litmus-rt.git - The LITMUS^RT kernel.

diff options

author	David S. Miller <davem@davemloft.net>	2006-01-31 21:31:38 -0500
committer	David S. Miller <davem@sunset.davemloft.net>	2006-03-20 04:11:18 -0500
commit	bd40791e1d289d807b8580abe1f117e9c62894e4 (patch)
tree	2b47e24c8dc0e668dfd7ba0e3879165180c49c65 /include/linux/node.h
parent	98c5584cfc47932c4f3ccf5eee2e0bae1447b85e (diff)

[SPARC64]: Dynamically grow TSB in response to RSS growth.

As the RSS grows, grow the TSB in order to reduce the likelyhood of hash collisions and thus poor hit rates in the TSB. This definitely needs some serious tuning. Signed-off-by: David S. Miller <davem@davemloft.net>

Diffstat (limited to 'include/linux/node.h')

0 files changed, 0 insertions, 0 deletions

/* * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS * * Copyright (c) 2010 Wolfson Microelectronics plc * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * * Based on an earlier driver for the same hardware by Liam Girdwood. * * 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/platform_device.h> #include <linux/clk.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <mach/audmux.h> #include "imx-ssi.h" #include "../codecs/wm8350.h" /* There is a silicon mic on the board optionally connected via a solder pad * SP1. Define this to enable it. */ #undef USE_SIMIC struct _wm8350_audio { unsigned int channels; snd_pcm_format_t format; unsigned int rate; unsigned int sysclk; unsigned int bclkdiv; unsigned int clkdiv; unsigned int lr_rate; }; /* in order of power consumption per rate (lowest first) */ static const struct _wm8350_audio wm8350_audio[] = { /* 16bit mono modes */ {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1, WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,}, /* 16 bit stereo modes */ {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000, WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000, WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000, WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000, WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000, WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600, WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600, WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600, WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200, WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, /* 24bit stereo modes */ {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000, WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000, WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600, WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200, WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, }; static int wm1133_ev1_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->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int i, found = 0; snd_pcm_format_t format = params_format(params); unsigned int rate = params_rate(params); unsigned int channels = params_channels(params); u32 dai_format; /* find the correct audio parameters */ for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { if (rate == wm8350_audio[i].rate && format == wm8350_audio[i].format && channels == wm8350_audio[i].channels) { found = 1; break; } } if (!found) return -EINVAL; /* codec FLL input is 14.75 MHz from MCLK */ snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; /* set codec DAI configuration */ snd_soc_dai_set_fmt(codec_dai, dai_format); /* set cpu DAI configuration */ snd_soc_dai_set_fmt(cpu_dai, dai_format); /* TODO: The SSI driver should figure this out for us */ switch (channels) { case 2: snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); break; case 1: snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0); break; default: return -EINVAL; } /* set MCLK as the codec system clock for DAC and ADC */ snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK, wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN); /* set codec BCLK division for sample rate */ snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV, wm8350_audio[i].bclkdiv); /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */ snd_soc_dai_set_clkdiv(codec_dai, WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate); snd_soc_dai_set_clkdiv(codec_dai, WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate); /* now configure DAC and ADC clocks */ snd_soc_dai_set_clkdiv(codec_dai, WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv); snd_soc_dai_set_clkdiv(codec_dai, WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv); return 0; } static struct snd_soc_ops wm1133_ev1_ops = { .hw_params = wm1133_ev1_hw_params, }; static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = { #ifdef USE_SIMIC SND_SOC_DAPM_MIC("SiMIC", NULL), #endif SND_SOC_DAPM_MIC("Mic1 Jack", NULL), SND_SOC_DAPM_MIC("Mic2 Jack", NULL), SND_SOC_DAPM_LINE("Line In Jack", NULL), SND_SOC_DAPM_LINE("Line Out Jack", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), }; /* imx32ads soc_card audio map */ static const struct snd_soc_dapm_route wm1133_ev1_map[] = { #ifdef USE_SIMIC /* SiMIC --> IN1LN (with automatic bias) via SP1 */ { "IN1LN", NULL, "Mic Bias" }, { "Mic Bias", NULL, "SiMIC" }, #endif /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */ { "IN1LN", NULL, "Mic Bias" }, { "IN1LP", NULL, "Mic1 Jack" }, { "Mic Bias", NULL, "Mic1 Jack" }, /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */ { "IN1RN", NULL, "Mic Bias" }, { "IN1RP", NULL, "Mic2 Jack" }, { "Mic Bias", NULL, "Mic2 Jack" }, /* Line in Jack --> AUX (L+R) */ { "IN3R", NULL, "Line In Jack" }, { "IN3L", NULL, "Line In Jack" }, /* Out1 --> Headphone Jack */ { "Headphone Jack", NULL, "OUT1R" }, { "Headphone Jack", NULL, "OUT1L" }, /* Out1 --> Line Out Jack */ { "Line Out Jack", NULL, "OUT2R" }, { "Line Out Jack", NULL, "OUT2L" }, }; static struct snd_soc_jack hp_jack; static struct snd_soc_jack_pin hp_jack_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE }, }; static struct snd_soc_jack mic_jack; static struct snd_soc_jack_pin mic_jack_pins[] = { { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE }, { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE }, }; static int wm1133_ev1_init(struct snd_soc_codec *codec) { struct snd_soc_card *card = codec->socdev->card; snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets, ARRAY_SIZE(wm1133_ev1_widgets)); snd_soc_dapm_add_routes(codec, wm1133_ev1_map, ARRAY_SIZE(wm1133_ev1_map)); /* Headphone jack detection */ snd_soc_jack_new(card, "Headphone", SND_JACK_HEADPHONE, &hp_jack); snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), hp_jack_pins); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ snd_soc_jack_new(card, "Microphone", SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), mic_jack_pins); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); snd_soc_dapm_force_enable_pin(codec, "Mic Bias"); return 0; } static struct snd_soc_dai_link wm1133_ev1_dai = { .name = "WM1133-EV1", .stream_name = "Audio", .cpu_dai = &imx_ssi_pcm_dai[0], .codec_dai = &wm8350_dai, .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, .symmetric_rates = 1, }; static struct snd_soc_card wm1133_ev1 = { .name = "WM1133-EV1", .platform = &imx_soc_platform, .dai_link = &wm1133_ev1_dai, .num_links = 1, }; static struct snd_soc_device wm1133_ev1_snd_devdata = { .card = &wm1133_ev1, .codec_dev = &soc_codec_dev_wm8350, }; static struct platform_device *wm1133_ev1_snd_device; static int __init wm1133_ev1_audio_init(void) { int ret; unsigned int ptcr, pdcr; /* SSI0 mastered by port 5 */ ptcr = MXC_AUDMUX_V2_PTCR_SYN | MXC_AUDMUX_V2_PTCR_TFSDIR | MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | MXC_AUDMUX_V2_PTCR_TCLKDIR | MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); ptcr = MXC_AUDMUX_V2_PTCR_SYN; pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); if (!wm1133_ev1_snd_device) return -ENOMEM; platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1_snd_devdata); wm1133_ev1_snd_devdata.dev = &wm1133_ev1_snd_device->dev; ret = platform_device_add(wm1133_ev1_snd_device); if (ret) platform_device_put(wm1133_ev1_snd_device); return ret; } module_init(wm1133_ev1_audio_init); static void __exit wm1133_ev1_audio_exit(void) { platform_device_unregister(wm1133_ev1_snd_device); } module_exit(wm1133_ev1_audio_exit); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS"); MODULE_LICENSE("GPL");


context:
space:
mode: