August 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: mxs-saif: add record function
by Dong Aisheng 24 Aug '11

24 Aug '11

1. add different clkmux mode handling for record. SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode that is using other saif's BITCLK&LRCLK. The clkmux mode is determined by saif's platform data and machine specific clkmux setting is done in pdata->init(). 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> --- include/sound/saif.h | 16 +++++ sound/soc/mxs/mxs-saif.c | 162 ++++++++++++++++++++++++++++++++++++++++++---- sound/soc/mxs/mxs-saif.h | 4 + 3 files changed, 168 insertions(+), 14 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 0000000..7a98d9b --- /dev/null +++ b/include/sound/saif.h @@ -0,0 +1,16 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __MACH_MXS_SAIF_H__ +#define __MACH_MXS_SAIF_H__ + +struct mxs_saif_platform_data { + unsigned int clkmux; + int (*init) (void); +}; +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 530017f..eb8ba79 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -27,15 +27,39 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/saif.h> #include <mach/dma.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> +#include <mach/digctl.h> #include "mxs-saif.h" static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * Following are the valid configurations for SAIF1 and SAIF2 on the i.MX28 + * + * HW_SAIF_CLKMUX_SEL: + * DIRECT(0x0): SAIF0 clock pins selected for SAIF0 input clocks, and SAIF1 + * clock pins selected for SAIF1 input clocks. + * CROSSINPUT(0x1): SAIF1 clock inputs selected for SAIF0 input clocks, and + * SAIF0 clock inputs selected for SAIF1 input clocks. + * EXTMSTR0(0x2): SAIF0 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + * EXTMSTR1(0x3): SAIF1 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -52,6 +76,36 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master. + * Note that for DIRECT mode, it's master is itself. + */ +static struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + struct mxs_saif *master_saif; + + /* get master saif according to different clkmux setting */ + switch (saif->clkmux) { + case MXS_DIGCTL_SAIF_CLKMUX_DIRECT: + master_saif = saif; + break; + case MXS_DIGCTL_SAIF_CLKMUX_CROSSINPUT: + master_saif = saif->id ? mxs_saif[1] : mxs_saif[0]; + break; + case MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0: + master_saif = mxs_saif[0]; + break; + case MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1: + master_saif = mxs_saif[1]; + break; + default: + return NULL; + } + + return master_saif; +} + +/* * Set SAIF clock and MCLK */ static int mxs_saif_set_clk(struct mxs_saif *saif, @@ -60,8 +114,27 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d clkmux 0x%x\n", + master_saif->id, saif->clkmux); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +148,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +212,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +258,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +271,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +360,21 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->clkmux == MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0) + scr = saif->id ? scr | BM_SAIF_CTRL_SLAVE_MODE : + scr & ~BM_SAIF_CTRL_SLAVE_MODE; + else if (saif->clkmux == MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1) + scr = saif->id ? scr & ~BM_SAIF_CTRL_SLAVE_MODE : + scr | BM_SAIF_CTRL_SLAVE_MODE; + else + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +489,11 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +501,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +530,37 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(100); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + /* wait a while for the current sample to complete */ + udelay(100); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,15 +644,24 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init()) + return -EINVAL; + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; + if (pdata) + saif->clkmux = pdata->clkmux; + if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8c..2a6aec1 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int clkmux; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- 1.7.0.4

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[alsa-devel] [PATCH 1/1] ASoC: soc-core: check rate for symmetry only when pcm is ongoing
by Dong Aisheng 24 Aug '11

24 Aug '11

For the playback and record using different dai links, checking !rtd->rate for symmetry may not be accurate because that pcm may be acutually not running and the default new open rate is 0, then the warning message "Not enforcing symmetric_rates" will happen each time with running arecord | aplay. Now we only check rate for symmetry when the pcm is really ongoing which seems more sensible. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..8e9931c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -851,6 +851,7 @@ struct snd_soc_pcm_runtime { /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; + unsigned int ongoing; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..f3c1a66 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -39,6 +39,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) !rtd->dai_link->symmetric_rates) return 0; + if (!rtd->ongoing) + return 0; + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. @@ -209,6 +212,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; + rtd->ongoing = 1; mutex_unlock(&rtd->pcm_mutex); return 0; @@ -322,6 +326,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } + rtd->ongoing = 0; mutex_unlock(&rtd->pcm_mutex); return 0; } -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Kconfig to support SMDK4212
by Sangbeom Kim 24 Aug '11

24 Aug '11

This patch adds Kconfig to support SMDK4212. SMDK4212 is based on samsung exynos4212 SoC. And WM8994 is used for audio codec. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b99091f..9a3138b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -173,7 +173,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_PCM help -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Tegra: wm8903 machine driver: Drop Ventana support
by Stephen Warren 24 Aug '11

24 Aug '11

Board file support for Ventana is not yet mainlined, and probably won't ever be given the move to Device-Tree. Consequently, the Ventana entry is being removed from arch/arm/tools/mach-types in the next merge window, since it was registered over a year ago. This will also remove function machine_is_ventana(), which is used by the ASoC Tegra WM8903 machine driver. This will cause compilation failures. Drop Ventana support to resolve this. Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to configure itself from Device-Tree, and hence we'll be able to re-instate Ventana support just by creating a .dts file for the board. Also note that Aebl support is in a similar boat. However, that board isn't scheduled for deprecation for at least another 5 months, and perhaps we will have completely removed non-Device-Tree support from tegra_wm8903.c by then and/or adjusted mach-types policy. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_wm8903.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 661373c..be27f1d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony() && !machine_is_ventana()) + if (!machine_is_harmony()) snd_soc_dapm_nc_pin(dapm, "IN1L"); if (!machine_is_seaboard() && !machine_is_aebl()) snd_soc_dapm_nc_pin(dapm, "IN1R"); @@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony() || machine_is_ventana()) { + if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } else if (machine_is_seaboard()) { -- 1.7.0.4

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Re: [alsa-devel] [linux-next] ASoC: build failure if CONFIG_I2C=m && CONFIG_SPI=y && CONFIG_SND_SOC=y
by Randy Dunlap 24 Aug '11

24 Aug '11

On Tue, 23 Aug 2011 19:46:00 +0800 Axel Lin wrote: > > hi, > I got below build failure if CONFIG_I2C=m && CONFIG_SPI=y && CONFIG_SND_SOC=y > > LD .tmp_vmlinux1 > sound/built-in.o: In function `snd_soc_codec_set_cache_io': > last.c:(.text+0x1cb80): undefined reference to `regmap_init_i2c' > make: *** [.tmp_vmlinux1] Error 1 I don't know if this is the same failure or just similar: ERROR: "regmap_exit" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_bulk_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_init_i2c" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_raw_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_write" [drivers/mfd/wm8400-core.ko] undefined! Maybe MFD_WM8400 needs to select REGMAP_I2C ? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] [PATCH] ASoC: Add samsung maintainer
by Sangbeom Kim 24 Aug '11

24 Aug '11

Cc: Jassi Brar <jassisinghbrar(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- MAINTAINERS | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 069ee3b..fadff23 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5532,6 +5532,7 @@ F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS M: Jassi Brar <jassisinghbrar(a)gmail.com> +M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung -- 1.7.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add Springbank I/O card to Speyside Kconfig
by Mark Brown 24 Aug '11

24 Aug '11

On Tue, Aug 23, 2011 at 10:41:50AM +0100, Joseph Pentland wrote: > Signed-off-by: Joseph Pentland <jp(a)opensource.wolfsonmicro.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] ALSA: ctxfi: Bump playback substreams to 256
by Takashi Iwai 24 Aug '11

24 Aug '11

At Wed, 24 Aug 2011 00:48:59 +0200, Maarten Lankhorst wrote: > > There are references in the code to 256 sources, so I tested it with 256 aplays, > of which the first and last with real data and the rest playing /dev/zero . > > Also increase amount of page tables, so the default aplay size works. > > Signed-off-by: Maarten Lankhorst <m.b.lankhorst(a)gmail.com> Applied now. Thanks. Takashi > --- > diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c > index 457d211..2c86226 100644 > --- a/sound/pci/ctxfi/ctpcm.c > +++ b/sound/pci/ctxfi/ctpcm.c > @@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, > int err; > int playback_count, capture_count; > > - playback_count = (IEC958 == device) ? 1 : 8; > + playback_count = (IEC958 == device) ? 1 : 256; > capture_count = (FRONT == device) ? 1 : 0; > err = snd_pcm_new(atc->card, "ctxfi", device, > playback_count, capture_count, &pcm); > diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c > index c749fa7..e134b3a 100644 > --- a/sound/pci/ctxfi/ctsrc.c > +++ b/sound/pci/ctxfi/ctsrc.c > @@ -20,7 +20,7 @@ > #include "cthardware.h" > #include <linux/slab.h> > > -#define SRC_RESOURCE_NUM 64 > +#define SRC_RESOURCE_NUM 256 > #define SRCIMP_RESOURCE_NUM 256 > > static unsigned int conj_mask; > diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h > index b23adfc..e6da60e 100644 > --- a/sound/pci/ctxfi/ctvmem.h > +++ b/sound/pci/ctxfi/ctvmem.h > @@ -18,7 +18,7 @@ > #ifndef CTVMEM_H > #define CTVMEM_H > > -#define CT_PTP_NUM 1 /* num of device page table pages */ > +#define CT_PTP_NUM 4 /* num of device page table pages */ > > #include <linux/mutex.h> > #include <linux/list.h> >

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Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 24 Aug '11

24 Aug '11

On Wed, Aug 24, 2011 at 7:30 AM, Chen Peter-B29397 <B29397(a)freescale.com> wrote: >> >> This is a flexible USB Audio Class 2.0 compliant gadget driver that >> implements a simple topology with a virtual sound card exposed at >> the function side. >> >> The driver doesn't expect any real audio codec to be present on the >> function - the audio streams are simply sinked to and sourced from a >> virtual ALSA sound card created. The user-space application may choose >> to do whatever it wants with the data received from the USB Host and >> choose to provide whatever it wants as audio data to the USB Host. >> >> Capture(USB-Out) and Playback(USB-In) can be run at independent >> configurations specified via module parameters while loading the driver. >> > > Have you a chance to test at Macbook (10.6.5+) which has USB Audio 2.0 support well? > Nopes, haven't had my hands on one yet. It would help if someone could verify. ..... >> +/* STD AS ISO OUT Endpoint */ >> +struct usb_endpoint_descriptor fs_epout_desc = { >> + .bLength = USB_DT_ENDPOINT_SIZE, >> + .bDescriptorType = USB_DT_ENDPOINT, >> + >> + .bEndpointAddress = USB_DIR_OUT, >> + .bmAttributes = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, >> + .bInterval = 1, >> +}; >> + >> +struct usb_endpoint_descriptor hs_epout_desc = { >> + .bLength = USB_DT_ENDPOINT_SIZE, >> + .bDescriptorType = USB_DT_ENDPOINT, >> + >> + .bmAttributes = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, >> + .bInterval = 4, >> +}; > The sync type is asynchronous, do you have plan to add feedback for handling > overflow and underflow? > umm... I was not sure if the feature is worth the overhead/complexity of maintaining timers in the driver. Though I welcome hearing why not so?

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add Springbank I/O card to Speyside Kconfig
by Liam Girdwood 23 Aug '11

23 Aug '11

On 23/08/11 10:41, Joseph Pentland wrote: > Signed-off-by: Joseph Pentland <jp(a)opensource.wolfsonmicro.com> > --- > sound/soc/samsung/Kconfig | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig > index b99091f..65f980e 100644 > --- a/sound/soc/samsung/Kconfig > +++ b/sound/soc/samsung/Kconfig > @@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE > select SND_SAMSUNG_I2S > select SND_SOC_WM8996 > select SND_SOC_WM9081 > + select SND_SOC_WM1250_EV1 > > config SND_SOC_SPEYSIDE_WM8962 > tristate "Audio support for Wolfson Speyside with WM8962" Acked-by: Liam Girdwood <lrg(a)ti.com> Missing 2/2 ?

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