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[alsa-devel] [PATCH] cs4236: update control names
by Krzysztof Helt 18 Nov '09

18 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version of the patch. The Master Digital Playback is not renamed to the Master DAC Playback. include/sound/cs4231-regs.h | 1 - sound/isa/cs423x/cs4236_lib.c | 49 +++++++++++++++++----------------------- sound/isa/wss/wss_lib.c | 8 +++--- 3 files changed, 25 insertions(+), 33 deletions(-) diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h index 9264753..66d28c2 100644 --- a/include/sound/cs4231-regs.h +++ b/include/sound/cs4231-regs.h @@ -70,7 +70,6 @@ #define AD1845_PWR_DOWN 0x1b /* power down control */ #define CS4235_LEFT_MASTER 0x1b /* left master output control */ #define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ -#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ #define AD1845_CLOCK 0x1d /* crystal clock select and total power down */ #define CS4235_RIGHT_MASTER 0x1d /* right master output control */ #define CS4231_REC_UPR_CNT 0x1e /* record upper count */ diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 1b1ad1c..4c4024a 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -777,7 +777,7 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, +CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, @@ -798,10 +798,10 @@ WSS_DOUBLE("CD Capture Switch", 0, CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), @@ -815,31 +815,27 @@ CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, static struct snd_kcontrol_new snd_cs4235_controls[] = { -WSS_DOUBLE("Master Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Volume", 0, +WSS_DOUBLE("Master Playback Volume", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), CS4235_OUTPUT_ACCU("Playback Volume", 0), -CS4236_DOUBLE("Master Digital Playback Switch", 0, - CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, - CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), - -WSS_DOUBLE("Master Digital Playback Switch", 1, +WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Master Digital Capture Switch", 1, +WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Master Digital Volume", 1, +WSS_DOUBLE("Synth Volume", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), CS4236_DOUBLE("Capture Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -WSS_DOUBLE("PCM Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), WSS_DOUBLE("PCM Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), @@ -855,28 +851,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), -CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), +CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), -WSS_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 0, +WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 0, +WSS_DOUBLE("Line Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -WSS_DOUBLE("Aux Playback Switch", 1, +WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 1, +WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 1, +WSS_DOUBLE("CD Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4236_DOUBLE1("Master Mono Switch", 0, - CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), - -CS4236_DOUBLE1("Mono Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), WSS_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 705db09..5b9d6c1 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2224,7 +2224,7 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, -WSS_DOUBLE("Mic Boost", 0, +WSS_DOUBLE("Mic Boost (+20dB)", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), @@ -2235,14 +2235,14 @@ WSS_DOUBLE("Line Playback Switch", 0, WSS_DOUBLE_TLV("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, +WSS_SINGLE("Beep Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; -- 1.6.4 ---------------------------------------------------------------------- Kawalerka za 79 000 zl! 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Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 18 Nov '09

18 Nov '09

Am Sonntag 12 Juli 2009 10:24:35 schrieb Christian Esken: > Am Montag 06 Juli 2009 08:26:11 schrieb Takashi Iwai: > > At Sun, 5 Jul 2009 13:29:48 +0200, > > > > Christian Esken wrote: > > > Am Sonntag 05 Juli 2009 09:44:33 schrieb Takashi Iwai: > > > > At Sun, 5 Jul 2009 00:19:55 +0200, > > > > > > > > Christian Esken wrote: > > > > > Am Samstag 27 Juni 2009 14:05:45 schrieb Christian Esken: > > > > > > Hello, > > > > > > > > > > > > unfortunately I cannot playback anything (PCM) with the new SB > > > > > > X-Fi driver. Driver: > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- > > > > > > driver-snapshot.tar.bz2 > > > > > > Distribution: openSuSE 11.1 > > > > > > Kernel: 2.6.27.23-0.1-xen > > > > > > > > > > > > My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The > > > > > > card is recognized by the driver, and any mixer (alsamixer, kmix) > > > > > > also recognizes the card. > > > > > > > > > > > > But playing back isn't possible. I tried speaker-test, audacity, > > > > > > amarok and aplay. Most applications hang, some also produce an > > > > > > error in the /var/log/messages (see > > > > > > below). > > > > > > > > > > > > I then used "./configure --with-debug=full --with-cards=ctxfi" > > > > > > due to the possible hd-audio conflict, but it didn't make a > > > > > > difference. > > > > > > > > > > > > Is there anything else I could try? > > > > > > > > > > I tested the same openSuSE version with a non-Xen 64 bit kernel, > > > > > and it doesn't work either. > > > > > > > > > > And I tried with this: > > > > > Driver: alsa-driver-20090704.tar.bz2 > > > > > Distribution: Kubuntu Karmic Alpha-2 > > > > > Kernel: 2.6.30 (32 Bit) > > > > > > > > > > Under the latter setup the X-Fi Titanium works (Simple 2 channel > > > > > test with speaker-test, audacity, amarok). :-)) > > > > > > > > How many RAM do you have? > > > > Could you boot with the limited memory size? > > > > > > It is just 2GB. So I am not limited by the 32 bit barrier. > > > > > > So booting with "limited" memory size doesn't make sense, right? > > > > Supposedly. Is it emu20k1 or emu20k2? > > I checked it again: It is a 20k2 chip. > > /var/log/boot.msg says: > <7>ALSA /home/chris/Desktop/alsa-driver/pci/ctxfi/ctatc.c:1268: ctxfi: chip > 20K2 model SB0880 (1102:0041) is found > > > http://pci-ids.ucw.cz/mods/PC/1102/000b shows 1102:0041 as "X-Fi Titanium > series [EMU20k2]". I did now test it with a different 64 Bit Linux distribution, and it works correctly: Driver: alsa-driver-20090712.tar.bz2 Distribution: Kubuntu Karmic Alpha-2 Kernel: 2.6.30 (64 Bit) All in all, I can produce a working environment with recent 32 and 64 bit kernels. This is good news. :-) Below is the summary on the tested distributions. Christian So we got this: Kubuntu Karmic Alpha-2 32 Bit, 2.0.30 (works) Kubuntu Karmic Alpha-2 64 Bit, 2.0.30 (works) SuSE 11.1 64 Bit, 2.0.27 (does not work) SuSE 11.1 64 Bit, 2.0.27 XEN (does not work) SuSE 11.2 Milestone3 64Bit, 2.0.30 (test canceled, due to unrelated problems) => I'll see whether I can redo the SuSE11.2 tests, after doing a bugreport at openSuSE.

3 6

[alsa-devel] [PATCH 00/11] Intel HDMI fixes
by Wu Fengguang 18 Nov '09

18 Nov '09

Hi Takashi, Here is some Intel HDMI fixes, most of them are inspired/written by David Härdeman (Big thanks!). - fix audio infoframe size - sticky infoframe/stream id/channel count to - not disturb (David's) fragile devices - avoid the first 0.5s audio samples be losted - avoid losing presence info during - module reloading (by querying the pin sense status) - power off (by disabling it) Thanks, Fengguang

2 13

[alsa-devel] There is no sound from speaker on Toshiba NB200
by Bin Gao 18 Nov '09

18 Nov '09

[re-send because the previous one is not plain text format) The quirk "SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4)" works on Toshiba NB200 but for playback, I only get sound from HP and never from SPK. As observed by hda-verb, the HP pipeline is DAC (nid 0x03) --> Mux (nid 0x0D) --> pin HP-OUT (nid 0x21) and it works fine. I noted that unsol event(HP plugged and unplugged) can be detected by the codec ALC272. According to the ALC272 datasheet, the SPK pin seems either 0x14 or 0x15, but there was still no sound from SPK even I manually set up the pipeline(DAC 0x03 --> Mux 0x0D --> pin 0x14 or 0x15) by hda-verb. For ALC663_ASUS_MODE4, dacs are defined as static hda_nid_t alc662_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04 }; and hp_nid is defined as 0x03. So as I understand, one HDA stream(from HDA controller) is copied to both DAC 0x02 and 0x03. Then I manually set up another pipeline (DAC 0x02 --> Mux 0x0C --> pin 0x14 or 0x15) , still no sound. I checked the init verbs of ALC663_ASUS_MODE4 from patch_realteck.c and didn't find any missing settings. Any ideas? Thanks, Bin

1 1

[alsa-devel] There is no sound from speaker on Toshiba NB200
by Bin Gao 18 Nov '09

18 Nov '09

The quirk "SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4)" works on Toshiba NB200 but for playback, I only get sound from HP and never from SPK. As observed by hda-verb, the HP pipeline is DAC (nid 0x03) --> Mux (nid 0x0D) --> pin HP-OUT (nid 0x21) and it works fine. I noted that unsol event(HP plugged and unplugged) can be detected by the codec ALC272. According to the ALC272 datasheet, the SPK pin seems either 0x14 or 0x15, but there was still no sound from SPK even I manually set up the pipeline(DAC 0x03 --> Mux 0x0D --> pin 0x14 or 0x15) by hda-verb. For ALC663_ASUS_MODE4, dacs are defined as static hda_nid_t alc662_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04 }; and hp_nid is defined as 0x03. So as I understand, one HDA stream(from HDA controller) is copied to both DAC 0x02 and 0x03. Then I manually set up another pipeline (DAC 0x02 --> Mux 0x0C --> pin 0x14 or 0x15) , still no sound. I checked the init verbs of ALC663_ASUS_MODE4 from patch_realteck.c and didn't find any missing settings. Any ideas? Thanks, Bin

1 0

[alsa-devel] [PATCHv2 2/7] S3C AUDIO: Rename 's3c24xx-pcm' driver to 's3c-dma'
by Jassi Brar 18 Nov '09

18 Nov '09

Making room for namespace for the PCM Controller driver the platform driver(s3c24xx-pcm) has been renamed to SoC agnostic name 's3c-dma'. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Makefile | 2 +- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 480 ++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-dma.h | 31 ++ sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 480 ------------------------ sound/soc/s3c24xx/s3c24xx-pcm.h | 31 -- sound/soc/s3c24xx/s3c24xx_simtec.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 2 +- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 2 +- 20 files changed, 527 insertions(+), 527 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-dma.c create mode 100644 sound/soc/s3c24xx/s3c-dma.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-pcm.h diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 7790406..3bdb74d 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,5 +1,5 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c24xx-pcm.o +snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 93e6c87..ac3a709 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -25,7 +25,7 @@ #include <asm/mach-types.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 12c7148..3896e52 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -24,7 +24,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 26409a9..1d05137 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <mach/gta02.h> #include "../codecs/wm8753.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 77de6c5..9183166 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -36,7 +36,7 @@ #include "../codecs/wm8753.h" #include "lm4857.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" /* define the scenarios */ diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c new file mode 100644 index 0000000..cb49400 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -0,0 +1,481 @@ +/* + * s3c-dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma.h> + +#include "s3c-dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; + unsigned long totbytes = params_buffer_bytes(params); + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +static int s3c_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, +}; + +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h new file mode 100644 index 0000000..8cbc071 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-dma.h @@ -0,0 +1,31 @@ +/* + * s3c-dma.h -- + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_soc_platform s3c24xx_soc_platform; +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 5a442aa..13c99ab 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -35,7 +35,7 @@ #include <mach/dma.h> #include "s3c-i2s-v2.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #undef S3C_IIS_V2_SUPPORTED diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 23718ea..553c480 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -37,7 +37,7 @@ #include <mach/regs-gpio.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 678b176..000dc79 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -35,7 +35,7 @@ #include <asm/dma.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" struct s3c24xx_ac97_info { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index afb4bc9..ed83747 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -38,7 +38,7 @@ #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c deleted file mode 100644 index cb49400..0000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ /dev/null @@ -1,480 +0,0 @@ -/* - * s3c24xx-pcm.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/io.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "s3c24xx-pcm.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -static void s3c_dma_enqueue(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) { - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - } else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -static int s3c_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; - unsigned long totbytes = params_buffer_bytes(params); - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int s3c_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c_dma_hw_params, - .hw_free = s3c_dma_hw_free, - .prepare = s3c_dma_prepare, - .trigger = s3c_dma_trigger, - .pointer = s3c_dma_pointer, - .mmap = s3c_dma_mmap, -}; - -static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h deleted file mode 100644 index 8cbc071..0000000 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c24xx-pcm.h -- - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d..ef75416 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -21,7 +21,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index 8346bd9..61166a5 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index 25797e0..af0bf57 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index c215d32..e8d6cc2 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -24,7 +24,7 @@ #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 719d63c..64c034b 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -35,7 +35,7 @@ #include <mach/map.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" static struct s3c2410_dma_client s3c64xx_dma_client_out = { diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index a2a4f53..106c72d 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -20,7 +20,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index cb8a916..7956a9c 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -19,7 +19,7 @@ #include <sound/soc-dapm.h> #include "../codecs/wm8580.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" #define S3C64XX_I2S_V4 2 -- 1.6.2.5

2 1

[alsa-devel] [PATCHv2 1/7] S3C AUDIO: Rename s3c24xx_pcm prefix to s3c_dma
by Jassi Brar 18 Nov '09

18 Nov '09

The s3c24xx_pcm prefix for the soc_platform is inappropriate when some Samsung SoCs have PCM controllers which will eventually have drivers and hence namespace ambiguities. To resolve naming ambiguities in future the following have been renamed in order 1) s3c24xx_pcm_dma_params -> s3c_dma_params 2) s3c24xx_pcm_preallocate_dma_buffer -> s3c_preallocate_dma_buffer 3) s3c24xx_pcm_dmamask -> s3c_dma_mask 4) s3c24xx_pcm_XXX -> s3c_dma_XXX Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.h | 4 +- sound/soc/s3c24xx/s3c2412-i2s.c | 4 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 ++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 10 ++-- sound/soc/s3c24xx/s3c24xx-pcm.c | 76 +++++++++++++++++++------------------- sound/soc/s3c24xx/s3c24xx-pcm.h | 6 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 4 +- 8 files changed, 58 insertions(+), 58 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 28b0ab2..5a442aa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -394,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index f66854a..ecf8eaa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,8 +49,8 @@ struct s3c_i2sv2_info { unsigned char master; - struct s3c24xx_pcm_dma_params *dma_playback; - struct s3c24xx_pcm_dma_params *dma_capture; + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; u32 suspend_iismod; u32 suspend_iiscon; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ac5e47b..23718ea 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -50,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index b25e9f9..678b176 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -188,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = { .name = "AC97 Mic Mono in" }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { .client = &s3c2443_dma_client_out, .channel = DMACH_PCM_OUT, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { .client = &s3c2443_dma_client_in, .channel = DMACH_PCM_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { +static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { .client = &s3c2443_dma_client_micin, .channel = DMACH_MIC_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, @@ -290,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); @@ -339,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index c76b8bb..afb4bc9 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, .dma_size = 2, }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, @@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; default: @@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 151a694..cb49400 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -32,7 +32,7 @@ #include "s3c24xx-pcm.h" -static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { +static const struct snd_pcm_hardware s3c_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -62,15 +62,15 @@ struct s3c24xx_runtime_data { dma_addr_t dma_start; dma_addr_t dma_pos; dma_addr_t dma_end; - struct s3c24xx_pcm_dma_params *params; + struct s3c_dma_params *params; }; -/* s3c24xx_pcm_enqueue +/* s3c_dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; @@ -132,19 +132,19 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, spin_lock(&prtd->lock); if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); } spin_unlock(&prtd->lock); } -static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); int ret = 0; @@ -197,7 +197,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; @@ -214,7 +214,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) +static int s3c_dma_prepare(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -247,12 +247,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); return ret; } -static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -287,7 +287,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } static snd_pcm_uframes_t -s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +s3c_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -322,7 +322,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, res); } -static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) +static int s3c_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; @@ -330,7 +330,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); if (prtd == NULL) @@ -342,7 +342,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) +static int s3c_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -350,14 +350,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); if (!prtd) - pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c_dma_close called with prtd == NULL\n"); kfree(prtd); return 0; } -static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, +static int s3c_dma_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -370,23 +370,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops s3c24xx_pcm_ops = { - .open = s3c24xx_pcm_open, - .close = s3c24xx_pcm_close, +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c24xx_pcm_hw_params, - .hw_free = s3c24xx_pcm_hw_free, - .prepare = s3c24xx_pcm_prepare, - .trigger = s3c24xx_pcm_trigger, - .pointer = s3c24xx_pcm_pointer, - .mmap = s3c24xx_pcm_mmap, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, }; -static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; + size_t size = s3c_dma_hardware.buffer_bytes_max; pr_debug("Entered %s\n", __func__); @@ -401,7 +401,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) return 0; } -static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -424,9 +424,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); +static u64 s3c_dma_mask = DMA_BIT_MASK(32); -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -434,19 +434,19 @@ static int s3c24xx_pcm_new(struct snd_card *card, pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c24xx_pcm_dmamask; + card->dev->dma_mask = &s3c_dma_mask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; @@ -457,9 +457,9 @@ static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_platform s3c24xx_soc_platform = { .name = "s3c24xx-audio", - .pcm_ops = &s3c24xx_pcm_ops, - .pcm_new = s3c24xx_pcm_new, - .pcm_free = s3c24xx_pcm_free_dma_buffers, + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); @@ -476,5 +476,5 @@ static void __exit s3c24xx_soc_platform_exit(void) module_exit(s3c24xx_soc_platform_exit); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h index 0088c79..8cbc071 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h @@ -9,13 +9,13 @@ * ALSA PCM interface for the Samsung S3C24xx CPU */ -#ifndef _S3C24XX_PCM_H -#define _S3C24XX_PCM_H +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H #define ST_RUNNING (1<<0) #define ST_OPENED (1<<1) -struct s3c24xx_pcm_dma_params { +struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index d68cae1..719d63c 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -46,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { [0] = { .channel = DMACH_I2S0_OUT, .client = &s3c64xx_dma_client_out, @@ -61,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { }, }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { [0] = { .channel = DMACH_I2S0_IN, .client = &s3c64xx_dma_client_in, -- 1.6.2.5

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[alsa-devel] [PATCHv2 5/7] S3C AUDIO: Added platform data header
by Jassi Brar 18 Nov '09

18 Nov '09

Many SoCs have their I2S pins MUXed with other functions. So we need to pass a callback for driver to configure the pins appropriately. Hence, the need of platform data and this header. As and when needed new callbacks and structure pointers maybe added to this header. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c/include/plat/audio.h | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/audio.h diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h new file mode 100644 index 0000000..f22d23b --- /dev/null +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -0,0 +1,17 @@ +/* arch/arm/plat-s3c/include/plat/audio.h + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * 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. + */ + +/** + * struct s3c_audio_pdata - common platform data for audio device drivers + * @cfg_gpio: Callback function to setup mux'ed pins in I2S/PCM/AC97 mode + */ +struct s3c_audio_pdata { + int (*cfg_gpio)(struct platform_device *); +}; -- 1.6.2.5

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[alsa-devel] Quirk to disable master volume control in PCM2702
by Javier Kohen 17 Nov '09

17 Nov '09

[Alan Stern on BCC] Hi ALSA developers, I've written the attached patch to disable the master volume control in this chipset. The datasheet [1] documents two independent channel volume controls, one master mute control and one master volume control. However, at least on my device (a Fubar USB DAC), the master volume control always returns USB stalls when queried for any of its parameters (min/max/res/cur). All other controls work as expected, both at the protocol and at the application levels. Read on for the gory details. Otherwise please take the patch at face value. I'm ready to make changes if necessary to make it more fitting to the rest of the code. It would be very appreciated if this can go into 2.6.32, or in a stable patch shortly after. Normally I wouldn't give this problem a second thought, since in the worst case a non-functional control appears in the mixer. Only that Alan Stern submitted a patch to the USB subsystem during the development of kernel 2.6.31 that enables the sending of the Clear-TT-Buffer request to USB hubs on stalls from devices. For some reason upon receiving the Clear-TT-Buffer the audio goes mute, even though the kernel keeps sending the audio stream to the device. There is an ongoing thread on the linux-usb ML where Alan argues that sending the request is the correct behavior and it shouldn't have the effect I'm seeing. Since I lack any knowledge of the USB protocol, I take his word. However, it seems that either the USB specs are wrong, or my USB hub or the card are buggy. In the meantime I'm trying to get another hub to compare. Unfortunately I have to rely on the hub, because the PCM2702 and my motherboard don't interact well. After a while the sound is degraded, first into a hiss, then into pure noise. Other users have complained on different forums; one of the wrote to alsa-user a month ago. I haven't been able to make any progress to track this issue with usbmon and ALSA verbose debug messages, to nobody's surprise, since I'm a complete layman in that area. Any help with this would be appreciated as well. [1] http://focus.ti.com/docs/prod/folders/print/pcm2702.html

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[alsa-devel] [PATCHv2 4/7] S3C64XX PCM: Added PCLK clock source for the PCM controllers
by Jassi Brar 17 Nov '09

17 Nov '09

Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c64xx/clock.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-s3c64xx/clock.c b/arch/arm/plat-s3c64xx/clock.c index 7a36e89..3ab564e 100644 --- a/arch/arm/plat-s3c64xx/clock.c +++ b/arch/arm/plat-s3c64xx/clock.c @@ -129,6 +129,18 @@ static struct clk init_clocks_disable[] = { .enable = s3c64xx_pclk_ctrl, .ctrlbit = S3C_CLKCON_PCLK_IIS1, }, { + .name = "pcm", + .id = 0, + .parent = &clk_p, + .enable = s3c64xx_pclk_ctrl, + .ctrlbit = S3C_CLKCON_PCLK_PCM0, + }, { + .name = "pcm", + .id = 1, + .parent = &clk_p, + .enable = s3c64xx_pclk_ctrl, + .ctrlbit = S3C_CLKCON_PCLK_PCM1, + }, { .name = "spi", .id = 0, .parent = &clk_p, -- 1.6.2.5

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