* SAMPR_CAPS_ALL -> SAMPR_CAPS_ALL_48/96/192
* All targets claiming SAMPR_CAPS_ALL now get appropriate subset
* No need to explicitly define HAVE_PLAY_FREQ
* Rates that are a multiple of 44 or 48KHz can be used for playback
Inspired by a patch by Roman Stolyarov, but substantially rewritten by myself.
Change-Id: Iaca7363521b1cb9921e047ba1004d3cbe9c9c23e
Gives us the lowest HW sample rate that's >= 22KHz.
Needed because some targets that don't support 22K support 11K or 8K, so
HW_SAMPR_MIN will give us much lower quality than is acceptable.
Take advantage of this new macro in the SDL, MIDI, and MIKMOD plugins,
and implement a crude "fast enough" test to enable higher sample rates
on more capable targets.
Change-Id: I6ad38026fb3410c62da028e78512e027729bb851
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>
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
