From 79dfc9687661c13ef95eb4c2226f3db4ccab52c9 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:08 -0400 Subject: ASoC: Blackfin AC97: add a few missing multichannel define handling Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b9017a7f..e69322978739 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- cgit v1.2.2 From d75150d7c49db42021b8f966d2cbdc215a530208 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Wed, 16 Sep 2009 20:25:09 -0400 Subject: ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-sport.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 469ce7fab20c..99051ff0954e 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) int sport_tx_start(struct sport_device *sport) { - unsigned flags; + unsigned long flags; pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, sport->tx_run, sport->rx_run); if (sport->tx_run) -- cgit v1.2.2 From 7d156a25bd3e8e6ff74faf02faecb5fc5fb4839e Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:10 -0400 Subject: ASoC: fix typos in Blackfin headers Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-i2s.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f2a911fe0cb..a1f97dd809d6 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-ac97.h + * sound/soc/blackfin/bf5xx-ac97.h * * 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 diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h index 7107d1a0b06b..264ecdcba35a 100644 --- a/sound/soc/blackfin/bf5xx-i2s.h +++ b/sound/soc/blackfin/bf5xx-i2s.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-i2s.h * * 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 -- cgit v1.2.2 From fab19bae0c2951ed8bc517a53848b027fead293d Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:11 -0400 Subject: ASoC: Blackfin I2S: add lost platform_device parameter to resume function Commit dc7d7b830ee1 trimmed the platform_device parameter from all of the suspend functions, but it also accidentally removed it from the resume function in the Blackfin I2S driver. So restore it. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abade27e1..19539c68c053 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } -static void bf5xx_i2s_remove(struct snd_soc_dai *dai) +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); peripheral_free_list(&sport_req[sport_num][0]); -- cgit v1.2.2 From ad80efc469f56d41f3f4adc1b2c86bf65689ebeb Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:12 -0400 Subject: ASoC: Blackfin I2S: fix resuming when device hasn't been used If the sound system hasn't been utilized yet and we suspend, then we attempt to save/restore using state that doesn't exist. So use a global handle instead to reconfigure properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 19539c68c053..1e9d161c76c4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -237,36 +237,31 @@ static void bf5xx_i2s_remove(struct platform_device *pdev, #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; + if (dai->capture.active) - sport_rx_stop(sport); + sport_rx_stop(sport_handle); if (dai->playback.active) - sport_tx_stop(sport); + sport_tx_stop(sport_handle); return 0; } static int bf5xx_i2s_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; -- cgit v1.2.2 From 0c31cf3e4af79ea18bbd365b07ef0de207673894 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Tue, 15 Sep 2009 18:13:29 -0400 Subject: ASoC: DaVinci: Fixes to McASP configuration McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7829d2..7a06c0a86665 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -512,34 +512,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +565,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, RXSSZ(fmt), RXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + return 0; } @@ -638,7 +660,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +676,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- cgit v1.2.2 From 98235a4bb01f288c82f7b878996e507f6159fd69 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Mon, 21 Sep 2009 11:20:51 +0800 Subject: ASoC: some minor changes for AD1836 and AD1938 codec drivers 1. delete redundant assignment to bus field in spi_driver structure 2. fix lost assignment to set_bias_level entry in ad1938 codec dai 3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836" Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 01343dc984fd..c48485f2c55d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 9a049a1995a3..34b30efc3cb0 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- cgit v1.2.2 From 877ae70763fa0b19ffbe25e7e119fc96f52fbbd3 Mon Sep 17 00:00:00 2001 From: Phil Vandry Date: Mon, 21 Sep 2009 11:36:08 -0400 Subject: ASoC: wm8753: fix mapping when MONOMIX is set to Stereo When MONOMIX is set to Stereo, Left PGA was not powered on but should be. Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue. Signed-off-by: Phil Vandry Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414cfbbd..5ad677ce80da 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"}, -- cgit v1.2.2 From df0fd5e5e117329436fdea568455545ca18a71f0 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 23 Sep 2009 11:51:05 -0400 Subject: ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Signed-off-by: Cliff Cai Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 8 ++++---- sound/soc/blackfin/bf5xx-tdm.c | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 1e9d161c76c4..084b68884ada 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = { * TFS. When Port G is selected and EMAC then there is a conflict between * the PHY interrupt line and TFS. Current settings prevent the conflict * by ignoring the TFS pin when Port G is selected. This allows both - * ssm2602 using Port G and EMAC concurrently. + * codecs and EMAC using Port G concurrently. */ -#ifdef CONFIG_BF527_SPORT0_PORTF -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#else +#ifdef CONFIG_BF527_SPORT0_PORTG #define LOCAL_SPORT0_TFS (0) +#else +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) #endif static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096badf09a5..ff546e91a22e 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = { * TFS. When Port G is selected and EMAC then there is a conflict between * the PHY interrupt line and TFS. Current settings prevent the conflict * by ignoring the TFS pin when Port G is selected. This allows both - * ssm2602 using Port G and EMAC concurrently. + * codecs and EMAC using Port G concurrently. */ -#ifdef CONFIG_BF527_SPORT0_PORTF -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#else +#ifdef CONFIG_BF527_SPORT0_PORTG #define LOCAL_SPORT0_TFS (0) +#else +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) #endif static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, -- cgit v1.2.2 From 81ac55aa14c863821248d9e82694c79bb556694d Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Fri, 11 Sep 2009 14:29:02 -0700 Subject: ASoC: DaVinci: Fix divide by zero error during 1st execution When both playback and capture stream were open davinci_i2s_hw_params was setting parameters for the wrong stream. The fix for davinci_i2s_hw_params is sufficient, but it looks like a race still happens in davici_pcm_open. This patch also makes the race smaller but the next patch provides a better fix. Signed-off-by: Troy Kisky Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 12 +++++------- 2 files changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c549ee6e..d32e1974fdf2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,8 +353,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 091dacb78b4d..002808b27f49 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; struct edmacc_param p_ram; int ret; - if (!dma_data) - return -ENODEV; - - prtd->params = dma_data; - /* Request master DMA channel */ ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, @@ -244,6 +237,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *params = rtd->dai->cpu_dai->dma_data; + if (!params) + return -ENODEV; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ @@ -257,6 +254,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = params; runtime->private_data = prtd; -- cgit v1.2.2 From 92e2a6f68219f8d4c862b1f29c653b05639e4c06 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Fri, 11 Sep 2009 14:29:03 -0700 Subject: ASoC: Davinci: Fix race with cpu_dai->dma_data This patch removes references to cpu_dai->dma_data. It makes struct davinci_pcm_dma_params part of struct davinci_mcbsp_dev or struct davinci_audio_dev. It removes the unused name variable from davinci_pcm_dma_params. Signed-off-by: Troy Kisky Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 36 ++++++++++---------------------- sound/soc/davinci/davinci-mcasp.c | 44 +++++++++------------------------------ sound/soc/davinci/davinci-mcasp.h | 7 ++++++- sound/soc/davinci/davinci-pcm.c | 3 ++- sound/soc/davinci/davinci-pcm.h | 1 - 5 files changed, 29 insertions(+), 62 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d32e1974fdf2..4ae707048021 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,22 +97,19 @@ enum { DAVINCI_MCBSP_WORD_32, }; -static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { - .name = "I2S PCM Stereo out", -}; - -static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { - .name = "I2S PCM Stereo in", -}; - struct davinci_mcbsp_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to a pointer of an array of struct davinci_pcm_dma_params in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 int mode; u32 pcr; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -355,7 +344,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, { struct davinci_mcbsp_dev *dev = dai->private_data; struct davinci_pcm_dma_params *dma_params = - dev->dma_params[substream->stream]; + &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; @@ -473,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -535,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->base = (void __iomem *)IO_ADDRESS(mem->start); - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -550,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_free_mem; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -558,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_free_mem; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; ret = snd_soc_register_dai(&davinci_i2s_dai); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7a06c0a86665..3174d96d929f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } -static int davinci_mcasp_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_audio_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - static void mcasp_start_rx(struct davinci_audio_dev *dev) { mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); @@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, { struct davinci_audio_dev *dev = cpu_dai->private_data; struct davinci_pcm_dma_params *dma_params = - dev->dma_params[substream->stream]; + &dev->dma_params[substream->stream]; int word_length; u8 numevt; @@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { - .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, @@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res; struct snd_platform_data *pdata; struct davinci_audio_dev *dev; - int count = 0; int ret = 0; dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); if (!dev) return -ENOMEM; - dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2, - GFP_KERNEL); - if (!dma_data) { - ret = -ENOMEM; - goto err_release_dev; - } - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "no mem resource?\n"); @@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; - dma_data[count].name = "I2S PCM Stereo out"; - dma_data[count].eventq_no = pdata->eventq_no; - dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->eventq_no = pdata->eventq_no; + dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count]; /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err_release_region; } - dma_data[count].channel = res->start; - count++; - dma_data[count].name = "I2S PCM Stereo in"; - dma_data[count].eventq_no = pdata->eventq_no; - dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + dma_data->channel = res->start; + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->eventq_no = pdata->eventq_no; + dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err_release_region; } - dma_data[count].channel = res->start; + dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); @@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) err_release_region: release_mem_region(mem->start, (mem->end - mem->start) + 1); err_release_data: - kfree(dma_data); -err_release_dev: kfree(dev); return ret; @@ -946,7 +925,6 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { struct snd_platform_data *pdata = pdev->dev.platform_data; - struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; @@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev) mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); - dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - kfree(dma_data); kfree(dev); return 0; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 554354c1cc2f..9d179cc88f7b 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,10 +39,15 @@ enum { }; struct davinci_audio_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to a pointer of an array of struct davinci_pcm_dma_params in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; unsigned int codec_fmt; /* McASP specific data */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 002808b27f49..359e99ec7244 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,7 +238,8 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *params = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *params = &pa[substream->stream]; if (!params) return -ENODEV; diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 63d96253c73a..8746606efc89 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -17,7 +17,6 @@ struct davinci_pcm_dma_params { - char *name; /* stream identifier */ int channel; /* sync dma channel ID */ unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ -- cgit v1.2.2 From 539d3d8cbe5cf7597d4c4c4428aec242f9ea5185 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Wed, 23 Sep 2009 10:12:08 -0400 Subject: ASoC: DaVinci: Correct McASP FIFO initialization McASP write FIFO registers should be modified for playback and read FIFO registers for capture. Check the PCM mode before manipulating the FIFO registers. Currently, irrespective of playback/capture both the FIFOs are enabled or disbaled. This resulted in errors in audio loopback mode. Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 3174d96d929f..5d1f98a4c978 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -378,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_start_tx(dev); - else + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_start_rx(dev); - - /* enable FIFO */ - if (dev->txnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -405,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_stop_tx(dev); - else + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_stop_rx(dev); - - /* disable FIFO */ - if (dev->txnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, -- cgit v1.2.2 From c877c25170e2655d519b29e91d6c91d5d1a72a6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2009 08:33:47 +0200 Subject: ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 wm8940 requires I2C. Signed-off-by: Takashi Iwai --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 6375b4ea525d..dcb3181bb340 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701 config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help -- cgit v1.2.2 From df1246d84ab7edc375e6b6d236654ac0866229c5 Mon Sep 17 00:00:00 2001 From: Barry Song Date: Thu, 1 Oct 2009 01:33:30 -0400 Subject: ASoC: fix kconfig order of Blackfin drivers Some of the Blackfin options don't directly follow the kconfig options they depend on, so kconfig is unable to display the proper tree. So sort the options such they expand/collapse properly. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 98 +++++++++++++++++++++++----------------------- 1 file changed, 49 insertions(+), 49 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index ac927ffdc961..97f1a251e446 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,15 +7,6 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + +config SND_BF5XX_SOC_AD1938 + tristate "SoC AD1938 Audio support for Blackfin" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1938 + help + Say Y if you want to add support for AD1938 codec on Blackfin. + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT Say y if you want AC97 driver to support up to 5.1 channel audio. this mode will consume much more memory for DMA. +config SND_BF5XX_HAVE_COLD_RESET + bool "BOARD has COLD Reset GPIO" + depends on SND_BF5XX_AC97 + default y if BFIN548_EZKIT + default n if !BFIN548_EZKIT + +config SND_BF5XX_RESET_GPIO_NUM + int "Set a GPIO for cold reset" + depends on SND_BF5XX_HAVE_COLD_RESET + range 0 159 + default 19 if BFIN548_EZKIT + default 5 if BFIN537_STAMP + default 0 + help + Set the correct GPIO for RESET the sound chip. + +config SND_BF5XX_SOC_AD1980 + tristate "SoC AD1980/1 Audio support for BF5xx" + depends on SND_BF5XX_AC97 + select SND_BF5XX_SOC_AC97 + select SND_SOC_AD1980 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_SPORT tristate @@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx" - depends on SND_BF5XX_AC97 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 - help - Say Y if you want to add support for AD1938 codec on Blackfin. - config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) @@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM default 0 help Set the correct SPORT for sound chip. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. -- cgit v1.2.2