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AS3525v1/v2:

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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
This commit is contained in:
Michael Sevakis 2011-12-08 19:20:00 +00:00
parent 2c7379757c
commit e42a3194de
17 changed files with 272 additions and 212 deletions
Download patch file Download diff file Expand all files Collapse all files

52
firmware/target/arm/as3525/audio-as3525.c
View file

@ -24,16 +24,33 @@
#include "audio.h"
#include "audiohw.h"
#include "sound.h"
#include "general.h"
int audio_channels = 2;
#if CONFIG_CPU == AS3525
int audio_output_source = AUDIO_SRC_PLAYBACK;
#endif
void audio_set_output_source(int source)
{
bitset32(&CGU_PERI, CGU_I2SOUT_APB_CLOCK_ENABLE);
if (source == AUDIO_SRC_PLAYBACK)
I2SOUT_CONTROL &= ~(1<<5);
if ((unsigned)source >= AUDIO_NUM_SOURCES)
source = AUDIO_SRC_PLAYBACK;
bool loopback = source != AUDIO_SRC_PLAYBACK;
#if CONFIG_CPU == AS3525
loopback = loopback && audio_channels > 1;
audio_output_source = source;
#endif
if (loopback)
I2SOUT_CONTROL |= (1<<5); /* loopback from i2sin fifo */
else
I2SOUT_CONTROL |= 1<<5; /* source = loopback from i2sin fifo */
I2SOUT_CONTROL &= ~(1<<5); /* normal i2sout */
}
void audio_input_mux(int source, unsigned flags)
@ -108,4 +125,33 @@ void audio_input_mux(int source, unsigned flags)
}
last_source = source;
#if CONFIG_CPU == AS3525
/* Sync on behalf of change in number of channels */
audio_set_output_source(audio_output_source);
#endif
}
#ifdef CONFIG_SAMPR_TYPES
unsigned int pcm_sampr_to_hw_sampr(unsigned int samplerate,
unsigned int type)
{
#ifdef HAVE_RECORDING
if (samplerate != HW_SAMPR_RESET && type == SAMPR_TYPE_REC)
{
/* Check if the samplerate is in the list of recordable rates.
* Fail to default if not */
int index = round_value_to_list32(samplerate, rec_freq_sampr,
REC_NUM_FREQ, false);
if (samplerate != rec_freq_sampr[index])
samplerate = REC_SAMPR_DEFAULT;
samplerate *= 2; /* Recording rates are 1/2 the codec clock */
}
#endif /* HAVE_RECORDING */
return samplerate;
(void)type;
}
#endif /* CONFIG_SAMPR_TYPES */