July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] XRUN handling
by melwyn lobo 19 Jul '10

19 Jul '10

Hi all, I have noticed when an Xrun occurs, the bytes between the hw_ptr and appl_ptr are not copied when both hw and appl pointers are reset in snd_pcm_prepare(). Consider the following parameters: buffer_size : 22352 period_size : 1016 period_time : 23038 tstamp_mode : NONE period_step : 1 avail_min : 1016 period_event : 0 start_threshold : 22352 stop_threshold : 22352 silence_threshold: 0 silence_size : 0 In case snd_pcm_playback_avail() is less than stop threshold but greater than 0, then these bytes are missing after Xrun prepare and start stage. Is my analysis correct? If so how do we ensure correct behaviour ie byte accuracy after Xrun. Thanks, M.

4 11

[alsa-devel] [PATCH 2/2]ASoC: multi-component - Fix S5P compile error
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch fix compile error on using I2S and WM8994 common code. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..55bc2be 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif

2 2

[alsa-devel] [PATCH 1/2]ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(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 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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] Clicking and stuttering audio when playing two or more files
by Adam Rosenberg 19 Jul '10

19 Jul '10

I have audio output perfectly from my second CS42448 CODEC now, but only when one file is playing. When I attempt to play two files, one or both of the files play with clicking and stuttering. I am also noticing that the PCM prepare operation is occurring more than once. What could be causing this problem? HW settings are: buffer_bytes_max = 32768 period_bytes min = 8 max = 32768 periods min = 1 max = 4096 PCM runtime is reporting: frames per period = 1114 periods = 2 total frames = 2228 frame bytes = 8 Thank you, Adam

2 1

[alsa-devel] [PATCH] Add conexant quirk for Lenovo G series support
by Jerone Young 19 Jul '10

19 Jul '10

This patch adds quirk add support for Lenovo G series laptops. Launchpad bug can be found here: https://bugs.launchpad.net/ubuntu/+source/linux-backports-modules-2.6.32/+b… Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 54f7419..c49030b 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2999,6 +2999,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), {} };

1 0

[alsa-devel] [PATCH] asoc: davinci: let platform data define edma queue numbers
by Sekhar Nori 17 Jul '10

17 Jul '10

Currently the EDMA queue to be used by for servicing ASP through internal RAM is fixed to EDMAQ_0 and that to service internal RAM from external RAM is fixed to EDMAQ_1. This may not be the desirable configuration on all platforms. For example, on DM365, queue 0 has large fifo size and is more suitable for video transfers. Having audio and video transfers on the same queue may lead to starvation on audio side. platform data as defined currently passes a queue number to the driver but that remains unused inside the driver. Fix this by defining one queue each for ASP and RAM transfers in the platform data and using it inside the driver. Since EDMAQ_0 maps to 0, thats the queue that will be used if the asp queue number is not initialized. None of the platforms currently utilize ping-pong transfers through internal RAM so that functionality remains unchanged too. This patch has been tested on DM644x and OMAP-L138 EVMs. Signed-off-by: Sekhar Nori <nsekhar(a)ti.com> --- This patch applies to latest of Linus's tree. arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 ++-- arch/arm/mach-davinci/include/mach/asp.h | 3 ++- sound/soc/davinci/davinci-i2s.c | 10 ++++++++++ sound/soc/davinci/davinci-mcasp.c | 6 ++++-- sound/soc/davinci/davinci-pcm.c | 5 +++-- sound/soc/davinci/davinci-pcm.h | 3 ++- 8 files changed, 25 insertions(+), 10 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 212d970..bc384d3 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -208,7 +208,7 @@ static struct snd_platform_data da830_evm_snd_data = { .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da830_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index 2ec3095..d4ec18d 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -342,7 +342,7 @@ static struct snd_platform_data da850_evm_snd_data = { .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da850_iis_serializer_direction, - .eventq_no = EVENTQ_1, + .asp_chan_q = EVENTQ_1, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index 6d88893..87521f2 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -323,7 +323,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), .tdm_slots = 2, .serial_dir = dm646x_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, { .tx_dma_offset = 0x400, @@ -332,7 +332,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), .tdm_slots = 32, .serial_dir = dm646x_dit_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, }; diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..5d0a4c7 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -52,7 +52,8 @@ struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..4c1a7d2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -526,6 +526,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; + enum dma_event_q asp_chan_q = EVENTQ_0; + enum dma_event_q ram_chan_q = EVENTQ_1; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -552,7 +554,15 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + asp_chan_q = pdata->asp_chan_q; + ram_chan_q = pdata->ram_chan_q; } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d395509..b247208 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); @@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..f026ac5 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream, /* Request ram master channel */ link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, - EVENTQ_1); + prtd->params->ram_chan_q); if (link < 0) goto exit1; @@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, EVENTQ_0); + davinci_pcm_dma_irq, substream, + prtd->params->asp_chan_q); if (link < 0) goto exit1; diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..b799a02 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -21,7 +21,8 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: fsi: Add specified ID for soc-audio
by Kuninori Morimoto 17 Jul '10

17 Jul '10

Specified ID is necessary, when some codecs are used with FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark. Thank you about your advice. it solved my issue. This patch is v2 of "ASoC: FSI - codecs settings" include/sound/sh_fsi.h | 3 +++ sound/soc/sh/fsi-ak4642.c | 4 ++-- sound/soc/sh/fsi-da7210.c | 4 ++-- 3 files changed, 7 insertions(+), 4 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..6463fd6 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -12,6 +12,9 @@ * published by the Free Software Foundation. */ +#define FSI_PORT_A 0 +#define FSI_PORT_B 1 + /* flags format * 0xABCDEEFF diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index be01854..2c61ed2 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -38,7 +38,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", - .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_A], .codec_dai = &ak4642_dai, .init = fsi_ak4642_dai_init, .ops = NULL, @@ -62,7 +62,7 @@ static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - fsi_snd_device = platform_device_alloc("soc-audio", -1); + fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); if (!fsi_snd_device) goto out; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 33b4d17..5774449 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -33,7 +33,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_da7210_dai = { .name = "DA7210", .stream_name = "DA7210", - .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_B], .codec_dai = &da7210_dai, .init = fsi_da7210_init, }; @@ -56,7 +56,7 @@ static int __init fsi_da7210_sound_init(void) { int ret; - fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); if (!fsi_da7210_snd_device) return -ENOMEM; -- 1.7.0.4

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[alsa-devel] [PATCH 13/15] sound/soc: davinci: check kzalloc() result (typo)
by Kulikov Vasiliy 17 Jul '10

17 Jul '10

The code checks 'davinci_vc' after kzalloc() and do not checks 'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that it is a typo (autocompletion?). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9aa980d..4867853 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) int ret; davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); - if (!davinci_vc) { + if (!davinci_vcif_dev) { dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); return -ENOMEM; -- 1.7.0.4

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[alsa-devel] [PATCH v2] ASoC: TWL6040: Correct widget handling for drivers
by Jorge Eduardo Candelaria 17 Jul '10

17 Jul '10

In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 1 files changed, 28 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 715caef..345d86a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + msleep(1); + return 0; +} + static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, else priv->non_lp--; + msleep(1); + return 0; } @@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls = static const struct snd_kcontrol_new hfdacr_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); -/* Headset driver switches */ -static const struct snd_kcontrol_new hsl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); - -static const struct snd_kcontrol_new hsr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); - -/* Handsfree driver switches */ -static const struct snd_kcontrol_new hfl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); - -static const struct snd_kcontrol_new hfr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); - static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HFDAC Right Playback", SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Left Driver", - SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Right Driver", - SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + /* Analog playback drivers */ + SND_SOC_DAPM_PGA_E("Handsfree Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + SND_SOC_DAPM_PGA_E("Handsfree Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSDAC Left Playback", "Switch", "HSDAC Left"}, {"HSDAC Right Playback", "Switch", "HSDAC Right"}, - {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, - {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + {"Headset Left Driver", NULL, "HSDAC Left Playback"}, + {"Headset Right Driver", NULL, "HSDAC Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, -- 1.7.0.4

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[alsa-devel] [PATCHv2 0/4] ASoC: multi-component - Samsung S3C Platform
by Seungwhan Youn 17 Jul '10

17 Jul '10

Hi, This is a second version of patch series adding support for multi component model on Samsung S3C Platform. This patch series can be apply upon "ASoC: multi-component - Fix S3C compile error" in the first patch of V1 that already applied by Liam. Changes since V1: - Support for GPIO, DMA and SFR settings pull from platform device resource. - Modify description of add WM9713 platform device on smdk board. The following patch series contains: [PATCHv2 1/4] ASoC: S3C64XX: Move GPIO settings to platform [PATCHv2 2/4] ASoC: S3C64XX: Move DMA settings to platform [PATCHv2 3/4] ASoC: multi-component - S3C - Get original functionality [PATCHv2 4/4] ASoC: multi-component - S3C - Add WM9713 platform device Thanks, Claude

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