Those who have keyclick enabled and are really eagar to record upon
boot can get the samplerate messed up because who gets to set the sample
rate last (recording or mixer) is not currently deterministic.
Change-Id: Icc43ed789cf23f928ca49657cb146445b0c558cb
Basically, just give it a good rewrite.
Software codec recording can be implemented in a more straightforward
and simple manner and made more robust through the better codec
control now available.
Encoded audio buffer uses a packed format instead of fixed-size
chunks and uses smaller data headers leading to more efficient usage.
The greatest benefit is with a VBR format like wavpack which needs
to request a maximum size but only actually ends up committing part
of that request.
No guard buffers are used for either PCM or encoded audio. PCM is
read into the codec's provided buffer and mono conversion done at
that time in the core if required. Any highly-specialized sample
conversion is still done within the codec itself, such as 32-bit
(wavpack) or interleaved mono (mp3).
There is no longer a separate filename array. All metadata goes
onto the main encoded audio buffer, eliminating any predermined
file limit on the buffer as well as not wasting the space for
unused path queue slots.
The core and codec interface is less awkward and a bit more sensible.
Some less useful interface features were removed. Threads are kept
on narrow code paths ie. the audio thread never calls encoding
functions and the codec thread never calls file functions as before.
Codecs no longer call file functions directly. Writes are buffered
in the core and data written to storage in larger chunks to speed up
flushing of data. In fact, codecs are no longer aware of the stream
being a file at all and have no access to the fd.
SPDIF frequency detection no longer requires a restart of recording
or plugging the source before entering the screen. It will poll
for changes and update when stopped or prerecording (which does
discard now-invalid prerecorded data).
I've seen to it that writing a proper header on full disk works
when the format makes it reasonably practical to do so. Other cases
may have incorrect data sizes but sample info will be in tact. File
left that way may play anyway.
mp3_enc.codec acquires the ability to write 'Info' headers with LAME
tags to make it gapless (bonus).
Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653
Reviewed-on: http://gerrit.rockbox.org/493
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Playback needs to receive a couple of settings-related messages even
when not playing.
Put the message reply back where it was when loading an encoder for
recording.
Change-Id: I8cc80f46e42a0afd119991d698510e1ebef38ead
Eliminates the pcmrec thread and keeps playback and recording engine
operation mutually-exclusive.
audio_thread.c contains the audio thread which branches to the
correct engine depending upon the request. It also handles the main
audio initialization.
Moves pcm_init into main.c just before dsp_init because I don't want
that one in audio_init in the new file.
(Also makes revision df6e1bc pointless ;)
Change-Id: Ifc1db24404e6d8dd9ac42d9f4dfbc207aa9a26e1
It should not access audio hardware and change settings unless it has
been initialized first and given control of it.
Change-Id: I5004602d7caa604ded751f6838b792d1ff24b3fb
The comment about the format was actually incorrect. The alpha information
is now negated during conversion to native format, according to the
corrected comment.
Change-Id: Ifdb9ffdf9b55e39e64983eec2d9d60339e570bd9
Mixer needn't keep peak data around that will never be used. Just
pass pcm_peaks structure to it instead of allocating for every
channel. Plugin API becomes incompatible.
vu_meter digital mode was still using global peak calculation;
switch it to playback channel like the rest.
Remove some accumulated soil peaks inside pcm.c and make it more
generic.
Change-Id: Ib4d268d80b6a9d09915eea1c91eab483c1a2c009
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
