; Copyright 2010 Castle Technology Ltd
;
; Licensed under the Apache License, Version 2.0 (the "License");
; you may not use this file except in compliance with the License.
; You may obtain a copy of the License at
;
; http://www.apache.org/licenses/LICENSE-2.0
;
; Unless required by applicable law or agreed to in writing, software
; distributed under the License is distributed on an "AS IS" BASIS,
; WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
; See the License for the specific language governing permissions and
; limitations under the License.
;
GET Hdr:ListOpts
GET Hdr:Macros
GET Hdr:System
GET Hdr:Machine.<Machine>
GET Hdr:ImageSize.<ImageSize>
$GetIO
GET Hdr:OSEntries
GET Hdr:HALEntries
GET Hdr:HALDevice
GET Hdr:AudioDevice
GET Hdr:MixerDevice
GET Hdr:Proc
GET hdr.omap3530
GET hdr.StaticWS
GET hdr.PRCM
AREA |Asm$$Code|, CODE, READONLY, PIC
EXPORT Audio_Init
IMPORT TPSRead
IMPORT TPSWrite
IMPORT memcpy
IMPORT DebugHALPrint
IMPORT DebugHALPrintReg
IMPORT HAL_IRQClear
IMPORT HAL_CounterDelay
; A brief rundown of OMAP HAL audio support:
; Audio in/out is typically provided via the TWL/TPS companion chip, using "port 2" of the TWL/TPS audio subsystem. "port 2" provides a I2S/TDM-compatible interface, capable of operating at a variety of sample rates.
; Two interfaces are used to link the audio subsystem to the OMAP: I2C is used to program the audio subsystem, and I2S is used to transmit & receive data (using the McBSP module on the OMAP side).
; Both the TWL/TPS and the OMAP are capable of acting as either the master or slave across the I2S link, so to keep things simple this implementation has the TWL/TPS acting as master and the OMAP acting as slave.
; Finally, to keep the McBSP buffer full, the SDMA module of the OMAP is used
; This means that, on the software side, audio is provided by the following network of components:
;
; HAL I2C driver
; |
; RISC OS I2C driver
; | |
; HAL mixer device <-> HAL audio device
; | |
; SoundControl module <-> SoundDMA module
; |
; DMAManager module
; |
; HAL DMA device
;
; (Note - there's also a small amount of communication between the HAL DMA device and the HAL audio device, to allow for the correct setting of the FIFO threshold/DMA packet size)
; With regards to the actual sound I/O features the TPS makes available:
; "Digital" data comes from ICs, "Analog" data are pins that may/may not be connected to speakers/microphones/etc. The "RX" path sends data out through the DACs, and the "TX" path accepts data from the ADCs.
;
; INPUTS - DIGITAL OUTPUTS - DIGITAL
;Stereo I2S data from CPU (tdm/i2s.din) Stereo I2S data to CPU (tdm/i2s.dout)
;PCM voice data from modem (pcm.vdr) PCM voice data to modem (pcm.vdx)
;PCM voice data from SUBMIC (pcm.vdr)
;PCM voice data from BT (bt.pcm.din) PCM voice data to BT (bt.pcm.dout)
;
; INPUTS - ANALOGUE OUTPUTS - ANALOGUE
;Mono headset microphone (hsmic) Stereo headset (hsor/hsol)
; Stereo hands-free speaker (ihf_l/r_p/m)
;Mono handset main microphone (mic_main) Mono handset earphone (earp/earm)
;Mono handset submicrophone (mic_sub)
;Digital microphone 0
;Digital microphone 1
;Stereo FM radio/auxiliary input (fml/fmr, auxl/auxr)
; Stereo predriver output (predriv_l/r)
;Mono carkit microphone (usb_din) Stereo carkit output (usb_d+/-)
;
; (The above ignores the 4-channel TDM capability, since we're using I2S)
; We'll be operating the mixer in the "option 1" setting, which gives "four simultaneous multimedia analog channels (four multimedia DACs and no modem voice path) and two simultaneous ADCs or four channels coming from digital microphones
; This basically means that the mixer looks like the satisfyingly confusing figure 14-6 in the TPS manual (swcu050d.pdf). In summary....
; Output: Mode: PGA Amplifier: VDL_GAIN ARXL1 ARXR1 ARXL2 ARXR2
; Headset left MIX * * *
; Headset right MIX * * *
; Predriver left MIX * * * * (1,2)
; Predriver right MIX * * * * (2)
; Hands-free left MUX * * * * (2,3)
; Hands-free right MUX * * * * (2,3)
; Carkit left MIX * * *
; Carkit right MIX * * *
; Earphone MIX * * * *
;
; (1) Also allows "Vmid" to be used as a mix source, whatever that is.
; (2) Stereo crossover is supported on ARXL2/ARXR2
; (3) Lacks an output amplifier. All other outputs have output amplifiers (albeit ones with only a handful of gain settings).
; The "analog mixing" allows each PGA amplifier to have two inputs:
; PGA Amplifier: Inputs:
; VDL_GAIN Ampli_L, ????? (seriously - the diagram has two lines that don't go anywhere!)
; ARXL1 DACL1, Ampli_L
; ARXR1 DACR1, Ampli_R
; ARXL2 DACL2, Ampli_L
; ARXR2 DACR2, Ampli_R
; Digital muxes allow the DACs to have multiple sources.
; These muxes are controlled by the RX_PATH_SEL register.
; DAC: Inputs: SDRL1 SDRR1 SDRM1 SDRL2 SDRR2 SDRM2
; DACL1 * * * *
; DACR1 * * * *
; DACL2 * *
; DACR2 * *
; SDRL1/SDRR1 is data received over the I2S link
; SDRL2/SDRR2 is data received over the I2S link, added to the output of the ATXARX(L/R)_PGA amplifiers (which source their data from the audio TX L1/R1 channels)
; The RX & TX I2S channels are enabled/disabled via the OPTION register
; And, speaking of the "TX" path, there are two input amplifiers (Ampli_L, Ampli_R), which feed two ADCs (ADCL, ARDC) (as well as potentially feeding back to the output PGA amplifiers). The Ampli_L & Ampli_R amplifiers take ther input from the following:
; Ampli_L Carkit mic, headset mic, handset main mic, FM radio/aux left
; Ampli_R Handset sub mic, FM radio/aux right
; The ADCL & ADCR output is combined with another source of data - the digital microphones - to produce four outputs: Audio TX L1, R1, L2, R2. These signals are fed into corresponding amplifiers (ATXL1PGA, ATXR1PGA, ATXL2PGA, ATXR2PGA) & filters, before being fed into the four mic channels of the I2S/TDM interface (and, in the case of the R1/L1 channels, into the loopback PGAs).
; The channels are mapped to the I2S/TDM mic channels as follows:
; Audio TX channel TDM mic channel
; L1 1
; R1 3
; L2 2
; R2 4
; Out of all the amplifiers, only the following have reasonably flexible gain controls: (ignoring BT, DTMF)
; ATXL1PGA, ATXR1PGA
; AVTXL2PGA, AVTXR2PGA
; ARXL1PGA, ARXR1PGA
; ARXL2PGA, ARXR2PGA
; VRX(L?)PGA (voice downlink)
; VSTPGA (voice sidetone)
; VRX2ARXPGA (voice RX to audio RX)
; ARX2VTXPGA (audio RX to voice TX)
; ARXL1_APGA, ARXR1_APGA
; ARXL2_APGA, ARXR2_APGA
; ATX2ARXL, ATX2ARXR (audio TX to audio RX)
; VDL_APGA (voice downlink again?)
; MICAMPL/R (stereo microphone amp - but which mic?)
; The following have rather restrictive gain settings:
; EAR (earpiece)
; HSR (headset left/right)
; L/R Predrivers
; Carkit L/R
; So, in summary.....
; * Since RISC OS only produces one stream of stereo data, we can assign one DAC/amplifier pair (DACL2/DACR2, to keep the voice path free for the future?) to playing the data generated by the OS
; * VDL_GAIN, DACL1, DACR1 amplifiers can be ignored (although VDL_GAIN would be useful for mono data received by Ampli_L?)
; * There are only two ADCs, so you can only really sample from one stereo source at a time (or two mono)
; * The plethora of outputs can select their mix from RISC OS sound data & the ADC data at will
; * In any case, RISC OS doesn't really have an API for sound recording (or setting the recording source), so we could just ignore all the inputs entirely until that gets rectified (although for most OMAP devices which RISC OS is targeting, it's likely that the stereo "auxiliary" input will be the one providing all the input data, so an API to select the recording source isn't entirely necessary)
; As for mapping inputs & outputs to those currently supported by the HAL mixer API:
; TPS HAL mixer category
; Headset left/right HEADPHONES
; Predriver left/right SPEAKER
; Hands-free left/right LINE_OUT
; Carkit left/right AUX_OUT (carkit is more likely to be useful than earphone, if a carkit-compatible USB driver is written)
; Earphone none?
; Headset mic MIC
; Handset main mic none?
; Handset sub mic none?
; Digital mic 0 none?
; Digital mic 1 none?
; Stereo FM/aux input LINE_IN
; Mono carkit mic AUX_IN (to match AUX_OUT)
; 'SYSTEM' should obviously control the volume of the sound RO generates - i.e. by using the ARXL2PGA/ARXR2PGA amplifiers
; For the beagleboard, the TWL/TPS is connected to the OMAP via I2C1 & McBSP2
; The audio out header is connected to the headset L/R channels on the TPS
; The audio in header is connected to the aux L/R channels on the TPS
; Flag to enable gobs of debug output
GBLL AudioDebug
AudioDebug SETL {FALSE}
Audio_Init
Push "v1-v4,lr"
[ AudioDebug
DebugTX "Audio_Init"
]
ADRL v1, AudioWS
MOV a1, v1
ADR a2, AudioTemplate
MOV a3, #Audio_DeviceSize
BL memcpy
STR sb, [v1, #:INDEX:AudioWorkspace]
; Get McBSP2 logical address
LDR v2, L4_Per_Log
ADD v2, v2, #L4_McBSP2-L4_Per
STR v2, [v1, #:INDEX:AudioRegs]
[ AudioDebug
DebugReg v2, "McBSP2 @ "
]
; Before we go any further, turn on the power to McBSP2 & make sure it's in a reset state
; TODO - This should probably go in AudioActivate!
LDR a1, L4_ClockMan_Log
ADD a1, a1, #CM_FCLKEN_PER
LDR a2, [a1]
ORR a2, a2, #1 ; McBSP2 clock enable
STR a2, [a1]
LDR a2, [a1, #CM_ICLKEN_PER-CM_FCLKEN_PER]!
ORR a2, a2, #1
STR a2, [a1]
; Clear CONTROL_DEVCONF.MCBSP2_CLKS?
LDR a1, L4_Core_Log
ADD a1, a1, #&2200
LDR a2, [a1, #&74]
BIC a2, a2, #1<<6
STR a2, [a1, #&74]
; Set McBSP2 to reset state
MOV a1, #0
STR a1, [v2, #MCBSPLP_SPCR1]
MOV a1, #&200
STR a1, [v2, #MCBSPLP_SPCR2]
; Perform full reset, in fact
MOV a1, #2
STR a1, [v2, #MCBSPLP_SYSCONFIG]
10
LDR a1, [v2, #MCBSPLP_SYSCONFIG]
TST a1, #2
BNE %BT10
; Set smart idle mode. No auto-idle available! :(
MOV a1, #2<<3
STR a1, [v2, #MCBSPLP_SYSCONFIG]
; One-time setup of transmitter registers
MOV a1, #&F ; Active-low RX/TX frame sync, RX on rising, TX on falling
STR a1, [v2, #MCBSPLP_PCR]
MOV a1, #&40 ; 16bit words, 1 word per frame
STR a1, [v2, #MCBSPLP_XCR1]
LDR a1, =&8041 ; dual-phase frame, (16 bit, 1 word), MSB first, 1-bit delay
STR a1, [v2, #MCBSPLP_XCR2]
; Program sample rate generator registers - not sure if this is required, since we run in slave mode
LDR a1, =&f01 ; 16 bit frame width-1, CLKGDV=1
STR a1, [v2, #MCBSPLP_SRGR1]
MOV a1, #&1f ; FPER=2*16 bit-1
STR a1, [v2, #MCBSPLP_SRGR2]
; Now resume HAL device setup...
ADD v2, v1, #Audio_DeviceSize
MOV a1, v2
ADR a2, MixerTemplate
MOV a3, #Mixer_DeviceSize
BL memcpy
; Fill in pointers to each other
STR v1, [v2, #HALDevice_MixerCtrlr]
STR v2, [v1, #HALDevice_AudioMixer]
LDRB a1, [sb, #BoardConfig_MixerChans]
STRB a1, [v2, #:INDEX:MixerDisableFlags]
; Unavailable channels, and input channels, start muted
ORR a1, a1, #(1<<MixerChannel_HeadsetMic)+(1<<MixerChannel_AuxInput)+(1<<MixerChannel_CarkitMic)
ADD a2, v2, #:INDEX:MixerSettings
10 AND a3, a1, #1
MOVS a1, a1, LSR #1
STR a3, [a2], #8
BNE %BT10
; Register devices
MOV a2, v1
MOV a1, #0
CallOS OS_AddDevice
MOV a2, v2
MOV a1, #0
CallOS OS_AddDevice
; Set TPS audio registers to default values
MOV a1, #TPSAUDIO_IIC*2
ADR a2, TPSDefaults
MOV a3, #1 ; Program CODEC_MODE first to make sure the power is off, then program everything else in one go
MOV a4, #CODEC_MODE
LDR v1, OSentries+4*OS_IICOpV
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT10
DebugTX "Audio_Init: TPS CODEC_MODE reset failed!"
10
]
MOV a1, #TPSAUDIO_IIC*2
ADD a2, a2, #1
MOV a3, #(MISC_SET_2+1)-OPTION
ADD a4, a4, #1
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT10
DebugTX "Audio_Init: TPS audio reset failed!"
10
]
; Write APLL_CTL with the right value
MOV a1, #TPSAUDIO_IIC*2
ADD a2, sb, #BoardConfig_APLL_CTL
MOV a3, #1
MOV a4, #APLL_CTL
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT10
DebugTX "Audio_Init: APLL_CTL write failed!"
10
]
[ AudioDebug
DebugTX "Audio_Init done"
]
Pull "v1-v4,pc"
TPSDefaults ; List of default settings for TPS registers
; These are the settings taken from TRM (swcu050d.pdf), and then overlaid with the default settings that are required for sound output (i.e. steps 3-17, 19-27, 31 of the example given in section 14.5.3), and then tidied up a bit further
DCB &00 ; CODEC_MODE
DCB &C0 ; OPTION - RX path 2 enabled
DCB &00 ; (unused)
DCB &00 ; MCBIAS_CTL
DCB &20 ; ANAMICL
DCB &00 ; ANAMICR
DCB &00 ; AVADC_CTL
DCB &00 ; ADCMICSEL
DCB &00 ; DIGMIXING
DCB &00 ; ATXL1PGA
DCB &00 ; ATXR1PGA
DCB &00 ; ATXL2PGA
DCB &00 ; AVTXR2PGA
DCB &01 ; AUDIO_IF
DCB &00 ; VOICE_IF
DCB &00 ; ARXR1PGA mute
DCB &00 ; ARXL1PGA mute
DCB &3F ; ARXR2PGA 0dB
DCB &3F ; ARXL2PGA 0dB
DCB &00 ; VRXPGA
DCB &00 ; VSTPGA
DCB &00 ; VRX2ARXPGA
DCB &0C ; AVDAC_CTL
DCB &00 ; ARX2VTXPGA
DCB &30 ; ARXL1_APGA_CTL mute
DCB &30 ; ARXR1_APGA_CTL mute
DCB &33 ; ARXL2_APGA_CTL 0dB
DCB &33 ; ARXR2_APGA_CTL 0dB
DCB &00 ; ATX2ARXPGA
DCB &00 ; BT_IF
DCB &00 ; BTPGA
DCB &00 ; BTSTPGA
DCB &00 ; EAR_CTL
DCB &24 ; HS_SEL - TODO - Also HS_OUTLOW_EN?
DCB &00 ; HS_GAIN_SET
DCB &34 ; HS_POPN_SET
DCB &00 ; PREDL_CTL - TODO - Also PREDL_OUTLOW_EN?
DCB &00 ; PREDR_CTL
DCB &00 ; PRECKL_CTL
DCB &00 ; PRECKR_CTL
DCB &00 ; HFL_CTL
DCB &00 ; HFR_CTL
DCB &05 ; ALC_CTL
DCB &00 ; ALC_SET1
DCB &00 ; ALC_SET2
DCB &00 ; BOOST_CTL
DCB &01 ; SOFTVOL_CTL - TODO - tweak delay setting
DCB &13 ; DTMF_FREQSEL
DCB &00 ; DTMF_TONEXT1H
DCB &00 ; DTMF_TONEXT1L
DCB &00 ; DTMF_TONEXT2H
DCB &00 ; DTMF_TONEXT2L
DCB &79 ; DTMF_TONOFF
DCB &11 ; DTMF_WANONOFF
DCB &00 ; CODEC_RX_SCRAMBLE_H
DCB &00 ; CODEC_RX_SCRAMBLE_M
DCB &00 ; CODEC_RX_SCRAMBLE_L
DCB &00 ; APLL_CTL - gets filled in properly afterwards
DCB &00 ; DTMF_CTL
DCB &00 ; DTMF_PGA_CTL2
DCB &00 ; DTMF_PGA_CTL1
DCB &02 ; MISC_SET_1
DCB &00 ; PCMBTMUX
DCB &00 ; (unused)
DCB &00 ; (unused)
DCB &00 ; (unused)
DCB &0A ; RX_PATH_SEL - Map I2S RX path 2 to digital paths 1 and 2 (somewhat unecessary to map to both since we only need one)
DCB &00 ; VDL_APGA_CTL
DCB &00 ; VIBRA_CTL
DCB &00 ; VIBRA_SET
DCB &00 ; (unused)
DCB &00 ; ANAMIC_GAIN
DCB &00 ; MISC_SET_2
ASSERT (. - TPSDefaults) = (MISC_SET_2+1)-CODEC_MODE
ALIGN
; Sample rate table
; The first 'reserved' byte is used to store the value that needs programming into the TPS CODEC_MODE register
GBLA numrate
numrate SETA 0
MACRO
$lab cdf $freq, $per, $mode ; CD-derived rate ($freq in Hz, since all integral)
$lab DCD $freq*1024 ; frequency value as reported by Sound_SampleRate
DCB $per ; period as reported via Sound_Configure
DCB $mode ; CODEC_MODE setting
DCW 0 ; padding to 8 bytes
numrate SETA numrate+1
MEND
ASSERT HALDevice_AudioRateTableSize = 8
ratetab cdf 8000, 125, &01 ; 8kHz (125usec) AC97/6
cdf 11025, 91, &11 ; 11.025kHz (~91 usec) CD/4
cdf 12000, 83, &21 ; 12kHz (~83 usec) AC97/4
cdf 16000, 63, &41 ; 16kHz (~63 usec) AC97/3
cdf 22050, 45, &51 ; 22.05kHz (~45 usec) CD/2
cdf 24000, 42, &61 ; 24kHz (~42 usec) AC97/2
cdf 32000, 31, &81 ; 32kHz (~31 usec) AC97*2/3
cdf 44100, 23, &91 ; 44.1kHz (~23 usec) CD/1
cdf 48000, 21, &a1 ; 48kHz (~21 usec) AC97/1
cdf 96000, 10, &e1 ; 96kHz (~10 usec) AC97*2
; Audio controller HAL device
AudioTemplate
DCW HALDeviceType_Audio + HALDeviceAudio_AudC
DCW HALDeviceID_AudC_TPS65950
DCD HALDeviceBus_Ser + HALDeviceSerBus_IIC
DCD 1:SHL:16 ; API version
DCD AudioDesc
DCD 0 ; Address - N/A
% 12 ; Reserved
DCD AudioActivate
DCD AudioDeactivate
DCD AudioReset
DCD AudioSleep
DCD McBSP2_IRQ ; Device
DCD 0 ; TestIRQ cannot be called
% 8
DCD 0 ; Filled in during init
DCD 1 ; Output channels (supported so far)
DCD 0 ; Input channels (supported so far)
ASSERT (.-AudioTemplate) = HALDevice_Audio_Size
; DMA channel parameters
DCD 0 ; flags
DCD McBSP2_DMA_TX + 1 ; logical channel
DCD 0 ; 'cycle speed'
DCD 2 ; transfer unit size
DCD L4_McBSP2+MCBSPLP_DXR ; *physical* address to send data to
; Enable/disable/IRQ routines
DCD PreEnable
DCD PostEnable
DCD PreDisable
DCD PostDisable
DCD IRQHandle
DCD numrate ; Number of sample rates
DCD ratetab ; Sample rate table
DCD AudioSetRate ; SetRate function
ASSERT (.-AudioTemplate) = HALDevice_Audio_Size_1
DCD 0 ; Filled in during init
DCD 0 ; Filled in during init
DCD 0
ALIGN
ASSERT (.-AudioTemplate) = Audio_DeviceSize
; Mixer HAL device
MixerTemplate
DCW HALDeviceType_Audio + HALDeviceAudio_Mixer
DCW HALDeviceID_Mixer_TPS65950
DCD HALDeviceBus_Ser + HALDeviceSerBus_IIC
DCD 1 ; API version
DCD MixerDesc
DCD 0 ; Address - N/A
% 12 ; Reserved
DCD MixerActivate
DCD MixerDeactivate
DCD MixerReset
DCD MixerSleep
DCD -1 ; Device
DCD 0 ; TestIRQ cannot be called
% 8
DCD 0 ; Filled in during init
DCD MixerChannels
DCD MixerGetFeatures
DCD MixerSetMix
DCD MixerGetMix
DCD MixerGetMixLimits
ASSERT (.-MixerTemplate) = HALDevice_Mixer_Size + 4
; Default settings will be filled in during init
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCD 0, 0
DCW 0 ; MixerHeadsetGain
DCW 0 ; MixerPredriverGain
DCW 0 ; MixerHandsFreeGain
DCW 0 ; MixerCarkitOutGain
DCW &3F3F ; MixerSystemGain
DCW &3333 ; MixerSystemGain2
% 4
ASSERT (.-MixerTemplate) = Mixer_DeviceSize
AudioDesc
= "TPS65950-compatible audio controller", 0
MixerDesc
= "TPS65950-compatible audio mixer", 0
ALIGN
AudioActivate
; TODO - Fix this!
; SoundDMA is higher up the module chain than SoundControl
; Which means we need the audio controller to do the mixer init, otherwise stuff might not work properly
Entry "v1,sb", 4
; Turn off codec, program mix params
LDR v1, [a1, #HALDevice_AudioMixer]
ADD v1, v1, #(:INDEX:MixerSettings)+8*MixerChannels
MOV a2, #MixerChannels-1
LDR sb, AudioWorkspace
[ AudioDebug
DebugTX "AudioActivate: Setting defaults"
]
ADR a3, ratetab
LDRB a3, [a3, #5] ; Just use first rate for now
STR a3, AudioMode
BL SetCodecMode
10 LDMDB v1!,{a3-a4}
BL ReallySetMix
SUBS a2, a2, #1
BGE %BT10
; Finished!
MOV a1, #1
EXIT
AudioDeactivate
MOV pc, lr
AudioReset
MOV pc, lr
AudioSleep
MOV a1, #0
MOV pc, lr
SetCodecMode
; Reprogram CODEC_MODE from softcopy
; Assumes a1=audio device ptr, sb=HAL workspace
Entry "a1-a4,v1"
ADR a2, AudioMode
MOV a1, #TPSAUDIO_IIC*2
MOV a3, #1
MOV a4, #CODEC_MODE
LDR v1, OSentries+4*OS_IICOpV
BL TPSWrite
[ AudioDebug
CMP a1, #0
EXIT EQ
DebugTX "SetCodecMode: TPS write failed!"
]
EXIT
PreEnable
; a2 = DMA buffer length
; Use the buffer length to calculate a DMA packet size that fits in the FIFO.
LDR a3, AudioWorkspace
MOV a4, #1
10
MOV a2, a2, LSR #1 ; Number of DMA elements/FIFO entries
CMP a2, #&200 ; FIFO size is &500. To try and avoid underflow, try to make sure there's no more than &200 free entries.
STRLE a2, [a3, #:INDEX:DMAPktSz_Audio]
MOVLE pc, lr
; Divide size by two until we reach our goal
TST a2, #1
MOVEQ a4, a4, LSL #1
BEQ %BT10
; If we're here, it means we can't divide any more, and we're still stuck with a large transfer size.
; For RISC OS, the largest non-fittable size is 4K-4, which will result in a2=&3FF and a4=2
; Since &3FF is smaller than the FIFO size we'll make an exception to our above rule and allow it to be used
; But to future-proof ourselves against buffers larger than 4K we'll fall back to using a4 if needed. Transferring only two elements at a time will hurt the memory bus a bit, but at least it'll work!
CMP a2, #&400
STRLE a2, [a3, #:INDEX:DMAPktSz_Audio]
STRGT a4, [a3, #:INDEX:DMAPktSz_Audio]
MOV pc, lr
PostEnable
; a2 = DMA buffer length
; This function is a mix of two things - the TPS programming steps from section 14.5.3 of the manual (swcu050d.pdf, page 715/716), and the McBSP programming steps
; The steps for both have been stripped down to the bare minimum that are required for (re)initialisation.
; The McBSP instance is initialised first, to allow the DMA to prefill the TX FIFO. The TWL/TPS is then initialised afterwards.
Entry "v1-v3,sb"
LDR sb, AudioWorkspace
[ AudioDebug
DebugTX "PostEnable"
]
Push "a1"
LDR v1, AudioRegs
; Set McBSP instance to reset state (although it should be reset already)
MOV a1, #0
STR a1, [v1, #MCBSPLP_SPCR1]
MOV a1, #&200
STR a1, [v1, #MCBSPLP_SPCR2]
; Reset IRQ state, and enable underflow/overflow IRQ
MVN a1, #0
STR a1, [v1, #MCBSPLP_IRQSTATUS]
MOV a1, #&1800
STR a1, [v1, #MCBSPLP_IRQENABLE]
; Program FIFO threshold value, using our precomputed packet size
LDR a2, DMAPktSz_Audio
[ AudioDebug
DebugReg a2, "Packet size="
]
SUB a2, a2, #1
STR a2, [v1, #MCBSPLP_THRSH2] ; Threshold = number of 16bit samples that SDMA sends, minus one. Any other value is wrong and will cause audio corruption.
; Wait 2 SRG clock cycles
; For now, just wait for a msec or so
MOV a1, #1024
BL HAL_CounterDelay
; Bring sample rate generator out of reset (required even with external clocks)
MOV v2, #&240
STR v2, [v1, #MCBSPLP_SPCR2]
; Bring transmitter out of reset
ORR v2, v2, #1
STR v2, [v1, #MCBSPLP_SPCR2]
; Wait some more, according to linux
MOV a1, #1024
BL HAL_CounterDelay
Pull "a1"
; Now start frame sync
ORR v2, v2, #&80
STR v2, [v1, #MCBSPLP_SPCR2]
; Wait for FIFO to fill by DMA
LDR v3, DMAPktSz_Audio
; Use the 32K timer to provide a clock reference for a timeout (in
; case DMA isn't running for some reason, e.g. DMAManager being
; naughty and sneakily resetting it during Service_PreReset)
LDR v2, L4_32KTIMER_Log
LDR a3, [v2, #REG_32KSYNCNT_CR]
10
LDR a2, [v1, #MCBSPLP_XBUFFSTAT]
CMP a2, v3
BLE %FT20
LDR ip, [v2, #REG_32KSYNCNT_CR]
SUB ip, ip, a3
CMP ip, #16384 ; Half a second should be more than enough
BLO %BT10
[ AudioDebug
DebugTX "FIFO fill timeout!"
]
; Since timeouts shouldn't happen, I don't think anyone will mind if
; we just bug out and don't start the audio codec
EXIT
20
[ AudioDebug
DebugReg a2, "Post-prime XBUFFSTAT="
]
; TPS Step 18 - turn codec power on
LDR v1, AudioMode
ORR v1, v1, #2
STR v1, AudioMode
BL SetCodecMode
; TPS Steps 28-30 - Set and wait for antipop
BL SetAndWaitForAntiPop
; TPS Step 32 - Disable codec, for some unmentioned reason
BIC v1, v1, #2
STR v1, AudioMode
BL SetCodecMode
; TPS Step 33 - Enable codec again
ORR v1, v1, #2
STR v1, AudioMode
BL SetCodecMode
; TPS Steps 34-37 - Enable headset (if needed!)
LDR a1, [a1, #HALDevice_AudioMixer]
LDR a2, MixerSettings+MixerChannel_HeadsetOut*8
TST a2, #1
BLEQ UpdateHeadset
EXIT
PreDisable
; Disable the TWL/TPS, and then McBSP
; This should avoid unwanted underflow IRQs
Entry "v1,sb"
LDR sb, AudioWorkspace
[ AudioDebug
DebugTX "PreDisable"
]
; Disable IRQs
LDR a2, AudioRegs
MOV a3, #0
STR a3, [a2, #MCBSPLP_IRQENABLE]
; Disable headset output if required
LDR v1, [a1, #HALDevice_AudioMixer]
LDR a2, [v1, #:INDEX:MixerSettings+MixerChannel_HeadsetOut*8]
ANDS a2, a2, #1
BNE %FT30
STRH a2, [v1, #:INDEX:MixerHeadsetGain]
Push "a1"
ADRL a2, HeadsetDisableSequence
MOV a3, #1
LDR v1, OSentries+4*OS_IICOpV
10
LDRB a4, [a2],#1
CMP a4, #255
LDREQB a4, [a2],#1
BLEQ DoHeadsetRampDelay
MOV a1, #TPSAUDIO_IIC*2
CMP a4, #0
BEQ %FT20
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT15
DebugReg a4,"PreDisable: Write failed, reg="
15
]
ADD a2, a2, #1
B %BT10
20
Pull "a1"
30
; Turn codec power off
LDR a2, AudioMode
BIC a2, a2, #2
STR a2, AudioMode
BL SetCodecMode
; Disable McBSP
LDR v1, AudioRegs
MOV a1, #0
STR a1, [v1, #MCBSPLP_SPCR1]
MOV a1, #&200
STR a1, [v1, #MCBSPLP_SPCR2]
EXIT
PostDisable
MOV pc, lr
IRQHandle
[ AudioDebug
Push "sb,lr"
LDR sb, AudioWorkspace
LDR a2, AudioRegs
LDR a2, [a2, #MCBSPLP_IRQSTATUS]
DebugReg a2,"IRQHandle: IRQSTATUS="
Pull "sb,lr"
]
; Just clear the IRQ and ask for an audio reset
LDR a3, AudioRegs
MOV a2, #0
STR a2, [a3, #MCBSPLP_IRQENABLE] ; Make sure we don't get bothered about this again - we can't guarantee when the OS will get round to resetting the audio, and could potentially get stuck in a loop if we allow the IRQ to keep firing
MVN a2, #0
STR a2, [a3, #MCBSPLP_IRQSTATUS]
MOV a1, #1
MOV pc, lr
AudioSetRate
; a2 = sample rate index (0-based)
Entry "sb"
LDR sb, AudioWorkspace
[ AudioDebug
DebugReg a2, "AudioSetRate: "
]
; Reload CODEC_MODE with the required value
; The manual states APLL_RATE can only be changed if the codec is off!
ADRL a3, ratetab
ASSERT HALDevice_AudioRateTableSize = 8
ADD a3, a3, a2, LSL #3
LDRB a3, [a3, #5]
LDR a2, AudioMode
CMP a2, a3
EXIT EQ ; Redundant update, ignore
TST a2, #2
[ AudioDebug
BEQ %FT10
DebugTX "AudioSetRate: Failed, codec is active!"
EXIT
10
|
EXIT NE
]
STR a3, AudioMode
BL SetCodecMode
EXIT
MixerActivate
MOV a1, #1
MixerDeactivate
MOV pc, lr
MixerReset
Entry "v1,sb", 4
; Turn off codec, program default mix params
ADR v1, MixerSettings+8*MixerChannels
LDR a1, [a1, #HALDevice_MixerCtrlr]
MOV a2, #MixerChannels-1
LDR sb, AudioWorkspace
[ AudioDebug
DebugTX "MixerReset: Setting defaults"
]
LDR a3, AudioMode
BIC a3, a3, #2
STR a3, AudioMode
BL SetCodecMode
10 LDMDB v1!,{a3-a4}
BL ReallySetMix
SUBS a2, a2, #1
BGE %BT10
EXIT
MixerSleep
; TODO?
MOV a1, #0
MOV pc, lr
MixerGetFeatures
; Check if this channel is disabled and needs the 'fixed' flag
MOV a4, #1
LDRB a3, MixerDisableFlags
TST a3, a4, LSL a2
ADR a1, MixerFeaturesTab
LDR a1, [a1, a2, LSL #2]
ORRNE a1, a1, #MixerFeaturesFlag_Fixed
MOV pc, lr
MixerFeaturesTab
; Stereo headset output
DCW 0
DCW MixerCategory_Headphones
; Stereo speaker predriver output
DCW MixerFeaturesFlag_DefaultMute
DCW MixerCategory_Speaker
; Stereo hands-free output
DCW MixerFeaturesFlag_DefaultMute
DCW MixerCategory_LineOut
; Stereo carkit output
DCW MixerFeaturesFlag_DefaultMute
DCW MixerCategory_AuxOut
; System audio
DCW 0
DCW MixerCategory_System
; Mono headset mic
DCW MixerFeaturesFlag_Mono :OR: MixerFeaturesFlag_DefaultMute
DCW MixerCategory_Mic
; Stereo FM/aux input
DCW MixerFeaturesFlag_DefaultMute
DCW MixerCategory_LineIn
; Mono carkit mic
DCW MixerFeaturesFlag_Mono :OR: MixerFeaturesFlag_DefaultMute
DCW MixerCategory_AuxIn
MixerGetMixLimits
ADR a1, MixerLimitsTab
ADD a1, a1, a2, LSL #3
ADD a1, a1, a2, LSL #2
LDMIA a1, {a1-a2,a4}
STMIA a3, {a1-a2,a4}
MOV pc, lr
MixerLimitsTab
; Stereo headset output
DCD -6*16
DCD 6*16
DCD 6*16
; Stereo speaker predriver output
DCD -6*16
DCD 6*16
DCD 6*16
; Stereo hands-free output
DCD 0
DCD 0
DCD 16
; Stereo carkit output
DCD -6*16
DCD 6*16
DCD 6*16
; System audio
DCD -62*16 + -18*16
DCD 12*16 + 12*16
DCD 16
; Mono headset mic
DCD 0
DCD 30*16
DCD 16
; Stereo FM/aux input. May be wrong? (relies on ALC, which fig 14-6 says isn't available)
DCD 0
DCD 30*16
DCD 16
; Mono carkit mic
DCD 0
DCD 30*16
DCD 16
MixerSetMix
; a1 = mixer device
; a2 = channel
; a3 = mute flag
; a4 = gain, in dB*16
Entry "v1"
; Double-check that we aren't being asked to program a disabled channel
; Although ReallySetMix will do another check, we must check here to make sure we don't update the stored settings (particularly the headset settings, which UpdateHeadset treats as gospel)
MOV ip, #1
LDRB v1, MixerDisableFlags
TST v1, ip, LSL a2
EXIT NE
ADR v1, MixerSettings
ADD v1, v1, a2, LSL #3
LDMIA v1, {ip, lr}
TEQ a3, ip
TEQEQ a4, lr
STMNEIA v1, {a3, a4}
LDRNE a1, [a1, #HALDevice_MixerCtrlr]
BLNE ReallySetMix
EXIT
ReallySetMix
; a1 = *audio* device
; a2-a4 as above
; Preserves all regs
; Channel gain & mute control can't easily be abstracted, need to do it on a case-by-case basis :(
; Common code to get ready for IIC transfers
Entry "a1-a4,v1-v5,sb,v7"
LDR sb, AudioWorkspace
[ AudioDebug
DebugTX "ReallySetMix:"
DebugReg a2,"Channel="
DebugReg a3,"Mute="
DebugReg a4,"Gain="
]
; Triple-check that we aren't being asked to program a disabled channel
; (ReallySetMix gets called unconditionally in a few places; easier to just check for invalid channels here rather than in every place it gets called from)
LDRB v5, [sb, #BoardConfig_MixerChans]
MOV v2, #1
TST v5, v2, LSL a2
EXIT NE
LDR v5, [a1, #HALDevice_AudioMixer]
MOV v2, a3
LDR v4, AudioMode
ADD v7, v5, #:INDEX: MixerSettings+4
LDR v1, OSentries+4*OS_IICOpV
MOV a1, #TPSAUDIO_IIC*2
ADD v7, v7, a2, LSL #3
ADR a3, SetMixTab
MOV v3, a4
LDR pc, [a3, a2, LSL #2]
; Following code entered with:
; a1, v1 ready for TPSRead/TPSWrite
; v2 = mute flag
; v3 = gain
; v4 = current CODEC_MODE setting
; v5 = mixer device
; sb = HAL WS
; v7 = gain writeback ptr
SetMixTab
DCD SetMixHeadsetOut
DCD SetMixPredriver
DCD SetMixHandsFree
DCD SetMixCarkitOut
DCD SetMixSystem
DCD SetMixHeadsetMic
DCD SetMixAuxInput
DCD SetMixCarkitMic
SetMixHeadsetOut ; HS_* regs
; If the codec is on, go through to UpdateHeadset
; Else do nothing
TST v4, #2
MOVNE a1, v5
BLNE UpdateHeadset
EXIT
SetMixPredriver ; PRED* regs
MOV v4, #&24 ; 0dB gain, AL2
MOV lr, #0
CMP v3, #6*16
MOVGE v4, #&14 ; 6dB gain, AL2
MOVGE lr, #6*16
CMP v3, #-6*16
MOVLE v4, #&34 ; -6dB gain, AL2
MOVLE lr, #-6*16
TST v2, #1
ORREQ v4, v4, v4, LSL #8 ; R channel settings
MOVNE v4, #0 ; Else disable everything
ADD a2, v5, #:INDEX:MixerPredriverGain
LDRH ip, [a2]
CMP ip, v4
EXIT EQ
STRH v4, [a2]
MOV a3, #2
STR lr, [v7]
MOV a4, #PREDL_CTL
BL TPSWrite
[ AudioDebug
CMP a1, #0
EXIT EQ
DebugTX "SetMixPredriver: TPS write failed!"
]
EXIT
SetMixHandsFree ; HF* regs
TST v2, #1
MOVEQ v4, #&3E ; AL2, enabled
ORREQ v4, v4, #&3E00 ; AR2, enabled
MOVNE v4, #0
ADD a2, v5, #:INDEX:MixerHandsFreeGain
LDRH ip, [a2]
CMP ip, v4
EXIT EQ
MOV lr, #0
STRH v4, [a2]
MOV a3, #2
STR lr, [v7]
MOV a4, #HFL_CTL
BL TPSWrite
[ AudioDebug
CMP a1, #0
EXIT EQ
DebugTX "SetMixHandsFree: TPS write failed!"
]
EXIT
SetMixCarkitOut ; PRECK* regs
MOV v4, #&64 ; 0dB gain, AL2
MOV lr, #0
CMP v3, #6*16
MOVGE v4, #&54 ; 6dB gain, AL2
MOVGE lr, #6*16
CMP v3, #-6*16
MOVLE v4, #&74 ; -6dB gain, AL2
MOVLE lr, #-6*16
TST v2, #1
ORREQ v4, v4, v4, LSL #8 ; R channel settings
MOVNE v4, #0 ; Else disable everything
ADD a2, v5, #:INDEX:MixerCarkitOutGain
LDRH ip, [a2]
CMP ip, v4
EXIT EQ
STRH v4, [a2]
MOV a3, #2
STR lr, [v7]
MOV a4, #PRECKL_CTL
BL TPSWrite
[ AudioDebug
CMP a1, #0
EXIT EQ
DebugTX "SetMixCarkitOut: TPS write failed!"
]
EXIT
SetMixSystem ; ARX(L/R)2PGA (digital) and ARX(L/R)2_APGA_CTL (analogue)
; The ARX(L/R)2PGA regs have both a coarse grain stage (0dB, 6dB, 12dB) and fine-grain stage (-62dB -> 0dB)
; ARX(L/R)2_APGA_CTL are (relatively) fine-grain, from -18dB to 12dB
; Under the assumption the analogue amplifier will be more liable to noise/distortion, we'll try and do as much as possible using the digital amplifier
; TODO - Once other channels start depending on the analogue amplifiers this technique of using the two amplifiers in unison probably won't be workable. It may be worth splitting the analogue amplifiers off into a seperate mixer so the user is given full control?
; First, apply digital coarse grain
MOV v4, #0
MOVS v3, v3, ASR #4
MOV ip, v3
ADDGT v4, v4, #&40 ; +6dB
SUBGTS v3, v3, #6
ADDGT v4, v4, #&40 ; +12dB
SUBGTS v3, v3, #6
; Now apply digital fine-grain
MOVGT v3, #0 ; Don't go any higher than 12dB
ADD lr, v4, v4, LSR #1
ADDS v3, v3, #&3F
SUB lr, lr, #63*16
MOVLE v3, #1 ; Clamp to -62dB
ADD lr, lr, v3, LSL #4
ORR v4, v4, v3
TST v2, #1
ORREQ v4, v4, v4, LSL #8 ; L channel settings
MOVNE v4, #0 ; Else disable everything
; We arrive here with:
; v4 = ARXR2PGA, ARXL2PGA settings
; ip = requested gain (dB)
; lr = digital gain achieved (dB*16)
; Apply analogue gain
RSB v3, ip, lr, ASR #4
BIC v3, v3, #1 ; Ensure multiple of 2dB
CMP v3, #-12 ; Actually +12dB
MOVLT v3, #-12
CMP v3, #18 ; Actually -18dB
MOVGT v3, #18
SUB lr, lr, v3, LSL #4 ; lr = final gain achieved
; Mung v3 into ARX(L/R)2_APGA_CTL values
MOV v3, v3, LSL #2
ADD v3, v3, #(6<<3)+3
ORR v3, v3, v3, LSL #8
TST v2, #1 ; Check mute
MOVNE v3, #0
; Now program everything
STR lr, [v7]
ADD a2, v5, #:INDEX:MixerSystemGain
LDRH ip, [a2]
MOV a3, #2
CMP ip, v4
BEQ %FT10
STRH v4, [a2]
MOV a4, #ARXR2PGA
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT10
DebugTX "SetMixSystem: TPS write failed!"
]
10
ADD a2, v5, #:INDEX:MixerSystemGain2
LDRH ip, [a2]
CMP ip, v3
EXIT EQ
STRH v3, [a2]
MOV a1, #TPSAUDIO_IIC*2
MOV a4, #ARXL2_APGA_CTL
BL TPSWrite
[ AudioDebug
CMP a1, #0
EXIT EQ
DebugTX "SetMixSystem: TPS write failed!"
]
EXIT
SetMixHeadsetMic ; TODO - Implement!
SetMixAuxInput ; TODO - Implement!
SetMixCarkitMic ; TODO - Implement!
EXIT
MixerGetMix
ADR a1, MixerSettings
ADD a1, a1, a2, LSL #3
LDMIA a1, {a1, a2}
MOV pc, lr
SetAndWaitForAntiPop
; Sets the CNCL_OFFSET_START bit in the ANAMICL register, then waits for it to clear
; Note: Assumes original register contents is &00!
; Also assumes that the codec power is on, otherwise (presumably) nothing will happen
; Input: sb = HAL workspace
Entry "a1-a4,v1", 4
MOV a1, #TPSAUDIO_IIC*2
MOV a2, sp
MOV a3, #1
MOV a4, #ANAMICL
LDR v1, OSentries+4*OS_IICOpV
MOV ip, #&a0
STR ip, [a2]
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT10
DebugTX "SetAndWaitForAntiPop: TPS write failed!"
]
10
; Linux uses a delay here to avoid spamming IIC, so we might as well use one too
MOV a1, #1024
BL HAL_CounterDelay
MOV a1, #TPSAUDIO_IIC*2
MOV a2, sp
MOV a3, #1
MOV a4, #ANAMICL
BL TPSRead
[ AudioDebug
CMP a1, #0
BEQ %FT15
DebugTX "SetAndWaitForAntiPop TPS read failed!"
15
]
CMP a1, #0
LDREQB a1, [a2]
TSTEQ a1, #&80 ; Check if CNCL_OFFSET_START is still set
BNE %BT10
EXIT
UpdateHeadset
; This performs the steps necessary to enable/disable the headset output
; Input:
; a1 = mixer device
; sb = HAL workspace
Entry "a1-a4,v1", 12
; Get mixer settings for headset
ADR lr, MixerSettings+MixerChannel_HeadsetOut*8
LDMIA lr, {a4,v1} ; a4 = mute flag, v1 = gain
; Calculate HS_GAIN_SET
MOV ip, #&a ; 0dB gain
MOV a3, #0
CMP v1, #6*16
MOVGE ip, #&5 ; 6dB gain
MOVGE a3, #6*16
CMP v1, #-6*16
MOVLE ip, #&f ; -6dB gain
MOVLE a3, #-6*16
STR a3, [lr, #4] ; Write back actual gain
TST a4, #1
ADRNE a2, HeadsetDisableSequence ; Muted, so disable it
MOVNE ip, #0
BNE %FT05
; Copy enable sequence to stack so we can modify it with the correct gain
ADR a3, HeadsetEnableSequence
MOV a2, sp
LDRH v1, MixerHeadsetGain
CMP v1, #0 ; Is the headset currently turned off?
LDMIA a3, {a3,a4,v1}
ORR a4, a4, ip, LSL #8 ; Set correct gain
STMEQIA a2, {a3,a4,v1} ; Full programming sequence if headset off
BICNE a4, a4, #&FF0000 ; If headset on, only change HS_GAIN_SET
STRNE a4, [a2]
05
; a2 = sequence to program
; ip = HS_GAIN_SET value
LDRH v1, MixerHeadsetGain
[ AudioDebug
DebugReg ip,"UpdateHeadset: New="
DebugReg v1,"Old="
]
CMP v1, ip
EXIT EQ ; Avoid reprogramming identical values
STRH ip, MixerHeadsetGain
MOV a3, #1
LDR v1, OSentries+4*OS_IICOpV
10
LDRB a4, [a2],#1
CMP a4, #255
LDREQB a4, [a2],#1
BLEQ DoHeadsetRampDelay
MOV a1, #TPSAUDIO_IIC*2
CMP a4, #0
EXIT EQ
BL TPSWrite
[ AudioDebug
CMP a1, #0
BEQ %FT15
DebugReg a4,"UpdateHeadset: Write failed, reg="
15
]
ADD a2, a2, #1
B %BT10
DoHeadsetRampDelay
; Wait for the headset anti-pop to take effect
Entry "a1-a4,ip"
LDR a1, HeadsetRampDelay
BL HAL_CounterDelay
EXIT
RAMP_DELAY_VAL * 3
HeadsetRampDelay
; Required delay is 2^(19+RAMP_DELAY_VAL)/26 microseconds, assuming 26MHz system clock
DCD (1:SHL:(19+RAMP_DELAY_VAL)) / 26
HeadsetEnableSequence
DCB HS_SEL, &24 ; 19. Select the headset outputs
DCB HS_POPN_SET,&40 + (RAMP_DELAY_VAL :SHL: 2) ; 34, 35. Program antipop bias ramp delay and enable vmid
DCB HS_GAIN_SET,&00 ; 36. Enable headset output stage & gain setting. NOTE - This gets filled in above.
DCB HS_POPN_SET,&42 + (RAMP_DELAY_VAL :SHL: 2) ; 37. Enable ramp
DCB &FF ; Wait for ramp time
DCB 0 ; Terminator
ALIGN
HeadsetDisableSequence
DCB HS_POPN_SET,&40 + (RAMP_DELAY_VAL :SHL: 2) ; Disable ramp
DCB &FF ; Wait for ramp time
DCB HS_GAIN_SET,&00 ; Disable gain
DCB HS_POPN_SET,&20 + (RAMP_DELAY_VAL :SHL: 2) ; Disable Vmid, enable external mute
DCB HS_SEL,&00 ; Disable headset outputs (for when external mute isn't available)
DCB 0 ; Terminator
ALIGN
END