replace applicable calls to strlcpy with calls to strmemccpy
which null terminates on truncation
in theory the strmemccpy calls should be slightly faster since they
don't traverse the rest of the source string on truncation
but I seriously doubt there is too much of that going on in the code base
Change-Id: Ia0251514e36a6242bbf3f03c5e0df123aba60ed2
allow buflib_free to check for invalid or already freed handles
within the function -- remove all the invalid handle guards thru core_free
Change-Id: Ibdcbc82760fc93b674c42283fca420d94907df8e
There are various allocations that can't be moved or shrunk.
Provide a global callback struct for this use case instead of
making each caller declare its own dummy struct.
Also fixed ROLO and x1000 installer code which incorrectly
used movable allocations.
Change-Id: I00088396b9826e02e69a4a33477fe1a7816374f1
Rewrite copy_buffer_mono_* functions for correctness.
Bad pointer arithmetic in copy_buffer_mono_l produced
wrong results, or panics on archs which can't handle
the unaligned pointer.
None of the functions handled zero size copies properly
though this probably wasn't an issue in practice.
Change-Id: I81c894e1b8a3440cb409092bec07fe3778a78959
re: coverity
write_write_order: In long(*s++) + *s++,
s is written in *s++ and written in long(*s++)
but the order in which the side effects take place is undefined because
there is no intervening sequence point.
Change-Id: I2911c240f3e85fcfbf77297e8579e02e217c5af5
This fixes the radioart crash that was the result of buffering.c working
on a freed buffer at the same time as buflib (radioart uses buffering.c for the
images). With this change the buffer is owned by buflib exclusively so this
cannot happen.
As a result, audio_get_buffer() doesn't exist anymore. Callers should call
core_alloc_maximum() directly. This buffer needs to be protected as usual
against movement if necessary (previously it was not protected at all which
cased the radioart crash), To get most of it they can adjust the willingness of
the talk engine to give its buffer away (at the expense of disabling voice
interface) with the new talk_buffer_set_policy() function.
Change-Id: I52123012208d04967876a304451d634e2bef3a33
If CPU is not boosted for some reason already, then the stop flush
can take longer than it really ought to.
Change-Id: I0572cc83067749e9945b3eb825f976db21d914f9
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
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>
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
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
* 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
put your station images in .rockbox/fmpresets/<preset name>.bmp or .jpg. Must be in preset mode and the preset name must match the filename
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@26078 a1c6a512-1295-4272-9138-f99709370657
Use smaller recording buffer and watermarks on these models with 2MB of ram
Rearrange watermark calculation expressions so we can use fractional
numbers of seconds but still with integer results
Only enable spinup time adjustement for ATA targets
Flash targets (sansas and ondiofm) should still work fine, but they were
not tested
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@26014 a1c6a512-1295-4272-9138-f99709370657
* Move strncpy() from core to the pluginlib
* Introduce strlcpy() and use that instead in most places (use memcpy in a few) in core and some plugins
* Drop strncpy() from the codec api as no codec used it
* Bump codec and plugin api versions
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21863 a1c6a512-1295-4272-9138-f99709370657
This should be a good first step to allow multi-driver targets, like the Elio (ATA/SD), or the D2 (NAND/SD).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18960 a1c6a512-1295-4272-9138-f99709370657
