp reader, they're sufficiently different.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30968 a1c6a512-1295-4272-9138-f99709370657
For images, rows need to be even (this is not true for anti-aliased font files).
Fix stride and size calculation. This makes images that have odd pixel rows display properly and fixes buffer overflows.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30966 a1c6a512-1295-4272-9138-f99709370657
Now 32bit BMPs with alpha channel can be up- and downscaled without losing
transparency information.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30938 a1c6a512-1295-4272-9138-f99709370657
This uses the alpha blending capabilities introduced with anti-aliased fonts
to draw bitmaps with transparency information. The bmp loader is extended to read
this information (pass FORMAT_TRANSPARENT in format). The alpha information will
be used when drawing the bitmap.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30937 a1c6a512-1295-4272-9138-f99709370657
The buffer_offset paramter of audio_init_recording() is removed as it
was unused in both implementations.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30310 a1c6a512-1295-4272-9138-f99709370657
Namely, introduce buffer_get_buffer() and buffer_release_buffer().
buffer_get_buffer() aquires all available and grabs a lock, attempting to
call buffer_alloc() or buffer_get_buffer() while this lock is locked will cause
a panicf() (doesn't actually happen, but is for debugging purpose).
buffer_release_buffer() unlocks that lock and can additionally increment the
audiobuf buffer to make an allocation. Pass 0 to only unlock if buffer was
used temporarily only.
buffer_available() is a replacement function to query audiobuflen, i.e. what's
left in the buffer.
Buffer init is moved up in the init chain and handles ipodvideo64mb internally.
Further changes happened to mp3data.c and talk.c as to not call the above API
functions, but get the buffer from callers. The caller is the audio system
which has the buffer lock while mp3data.c and talk mess with the buffer.
mpeg.c now implements some buffer related functions of playback.h, especially
audio_get_buffer(), allowing to reduce #ifdef hell a tiny bit.
audiobuf and audiobufend are local to buffer.c now.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30308 a1c6a512-1295-4272-9138-f99709370657
1) fix bug in fmt_gain()
2) take into account steps field of sound_settings_info struct when inc/dec gain
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29593 a1c6a512-1295-4272-9138-f99709370657
* Remove THREAD_ID_CURRENT macro in favor of a thread_self() function, this allows thread functions to be simpler.
* thread_self_entry() shortcut for kernel.c.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29521 a1c6a512-1295-4272-9138-f99709370657
- Support is limited to non-desync jpeg in id3v2 tags. Other formats (hopefully) follow in the future.
- Embedded album art takes precedence over files in album art files.
- No additional buffers are used, the jpeg is read directly from the audio file.
Flyspray: FS#11216
Author: Yoshihisa Uchida and I
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29259 a1c6a512-1295-4272-9138-f99709370657
%pL for the left channel, %pR for the right channel... usable as a value, conditional or bar (exactly the same as %pv/%bl/etc)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29043 a1c6a512-1295-4272-9138-f99709370657