From 4c3f9d5fcb46d769f4a52a044fead863419c1d58 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 18 Aug 2010 00:25:12 +0100 Subject: [PATCH 1/3] ASoC: core - fix build warning on x86_64 Output size_t type as a "%Zu" to avoid warnings. Signed-off-by: Liam Girdwood --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3d480eb3555f..7093c1787128 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2916,7 +2916,7 @@ int snd_soc_register_dais(struct device *dev, struct snd_soc_dai *dai; int i, ret = 0; - dev_dbg(dev, "dai register %s #%d\n", dev_name(dev), count); + dev_dbg(dev, "dai register %s #%Zu\n", dev_name(dev), count); for (i = 0; i < count; i++) { From 8e9d869028f3ce13631af5ef41910ad8d8e6c246 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 6 Aug 2010 12:16:12 -0500 Subject: [PATCH 2/3] asoc/multi-component: fsl: add support for variable SSI FIFO depth Add code that programs the DMA and SSI controllers differently based on the FIFO depth of the SSI. The SSI devices on the MPC8610 and the P1022 are identical in every way except one: the transmit and receive FIFO depth. On the MPC8610, the depth is eight. On the P1022, it's fifteen. The device tree nodes for the SSI include a "fsl,fifo-depth" property that specifies the FIFO depth. Signed-off-by: Timur Tabi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/fsl/fsl_dma.c | 67 +++++++++++++++++++++++++++++++---------- sound/soc/fsl/fsl_ssi.c | 25 +++++++++++++-- 2 files changed, 73 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 57774cb91ae3..dfe1cb94a70f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -60,6 +60,7 @@ struct dma_object { struct snd_soc_platform_driver dai; dma_addr_t ssi_stx_phys; dma_addr_t ssi_srx_phys; + unsigned int ssi_fifo_depth; struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; @@ -99,6 +100,7 @@ struct fsl_dma_private { unsigned int irq; struct snd_pcm_substream *substream; dma_addr_t ssi_sxx_phys; + unsigned int ssi_fifo_depth; dma_addr_t ld_buf_phys; unsigned int current_link; dma_addr_t dma_buf_phys; @@ -431,6 +433,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) else dma_private->ssi_sxx_phys = dma->ssi_srx_phys; + dma_private->ssi_fifo_depth = dma->ssi_fifo_depth; dma_private->dma_channel = dma->channel; dma_private->irq = dma->irq; dma_private->substream = substream; @@ -544,11 +547,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, struct device *dev = rtd->platform->dev; /* Number of bits per sample */ - unsigned int sample_size = + unsigned int sample_bits = snd_pcm_format_physical_width(params_format(hw_params)); /* Number of bytes per frame */ - unsigned int frame_size = 2 * (sample_size / 8); + unsigned int sample_bytes = sample_bits / 8; /* Bus address of SSI STX register */ dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; @@ -588,7 +591,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * that offset here. While we're at it, also tell the DMA controller * how much data to transfer per sample. */ - switch (sample_size) { + switch (sample_bits) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -602,22 +605,42 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, break; default: /* We should never get here */ - dev_err(dev, "unsupported sample size %u\n", sample_size); + dev_err(dev, "unsupported sample size %u\n", sample_bits); return -EINVAL; } /* - * BWC should always be a multiple of the frame size. BWC determines - * how many bytes are sent/received before the DMA controller checks the - * SSI to see if it needs to stop. For playback, the transmit FIFO can - * hold three frames, so we want to send two frames at a time. For - * capture, the receive FIFO is triggered when it contains one frame, so - * we want to receive one frame at a time. + * BWC determines how many bytes are sent/received before the DMA + * controller checks the SSI to see if it needs to stop. BWC should + * always be a multiple of the frame size, so that we always transmit + * whole frames. Each frame occupies two slots in the FIFO. The + * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two + * (MR[BWC] can only represent even powers of two). + * + * To simplify the process, we set BWC to the largest value that is + * less than or equal to the FIFO watermark. For playback, this ensures + * that we transfer the maximum amount without overrunning the FIFO. + * For capture, this ensures that we transfer the maximum amount without + * underrunning the FIFO. + * + * f = SSI FIFO depth + * w = SSI watermark value (which equals f - 2) + * b = DMA bandwidth count (in bytes) + * s = sample size (in bytes, which equals frame_size * 2) + * + * For playback, we never transmit more than the transmit FIFO + * watermark, otherwise we might write more data than the FIFO can hold. + * The watermark is equal to the FIFO depth minus two. + * + * For capture, two equations must hold: + * w > f - (b / s) + * w >= b / s + * + * So, b > 2 * s, but b must also be <= s * w. To simplify, we set + * b = s * w, which is equal to + * (dma_private->ssi_fifo_depth - 2) * sample_bytes. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - mr |= CCSR_DMA_MR_BWC(2 * frame_size); - else - mr |= CCSR_DMA_MR_BWC(frame_size); + mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes); out_be32(&dma_channel->mr, mr); @@ -871,6 +894,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, struct device_node *np = of_dev->dev.of_node; struct device_node *ssi_np; struct resource res; + const uint32_t *iprop; int ret; /* Find the SSI node that points to us. */ @@ -881,15 +905,17 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, } ret = of_address_to_resource(ssi_np, 0, &res); - of_node_put(ssi_np); if (ret) { - dev_err(&of_dev->dev, "could not determine device resources\n"); + dev_err(&of_dev->dev, "could not determine resources for %s\n", + ssi_np->full_name); + of_node_put(ssi_np); return ret; } dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { dev_err(&of_dev->dev, "could not allocate dma object\n"); + of_node_put(ssi_np); return -ENOMEM; } @@ -902,6 +928,15 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); + if (iprop) + dma->ssi_fifo_depth = *iprop; + else + /* Older 8610 DTs didn't have the fifo-depth property */ + dma->ssi_fifo_depth = 8; + + of_node_put(ssi_np); + ret = snd_soc_register_platform(&of_dev->dev, &dma->dai); if (ret) { dev_err(&of_dev->dev, "could not register platform\n"); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7939c337ed9d..d1c855ade8fb 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -93,6 +93,7 @@ struct fsl_ssi_private { unsigned int playback; unsigned int capture; int asynchronous; + unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; struct platform_device *pdev; @@ -337,11 +338,20 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We - * don't use FIFO 1. Since the SSI only supports stereo, the - * watermark should never be an odd number. + * don't use FIFO 1. We program the transmit water to signal a + * DMA transfer if there are only two (or fewer) elements left + * in the FIFO. Two elements equals one frame (left channel, + * right channel). This value, however, depends on the depth of + * the transmit buffer. + * + * We program the receive FIFO to notify us if at least two + * elements (one frame) have been written to the FIFO. We could + * make this value larger (and maybe we should), but this way + * data will be written to memory as soon as it's available. */ out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | + CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); /* * We keep the SSI disabled because if we enable it, then the @@ -622,6 +632,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, struct device_attribute *dev_attr = NULL; struct device_node *np = of_dev->dev.of_node; const char *p, *sprop; + const uint32_t *iprop; struct resource res; char name[64]; @@ -678,6 +689,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, else ssi_private->cpu_dai_drv.symmetric_rates = 1; + /* Determine the FIFO depth. */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + ssi_private->fifo_depth = *iprop; + else + /* Older 8610 DTs didn't have the fifo-depth property */ + ssi_private->fifo_depth = 8; + /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; dev_attr->attr.name = "statistics"; From 3fabe089ad8b8f238bc9de3e7586ae8d2a81f57c Mon Sep 17 00:00:00 2001 From: "Matti J. Aaltonen" Date: Fri, 20 Aug 2010 12:32:46 +0300 Subject: [PATCH 3/3] ASoC: TI WL1273 FM Radio Codec. This is an ALSA codec for the Texas Instruments WL1273 FM Radio. Signed-off-by: Matti J. Aaltonen Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/wl1273.c | 525 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 101 ++++++++ 2 files changed, 626 insertions(+) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c new file mode 100644 index 000000000000..0cd590970883 --- /dev/null +++ b/sound/soc/codecs/wl1273.c @@ -0,0 +1,525 @@ +/* + * ALSA SoC WL1273 codec driver + * + * Author: Matti Aaltonen, + * + * Copyright: (C) 2010 Nokia Corporation + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "wl1273.h" + +enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX }; + +/* codec private data */ +struct wl1273_priv { + enum wl1273_mode mode; + struct wl1273_core *core; + unsigned int channels; +}; + +static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, + int rate, int width) +{ + struct device *dev = &core->i2c_dev->dev; + int r = 0; + u16 mode; + + dev_dbg(dev, "rate: %d\n", rate); + dev_dbg(dev, "width: %d\n", width); + + mutex_lock(&core->lock); + + mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE; + + switch (rate) { + case 48000: + mode |= WL1273_IS2_RATE_48K; + break; + case 44100: + mode |= WL1273_IS2_RATE_44_1K; + break; + case 32000: + mode |= WL1273_IS2_RATE_32K; + break; + case 22050: + mode |= WL1273_IS2_RATE_22_05K; + break; + case 16000: + mode |= WL1273_IS2_RATE_16K; + break; + case 12000: + mode |= WL1273_IS2_RATE_12K; + break; + case 11025: + mode |= WL1273_IS2_RATE_11_025; + break; + case 8000: + mode |= WL1273_IS2_RATE_8K; + break; + default: + dev_err(dev, "Sampling rate: %d not supported\n", rate); + r = -EINVAL; + goto out; + } + + switch (width) { + case 16: + mode |= WL1273_IS2_WIDTH_32; + break; + case 20: + mode |= WL1273_IS2_WIDTH_40; + break; + case 24: + mode |= WL1273_IS2_WIDTH_48; + break; + case 25: + mode |= WL1273_IS2_WIDTH_50; + break; + case 30: + mode |= WL1273_IS2_WIDTH_60; + break; + case 32: + mode |= WL1273_IS2_WIDTH_64; + break; + case 40: + mode |= WL1273_IS2_WIDTH_80; + break; + case 48: + mode |= WL1273_IS2_WIDTH_96; + break; + case 64: + mode |= WL1273_IS2_WIDTH_128; + break; + default: + dev_err(dev, "Data width: %d not supported\n", width); + r = -EINVAL; + goto out; + } + + dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE); + dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode); + dev_dbg(dev, "mode: 0x%04x\n", mode); + + if (core->i2s_mode != mode) { + r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode); + if (r) + goto out; + + core->i2s_mode = mode; + r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); + if (r) + goto out; + } +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, + int channel_number) +{ + struct i2c_client *client = core->i2c_dev; + struct device *dev = &client->dev; + int r = 0; + + dev_dbg(dev, "%s\n", __func__); + + mutex_lock(&core->lock); + + if (core->channel_number == channel_number) + goto out; + + if (channel_number == 1 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_MONO); + else if (channel_number == 1 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_MONO); + else if (channel_number == 2 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_STEREO); + else if (channel_number == 2 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_STEREO); + else + r = -EINVAL; +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wl1273->mode; + + return 0; +} + +static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; + +static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + /* Do not allow changes while stream is running */ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route)) + return -EINVAL; + + wl1273->mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const struct soc_enum wl1273_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route); + +static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->audio_mode; + + return 0; +} + +static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int val, r = 0; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + val = ucontrol->value.integer.value[0]; + if (wl1273->core->audio_mode == val) + return 0; + + r = wl1273_fm_set_audio(wl1273->core, val); + if (r < 0) + return r; + + return 1; +} + +static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; + +static const struct soc_enum wl1273_audio_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), + wl1273_audio_strings); + +static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->volume; + + return 0; +} + +static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int r; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + r = wl1273_fm_set_volume(wl1273->core, + ucontrol->value.integer.value[0]); + if (r) + return r; + + return 1; +} + +static const struct snd_kcontrol_new wl1273_controls[] = { + SOC_ENUM_EXT("Codec Mode", wl1273_enum, + snd_wl1273_get_audio_route, snd_wl1273_set_audio_route), + SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum, + snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put), + SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0, + snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), +}; + +static int wl1273_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->mode) { + case WL1273_MODE_BT: + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 8000, 8000); + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1); + break; + case WL1273_MODE_FM_RX: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Cannot play in RX mode.\n"); + return -EINVAL; + } + break; + case WL1273_MODE_FM_TX: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Cannot capture in TX mode.\n"); + return -EINVAL; + } + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +static int wl1273_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_core *core = wl1273->core; + unsigned int rate, width, r; + + if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) { + pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n"); + return -EINVAL; + } + + rate = params_rate(params); + width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + + if (wl1273->mode == WL1273_MODE_BT) { + if (rate != 8000) { + pr_err("Rate %d not supported.\n", params_rate(params)); + return -EINVAL; + } + + if (params_channels(params) != 1) { + pr_err("Only mono supported.\n"); + return -EINVAL; + } + + return 0; + } + + if (wl1273->mode == WL1273_MODE_FM_TX && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Only playback supported with TX.\n"); + return -EINVAL; + } + + if (wl1273->mode == WL1273_MODE_FM_RX && + substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Only capture supported with RX.\n"); + return -EINVAL; + } + + if (wl1273->mode != WL1273_MODE_FM_RX && + wl1273->mode != WL1273_MODE_FM_TX) { + pr_err("Unexpected mode: %d.\n", wl1273->mode); + return -EINVAL; + } + + r = snd_wl1273_fm_set_i2s_mode(core, rate, width); + if (r) + return r; + + wl1273->channels = params_channels(params); + r = snd_wl1273_fm_set_channel_number(core, wl1273->channels); + if (r) + return r; + + return 0; +} + +static struct snd_soc_dai_ops wl1273_dai_ops = { + .startup = wl1273_startup, + .hw_params = wl1273_hw_params, +}; + +static struct snd_soc_dai_driver wl1273_dai = { + .name = "wl1273-fm", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .ops = &wl1273_dai_ops, +}; + +/* Audio interface format for the soc_card driver */ +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt) +{ + struct wl1273_priv *wl1273; + + if (codec == NULL || fmt == NULL) + return -EINVAL; + + wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->mode) { + case WL1273_MODE_FM_RX: + case WL1273_MODE_FM_TX: + *fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + case WL1273_MODE_BT: + *fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + default: + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(wl1273_get_format); + +static int wl1273_probe(struct snd_soc_codec *codec) +{ + struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_priv *wl1273; + int r; + + dev_dbg(codec->dev, "%s.\n", __func__); + + if (!core) { + dev_err(codec->dev, "Platform data is missing.\n"); + return -EINVAL; + } + + wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL); + if (wl1273 == NULL) { + dev_err(codec->dev, "Cannot allocate memory.\n"); + return -ENOMEM; + } + + wl1273->mode = WL1273_MODE_BT; + wl1273->core = *core; + + snd_soc_codec_set_drvdata(codec, wl1273); + mutex_init(&codec->mutex); + + r = snd_soc_add_controls(codec, wl1273_controls, + ARRAY_SIZE(wl1273_controls)); + if (r) + kfree(wl1273); + + return r; +} + +static int wl1273_remove(struct snd_soc_codec *codec) +{ + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s\n", __func__); + kfree(wl1273); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { + .probe = wl1273_probe, + .remove = wl1273_remove, +}; + +static int __devinit wl1273_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273, + &wl1273_dai, 1); +} + +static int __devexit wl1273_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +MODULE_ALIAS("platform:wl1273-codec"); + +static struct platform_driver wl1273_platform_driver = { + .driver = { + .name = "wl1273-codec", + .owner = THIS_MODULE, + }, + .probe = wl1273_platform_probe, + .remove = __devexit_p(wl1273_platform_remove), +}; + +static int __init wl1273_init(void) +{ + return platform_driver_register(&wl1273_platform_driver); +} +module_init(wl1273_init); + +static void __exit wl1273_exit(void) +{ + platform_driver_unregister(&wl1273_platform_driver); +} +module_exit(wl1273_exit); + +MODULE_AUTHOR("Matti Aaltonen "); +MODULE_DESCRIPTION("ASoC WL1273 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h new file mode 100644 index 000000000000..14ed027fdcfc --- /dev/null +++ b/sound/soc/codecs/wl1273.h @@ -0,0 +1,101 @@ +/* + * sound/soc/codec/wl1273.h + * + * ALSA SoC WL1273 codec driver + * + * Copyright (C) Nokia Corporation + * Author: Matti Aaltonen + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __WL1273_CODEC_H__ +#define __WL1273_CODEC_H__ + +/* I2S protocol, left channel first, data width 16 bits */ +#define WL1273_PCM_DEF_MODE 0x00 + +/* Rx */ +#define WL1273_AUDIO_ENABLE_I2S (1 << 0) +#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1) + +/* Tx */ +#define WL1273_AUDIO_IO_SET_ANALOG 0 +#define WL1273_AUDIO_IO_SET_I2S 1 + +#define WL1273_POWER_SET_OFF 0 +#define WL1273_POWER_SET_FM (1 << 0) +#define WL1273_POWER_SET_RDS (1 << 1) +#define WL1273_POWER_SET_RETENTION (1 << 4) + +#define WL1273_PUPD_SET_OFF 0x00 +#define WL1273_PUPD_SET_ON 0x01 +#define WL1273_PUPD_SET_RETENTION 0x10 + +/* I2S mode */ +#define WL1273_IS2_WIDTH_32 0x0 +#define WL1273_IS2_WIDTH_40 0x1 +#define WL1273_IS2_WIDTH_22_23 0x2 +#define WL1273_IS2_WIDTH_23_22 0x3 +#define WL1273_IS2_WIDTH_48 0x4 +#define WL1273_IS2_WIDTH_50 0x5 +#define WL1273_IS2_WIDTH_60 0x6 +#define WL1273_IS2_WIDTH_64 0x7 +#define WL1273_IS2_WIDTH_80 0x8 +#define WL1273_IS2_WIDTH_96 0x9 +#define WL1273_IS2_WIDTH_128 0xa +#define WL1273_IS2_WIDTH 0xf + +#define WL1273_IS2_FORMAT_STD (0x0 << 4) +#define WL1273_IS2_FORMAT_LEFT (0x1 << 4) +#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4) +#define WL1273_IS2_FORMAT_USER (0x3 << 4) + +#define WL1273_IS2_MASTER (0x0 << 6) +#define WL1273_IS2_SLAVEW (0x1 << 6) + +#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7) +#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7) + +#define WL1273_IS2_SDOWS_RR (0x0 << 8) +#define WL1273_IS2_SDOWS_RF (0x1 << 8) +#define WL1273_IS2_SDOWS_FR (0x2 << 8) +#define WL1273_IS2_SDOWS_FF (0x3 << 8) + +#define WL1273_IS2_TRI_OPT (0x0 << 10) +#define WL1273_IS2_TRI_ALWAYS (0x1 << 10) + +#define WL1273_IS2_RATE_48K (0x0 << 12) +#define WL1273_IS2_RATE_44_1K (0x1 << 12) +#define WL1273_IS2_RATE_32K (0x2 << 12) +#define WL1273_IS2_RATE_22_05K (0x4 << 12) +#define WL1273_IS2_RATE_16K (0x5 << 12) +#define WL1273_IS2_RATE_12K (0x8 << 12) +#define WL1273_IS2_RATE_11_025 (0x9 << 12) +#define WL1273_IS2_RATE_8K (0xa << 12) +#define WL1273_IS2_RATE (0xf << 12) + +#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \ + WL1273_IS2_FORMAT_STD | \ + WL1273_IS2_MASTER | \ + WL1273_IS2_TRI_AFTER_SENDING | \ + WL1273_IS2_SDOWS_RR | \ + WL1273_IS2_TRI_OPT | \ + WL1273_IS2_RATE_48K) + +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); + +#endif /* End of __WL1273_CODEC_H__ */