The voice engine can now request more voice data during decoding, it does
not require the entire clip to be available before start of decoding anymore.
Therefore the commit buffer does not need to hold an entire voice clip anymore,
and can be made greatly smaller.
Change-Id: I3eca9026448e725b9b8d0dae1efca0ad185371da
Replaces the NATIVE_FREQUENCY constant with a configurable frequency.
The user may select 48000Hz if the hardware supports it. The default is
still 44100Hz and the minimum is 44100Hz. The setting is located in the
playback settings, under "Frequency".
"Frequency" was duplicated in english.lang for now to avoid having to
fix every .lang file for the moment and throwing everything out of sync
because of the new play_frequency feature in features.txt. The next
cleanup should combine it with the one included for recording and
generalize the ID label.
If the hardware doesn't support 48000Hz, no setting will be available.
On particular hardware where very high rates are practical and desireable,
the upper bound can be extended by patching.
The PCM mixer can be configured to play at the full hardware frequency
range. The DSP core can configure to the hardware minimum up to the
maximum playback setting (some buffers must be reserved according to
the maximum rate).
If only 44100Hz is supported or possible on a given target for playback,
using the DSP and mixer at other samperates is possible if the hardware
offers them.
Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622
Reviewed-on: http://gerrit.rockbox.org/479
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Further decouples voice_thread.c from other playback areas. Also allows
other audio sources, such as FM radio, to be attenuated when voice is
playing by implementing a callback.
Defined as another playback event rather than a new event class:
PLAYBACK_EVENT_VOICE_PLAYING
Change-Id: I2e3e218be6cd6bebbf39e7883a8c0e4ed42b62bb
Because a pointer was uninitialized the move_callback() corrupted memory
by derefencing it and moving stuff to uninitialized value.
This reverts part of 8bbd4d9, where the offending line was removed. I can
only guess it was removed by accident.
Change-Id: I83cee5b396cf3cc99b000e5284fac72fb8ca8db2
...from the compiled code standpoint anyway. frame_out was being
incremented before updating size...sometimes...depending on what
GCC was up to. This seems to help.
Change-Id: Ie4ee3337a2937bd2c26f0a9c4a1a00467954821b
Can't wait for the voice thread to initialize it since it concievably
could be moved before the voice thread actually does so and the move
callback accesses data that must be first set up in the voice thread
function.
Change-Id: Ia0d09539854db85e132e09d26cb129f02f5d93ff
Buffers are not allocated and thread is not created until the first
call where voice is required.
Adds a different callback (sync_callback) to buflib so that other
sorts of synchonization are possible, such as briefly locking-out the
PCM callback for a buffer move. It's sort of a messy addition but it
is needed so voice decoding won't have to be stopped when its buffer
is moved.
Change-Id: I4d4d8c35eed5dd15fb7ee7df9323af3d036e92b3
Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.
Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.
Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.
Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Use generic void * and size_t and make mp3_play_data and its callback
agree on types. Use mp3_play_callback_t instead of prototyping
right in the function call (so it's not so messy to look at). Change
doesn't appear to require plugin API version increment.
Change-Id: Idcab2740ee316a2beb6e0a87b8f4934d9d6b3dd8
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>
This is to a) to cleanup firmware/common and firmware/include a bit, but also b) for Rockbox as an application which should use the host system's c library and headers, separating makes it easy to exclude our files from the build.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25850 a1c6a512-1295-4272-9138-f99709370657
-Allows loading multiple thumbnails back-to-back in the one thumbnail buffer.
-Mutex to prevent race conditions with talk queue indices and
thumbnail buffer state.
-Synchronous shutup.
-Shutup is a noop if no voice is queued.
-mp3_play_stop() does nothing until the audio thread is ready.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18046 a1c6a512-1295-4272-9138-f99709370657
playback is NOT in progress, but that is indeed the case for the only caller
(shutting down message).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15790 a1c6a512-1295-4272-9138-f99709370657
