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[alsa-devel] [topic/asoc][PATCH] ASoC: OMAP: Update contact addresses
by Jarkko Nikula 17 Apr '09

17 Apr '09

My email address is going to expire soon so update it. Adding also Peter Ujfalusi <peter.ujfalusi(a)nokia.com> as a second contact to OMAP core drivers since I won't have anymore access to non-public OMAP documentation in the future and Peter is working with these drivers as well. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 5 +++-- sound/soc/omap/omap-mcbsp.h | 3 ++- sound/soc/omap/omap-pcm.c | 5 +++-- sound/soc/omap/omap-pcm.h | 3 ++- 5 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e54e1c2..b60b1df 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -416,6 +416,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 4d6ef0b..495192a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -546,6 +547,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index df7ad13..c8147aa 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 1bdbb04..07cf7f4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void) } module_exit(omap_soc_platform_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index e4369bd..8d9d269 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- 1.6.2.1

2 1

[alsa-devel] [PATCH] us122l: Fix signedness in comparisions
by Karsten Wiese 17 Apr '09

17 Apr '09

Within 2.6.30's mergewindow, struct urb's transfer_buffer_length has become unsigned. This changed an "int > int" comparision to an "unsigned > int" one in snd_usb_122l. Fix this by using a local int variable instead of urb->transfer_buffer_length in comparisions. Shorten playback_prep_freqn() a bit and tweak error-paths in usb_stream_prepare_playback(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/usb_stream.c | 67 +++++++++++++++++++----------------------- 1 files changed, 30 insertions(+), 37 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 24393da..12ae034 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -33,32 +33,26 @@ static unsigned usb_stream_next_packet_size(struct usb_stream_kernel *sk) static void playback_prep_freqn(struct usb_stream_kernel *sk, struct urb *urb) { struct usb_stream *s = sk->s; - unsigned l = 0; - int pack; - - urb->iso_frame_desc[0].offset = 0; - urb->iso_frame_desc[0].length = usb_stream_next_packet_size(sk); - sk->out_phase = sk->out_phase_peeked; - urb->transfer_buffer_length = urb->iso_frame_desc[0].length; - - for (pack = 1; pack < sk->n_o_ps; pack++) { - l = usb_stream_next_packet_size(sk); - if (s->idle_outsize + urb->transfer_buffer_length + l > - s->period_size) + int pack, lb = 0; + + for (pack = 0; pack < sk->n_o_ps; pack++) { + int l = usb_stream_next_packet_size(sk); + if (s->idle_outsize + lb + l > s->period_size) goto check; sk->out_phase = sk->out_phase_peeked; - urb->iso_frame_desc[pack].offset = urb->transfer_buffer_length; + urb->iso_frame_desc[pack].offset = lb; urb->iso_frame_desc[pack].length = l; - urb->transfer_buffer_length += l; + lb += l; } - snd_printdd(KERN_DEBUG "%i\n", urb->transfer_buffer_length); + snd_printdd(KERN_DEBUG "%i\n", lb); check: urb->number_of_packets = pack; - s->idle_outsize += urb->transfer_buffer_length - s->period_size; + urb->transfer_buffer_length = lb; + s->idle_outsize += lb - s->period_size; snd_printdd(KERN_DEBUG "idle=%i ul=%i ps=%i\n", s->idle_outsize, - urb->transfer_buffer_length, s->period_size); + lb, s->period_size); } static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, @@ -282,21 +276,20 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct usb_stream *s = sk->s; struct urb *io; struct usb_iso_packet_descriptor *id, *od; - int p, l = 0; + int p = 0, lb = 0, l = 0; io = sk->idle_outurb; od = io->iso_frame_desc; - io->transfer_buffer_length = 0; - for (p = 0; s->sync_packet < 0; ++p, ++s->sync_packet) { + for (; s->sync_packet < 0; ++p, ++s->sync_packet) { struct urb *ii = sk->completed_inurb; id = ii->iso_frame_desc + ii->number_of_packets + s->sync_packet; l = id->actual_length; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } for (; @@ -304,38 +297,38 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, ++p, ++s->sync_packet) { l = inurb->iso_frame_desc[s->sync_packet].actual_length; - if (s->idle_outsize + io->transfer_buffer_length + l > - s->period_size) + if (s->idle_outsize + lb + l > s->period_size) goto check_ok; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } check_ok: s->sync_packet -= inurb->number_of_packets; - if (s->sync_packet < -2 || s->sync_packet > 0) { + if (unlikely(s->sync_packet < -2 || s->sync_packet > 0)) { snd_printk(KERN_WARNING "invalid sync_packet = %i;" " p=%i nop=%i %i %x %x %x > %x\n", s->sync_packet, p, inurb->number_of_packets, - s->idle_outsize + io->transfer_buffer_length + l, - s->idle_outsize, io->transfer_buffer_length, l, + s->idle_outsize + lb + l, + s->idle_outsize, lb, l, s->period_size); return -1; } - if (io->transfer_buffer_length % s->cfg.frame_size) { + if (unlikely(lb % s->cfg.frame_size)) { snd_printk(KERN_WARNING"invalid outsize = %i\n", - io->transfer_buffer_length); + lb); return -1; } - s->idle_outsize += io->transfer_buffer_length - s->period_size; + s->idle_outsize += lb - s->period_size; io->number_of_packets = p; - if (s->idle_outsize > 0) { - snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); - return -1; - } - return 0; + io->transfer_buffer_length = lb; + if (s->idle_outsize <= 0) + return 0; + + snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); + return -1; } static void prepare_inurb(int number_of_packets, struct urb *iu) -- 1.6.0.6

2 2

[alsa-devel] [PATCH] MAINTAINERS - Update SOUND
by Joe Perches 17 Apr '09

17 Apr '09

Signed-off-by: Joe Perches <joe(a)perches.com> diff --git a/MAINTAINERS b/MAINTAINERS index 0cb20d8..c81792f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5184,13 +5184,17 @@ F: drivers/memstick/host/tifm_ms.c SOUND P: Jaroslav Kysela M: perex(a)perex.cz P: Takashi Iwai M: tiwai(a)suse.de L: alsa-devel(a)alsa-project.org (subscribers-only) +W: http://www.alsa-project.org/main/index.php/Main_Page +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git S: Maintained +F: Documentation/sound/ +F: include/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) P: Liam Girdwood M: lrg(a)slimlogic.co.uk P: Mark Brown

2 1

[alsa-devel] [PATCH] ALSA: off by 1 in snd_emu10k1_wait()
by Roel Kluin 17 Apr '09

17 Apr '09

Is this useful? It was in Linus' git tree. ----------------------------->8-----------------8<------------------------------ With `while (count++ < 16384)' count reaches 16385. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 4bfc31d..c1a5aa1 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -490,7 +490,7 @@ void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait) if (newtime != curtime) break; } - if (count >= 16384) + if (count > 16384) break; curtime = newtime; }

2 1

[alsa-devel] [PATCH v2 0/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Naresh Medisetty 17 Apr '09

17 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 3 patches submitted to the DaVinci git: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x [PATCH] ARM: DaVinci: Fix the machine_is_davinci_dm6467_evm() error Patch 1: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 2: Adds McASP clock support for the two instances of mcasp casp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 6: Introduces platfrom driver model for dm644x and dm355 Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

3 2

[alsa-devel] [2.6.30-rc2][PATCH 0/2] Yet another DSP format fix into OMAP McBSP DAI
by Jarkko Nikula 17 Apr '09

17 Apr '09

Hopefully this is the last fourth DSP format fix into McBSP DAI... Tested with N810 modified to use DSP_B and with AIC33 codec. Also with Beagle by using McBSP3 routed to expansion pin header and using scope. Luckily TLV320AIC23 and TLV320AIC33 codec drivers seems to be correct but machine driver for OSK5912 must be corrected to not use inverted FS polarity since AIC23 doesn't have support for it. -- Jarkko

6 25

[alsa-devel] [ALSA - patch_sigmatel.c for HP dv4-1222nr laptop 9/9Patch to patch_sigmatel.c for HP dv4-1222nr laptop
by renidragsemaj＠yahoo.com 17 Apr '09

17 Apr '09

From: James Gardiner <renidragsemaj(a)yahoo.com> This patch to patch_sigmatel.c is for the HP dv4-1222nr laptop. It activates the included microphone(s) and the mute status change light for this computer. I don't know if this will work for any other dv4 laptop. The commit isn't perfect, for it doesn't seem to work too well with pulseaudio - the mic boost doesn't work with that audioserver. Otherwise, it works well with esound, even across reboots and suspend/hibernate/resume. The built-in microphone sits on node 0x18 and feeds through node 0x1c. The gpio for the mute light is GPIO #0. I borrowed the code from the HP HDX. Occasionally if you press the mute button too fast in succession it lowers the actual volume to 0 and you'll have to adjust it again...But if you are doing that you probably are getting a kick out of making me feel bad about the behavior. :) No autodetect of the laptop seems to work for me, but "model=hp-dv4-1222nr" as an option to modprobe will make the changes. Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61996a2..4286155 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1847,6 +1848,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4534,6 +4536,28 @@ static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, } #endif +static int stac92xx_hp_dv4_1222nr_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + /* Shamelessly stolen from the function stac92xx_hp_hdx_check_power_status */ + /* by James Gardiner */ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x01; /* orange */ + else + spec->gpio_data |= 0x01; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -5202,6 +5226,19 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: /* James Gardiner */ + spec->num_dmics = 1; + spec->num_dmuxes = 2; + stac92xx_auto_set_pinctl(codec, 0x1c, 2); + stac92xx_auto_set_pinctl(codec, 0x18, AC_PINCTL_IN_EN); +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_dv4_1222nr_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); -- 1.5.6.3

3 3

[alsa-devel] [PATCH v2 3/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Naresh Medisetty 17 Apr '09

17 Apr '09

1)Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2)Move towards definition of the asp clocks using phyisical name(for dm355 and dm644x) Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- arch/arm/mach-davinci/dm646x.c | 67 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- 4 files changed, 157 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index c20211c..222c3f3 100755 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -23,6 +23,7 @@ #include <mach/psc.h> #include <mach/mux.h> #include <mach/irqs.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -351,8 +352,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -540,6 +541,29 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +}; + +static struct platform_device dm355_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm355_evm_snd_data, + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), + .resource = dm355_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ void __init dm355_init(void) @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) davinci_cfg_reg(DM355_INT_EDMA_CC); platform_device_register(&dm355_edma_device); + platform_device_register(&dm355_asoc_device); return 0; } postcore_initcall(dm355_init_devices); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 9878c17..01112ca 100755 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -296,7 +297,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -463,6 +464,29 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_evm_snd_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +static struct platform_device dm644x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm644x_evm_snd_data, + .num_resources = ARRAY_SIZE(dm644x_evm_snd_resources), + .resource = dm644x_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -495,6 +519,7 @@ static int __init dm644x_init_devices(void) return 0; platform_device_register(&dm644x_edma_device); + platform_device_register(&dm644x_asoc_device); return 0; } postcore_initcall(dm644x_init_devices); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 39ffd17..bb8c65a 100755 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -499,6 +500,70 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + +static struct resource dm6467_evm_snd_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct platform_device dm646x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = dm6467_evm_snd_data, + .num_resources = ARRAY_SIZE(dm6467_evm_snd_resources), + .resource = dm6467_evm_snd_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "davinci-dit", + .id = -1, + +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -532,6 +597,8 @@ static int __init dm646x_init_devices(void) return 0; platform_device_register(&dm646x_edma_device); + platform_device_register(&dm646x_asoc_device); + platform_device_register(&dm646x_dit_device); return 0; } postcore_initcall(dm646x_init_devices); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..1bb1f0b 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,56 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct evm_snd_platform_data { + char *clk_name; + int tx_dma_ch; + int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ -- 1.5.6

1 0

[alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Naresh Medisetty 17 Apr '09

17 Apr '09

Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index f4a71da..39ffd17 100755 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -220,6 +220,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -307,6 +319,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

1 0

[alsa-devel] [PATCH v2 1/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Naresh Medisetty 17 Apr '09

17 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). The codec on the DM646x EVM is AIC32 which is connected to Serializer0 and Serializer1 of McASP0. The McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 15 +- sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-i2s-mcasp.c | 890 +++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s-mcasp.h | 56 ++ sound/soc/davinci/davinci-pcm.h | 17 +- 5 files changed, 972 insertions(+), 9 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c..0ae84d4 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM @@ -16,7 +19,17 @@ config SND_DAVINCI_SOC_EVM select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. + +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM6467 EVM platform. + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..5e9ba49 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,890 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + + + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_i2s_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_i2s_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int davinci_i2s_mcasp_probe(struct platform_device *pdev) +{ + int tmp, link_cnt, num_links = 2; + int count = 0; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (num_links << 1), GFP_KERNEL); + + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + for (link_cnt = 0; link_cnt < num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, pdata->clk_name); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + goto err_release_region; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + davinci_iis_mcasp_dai[link_cnt].private_data = &dev[link_cnt]; + } + ret = snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + for (tmp = link_cnt; tmp >= 0; tmp--) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_i2s_mcasp_remove(struct platform_device *pdev) +{ + int i, num_links = 2; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *dma_data; + + snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + + for (i = 0; i < num_links; i++) { + dev = davinci_iis_mcasp_dai[i].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + + dev = davinci_iis_mcasp_dai[0].private_data; + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + return 0; +} + +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_i2s_mcasp_init(void) +{ + + return platform_driver_register(&davinci_asoc_driver); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + + platform_driver_unregister(&davinci_asoc_driver); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..6469e7d --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,56 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#ifndef DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..3cbe281 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,18 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

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