From cf4f7fc3e7336e2e946880890e60ed36178889ea Mon Sep 17 00:00:00 2001 From: Fabio Falzoi Date: Mon, 4 Aug 2014 17:08:07 +0200 Subject: ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS Add SND_SOC_DAIFMT_CBM_CFS support for Freescale architecture. Successfully tested on i.MX 6Quad Wandboard and UDOO boards connected to the pcm1792a codec. In CBM_CFS mode, when using a sample size of 16 bits, we cannot use CCSR_SSI_SCR_I2S_MODE_MASTER since we get a frame sync every 16 bits. Signed-off-by: Michael Trimarchi Signed-off-by: Fabio Falzoi Tested-by: Angelo Adamo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb5776a39b..2fc3e6683e4f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -259,6 +259,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) SND_SOC_DAIFMT_CBS_CFS; } +static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private) +{ + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFS; +} /** * fsl_ssi_isr: SSI interrupt handler * @@ -705,6 +710,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + if (!fsl_ssi_is_ac97(ssi_private)) { + u8 i2smode; + /* + * Switch to normal net mode in order to have a frame sync + * signal every 32 bits instead of 16 bits + */ + if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16) + i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL | + CCSR_SSI_SCR_NET; + else + i2smode = ssi_private->i2s_mode; + + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : i2smode); + } + /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -724,11 +746,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK, wl); - if (!fsl_ssi_is_ac97(ssi_private)) - regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, - channels == 1 ? 0 : ssi_private->i2s_mode); - return 0; } @@ -780,6 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; regmap_update_bits(regs, CCSR_SSI_STCCR, @@ -853,6 +871,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, case SND_SOC_DAIFMT_CBM_CFM: scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr &= ~CCSR_SSI_STCR_TXDIR; + strcr |= CCSR_SSI_STCR_TFDIR; + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; default: return -EINVAL; } -- cgit v1.2.3 From d177143c3670aa57ee08c73880beb55ee9d8ab7c Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 8 Aug 2014 14:47:21 +0800 Subject: ASoC: fsl_esai: refine esai for TDM support Original driver didn't store the number of slots, just fix the slot number to 2, use this default number to calculate bclk and pins for TX/RX. In this patch, add one parameter for slots, and update the calculation of bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in TDM mode. Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 14 +++++++++++--- sound/soc/fsl/fsl_esai.h | 8 ++++---- 2 files changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e7dd03..f252370073e5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -38,6 +38,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot + * @slots: number of slots * @hck_rate: clock rate of desired HCKx clock * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; u32 fifo_depth; u32 slot_width; + u32 slots; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -363,6 +365,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; + esai_priv->slots = slots; return 0; } @@ -510,10 +513,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * 2; + bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -530,7 +534,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | - (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins)); regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); @@ -565,6 +569,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -579,7 +584,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, - tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: @@ -783,6 +788,9 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot size */ esai_priv->slot_width = 32; + /* Set a default slot number */ + esai_priv->slots = 2; + /* Set a default master/slave state */ esai_priv->slave_mode = true; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 75e14033e8d8..91a550f4a10d 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -130,8 +130,8 @@ #define ESAI_xFCR_RE_WIDTH 4 #define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) #define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) -#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) -#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK) #define ESAI_xFCR_xFR_SHIFT 1 #define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) #define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) @@ -272,8 +272,8 @@ #define ESAI_xCR_RE_WIDTH 4 #define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) #define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) -#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) -#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK) /* * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 -- cgit v1.2.3 From 376d1a92ca587d3974d4791cdb99baa8b8e7f0dd Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 5 Aug 2014 17:20:21 +0800 Subject: ASoC: fsl_sai: Initialize with software reset This patch adds software reset code in dai_probe() so as to make a true init by clearing SAI's internal logic, including the bit clock generation, status flags, and FIFO pointers. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 9 +++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa049797897..7b1eecbc4f60 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f595d75..8e1feab7c2a0 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -48,6 +48,7 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 #define FSL_SAI_CSR_xF_W_SHIFT 18 #define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) -- cgit v1.2.3 From af96ff5b7448dc776dc24a5c4313c6ec1ee94e53 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 15:07:25 +0800 Subject: ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode There is one design rule according to SAI's reference manual: If the transmitter bit clock and frame sync are to be used by both transmitter and receiver, the transmitter must be configured for asynchronous operation and the receiver for synchronous operation. And SYNC of TCR2 is a 2-width control bit: 00 Asynchronous mode. 01 Synchronous with receiver. 10 Synchronous with another SAI transmitter. 11 Synchronous with another SAI receiver. So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC bit of RCR2 to 0x1 (Synchronous with transmitter). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7b1eecbc4f60..3d865ad466ad 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -333,8 +333,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * The transmitter bit clock and frame sync are to be * used by both the transmitter and receiver. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, FSL_SAI_CR2_SYNC); -- cgit v1.2.3 From 08fdf65e37d560581233e06a659f73deeb3766f9 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 5 Aug 2014 15:32:05 +0800 Subject: ASoC: fsl_sai: Add asynchronous mode support SAI supports these operation modes: 1) asynchronous mode Both Tx and Rx are set to be asynchronous. 2) synchronous mode (Rx sync with Tx) Tx is set to be asynchronous, Rx is set to be synchronous. 3) synchronous mode (Tx sync with Rx) Rx is set to be asynchronous, Tx is set to be synchronous. 4) synchronous mode (Tx/Rx sync with another SAI's Tx) 5) synchronous mode (Tx/Rx sync with another SAI's Rx) * 4) and 5) are beyond this patch because they are related with another SAI. As the initial version of this SAI driver, it supported 2) as default while the others were totally missing. So this patch just adds supports for 1) and 3). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl-sai.txt | 16 ++++++++++++ sound/soc/fsl/fsl_sai.c | 30 +++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 0f4e23828190..77864f4dd352 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -24,6 +24,22 @@ Required properties: - big-endian-data: If this property is absent, the little endian mode will be in use as default, or the big endian mode will be in use for all the fifo data. +- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transimitter and receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. +- fsl,sai-asynchronous: This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which means both transimitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + +Note: +- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transimitter and receiver will send and receive data by following clocks + of transimitter. +- fsl,sai-asynchronous will be ignored if fsl,sai-synchronous-rx property is + already present. Example: sai2: sai@40031000 { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 3d865ad466ad..ef7c758627b1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -330,12 +330,14 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, u32 xcsr, count = 100; /* - * The transmitter bit clock and frame sync are to be - * used by both the transmitter and receiver. + * Asynchronous mode: Clear SYNC for both Tx and Rx. + * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. + * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, - FSL_SAI_CR2_SYNC); + sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); /* * It is recommended that the transmitter is the last enabled @@ -625,6 +627,26 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + /* Sync Tx with Rx as default by following old DT binding */ + sai->synchronous[RX] = true; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 1; + fsl_sai_dai.symmetric_channels = 1; + fsl_sai_dai.symmetric_samplebits = 1; + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { + /* Sync Rx with Tx */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = true; + } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* Discard all settings for asynchronous mode */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 0; + fsl_sai_dai.symmetric_channels = 0; + fsl_sai_dai.symmetric_samplebits = 0; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 8e1feab7c2a0..b3d8864cd5f2 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -136,9 +136,13 @@ struct fsl_sai { bool big_endian_data; bool is_dsp_mode; bool sai_on_imx; + bool synchronous[2]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; +#define TX 1 +#define RX 0 + #endif /* __FSL_SAI_H */ -- cgit v1.2.3 From ce7344a4ebabe90e064d3e087727f45624cdc942 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 8 Aug 2014 18:41:19 +0800 Subject: ASoC: fsl_sai: Make Synchronous and Asynchronous modes exclusive The previous patch (ASoC: fsl_sai: Add asynchronous mode support) added new Device Tree bindings for Asynchronous and Synchronous modes support. However, these two shall not be present at the same time. So this patch just simply makes them exclusive so as to avoid incorrect Device Tree binding usage. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 3 +-- sound/soc/fsl/fsl_sai.c | 7 +++++++ 2 files changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 77864f4dd352..dc9f9c356268 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -38,8 +38,7 @@ Note: default synchronous mode (sync Rx with Tx) will be used, which means both transimitter and receiver will send and receive data by following clocks of transimitter. -- fsl,sai-asynchronous will be ignored if fsl,sai-synchronous-rx property is - already present. +- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. Example: sai2: sai@40031000 { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ef7c758627b1..4c9e71c2f52a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -634,6 +634,13 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_channels = 1; fsl_sai_dai.symmetric_samplebits = 1; + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && + of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* error out if both synchronous and asynchronous are present */ + dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + return -EINVAL; + } + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { /* Sync Rx with Tx */ sai->synchronous[RX] = false; -- cgit v1.2.3 From 855675f6e6a65688a7f4cf45b9b5a98cf6c6f5c3 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 15:07:25 +0800 Subject: ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode There is one design rule according to SAI's reference manual: If the transmitter bit clock and frame sync are to be used by both transmitter and receiver, the transmitter must be configured for asynchronous operation and the receiver for synchronous operation. And SYNC of TCR2 is a 2-width control bit: 00 Asynchronous mode. 01 Synchronous with receiver. 10 Synchronous with another SAI transmitter. 11 Synchronous with another SAI receiver. So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC bit of RCR2 to 0x1 (Synchronous with transmitter). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4c9e71c2f52a..60fe7c77ba22 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -334,8 +334,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); -- cgit v1.2.3 From eadb0019d206591e34e864b62059b292e157d8fc Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Fri, 29 Aug 2014 15:12:12 +0800 Subject: ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'. The 'big-endian-data' property is originally used to indicate whether the LSB firstly or MSB firstly will be transmitted to the CODEC or received from the CODEC, and there has nothing relation to the memory data. Generally, if the audio data in big endian format, which will be using the bytes reversion, Here this can only be used to bits reversion. So using the 'lsb-first' instead of 'big-endian-data' can make the code to be readable easier and more easy to understand what this property is used to do. This property used for configuring whether the LSB or the MSB is transmitted first for the fifo data. Signed-off-by: Xiubo Li Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/fsl_sai.h | 2 +- 3 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 06a405e7f3e7..4956b14d4b06 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -20,9 +20,9 @@ Required properties: See ../pinctrl/pinctrl-bindings.txt for details of the property values. - big-endian: Boolean property, required if all the FTM_PWM registers are big-endian rather than little-endian. -- big-endian-data: If this property is absent, the little endian mode will - be in use as default, or the big endian mode will be in use for all the - fifo data. +- lsb-first: Configures whether the LSB or the MSB is transmitted first for + the fifo data. If this property is absent, the MSB is transmitted first as + default, or the LSB is transmitted first. - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx with Rx) which means both the transimitter and receiver will send and receive data by following @@ -53,5 +53,5 @@ sai2: sai@40031000 { dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; big-endian; - big-endian-data; + lsb-first; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a6eb7849959c..7eeb1dd8ce27 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -573,7 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 2cded440d567..34667209b607 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -132,7 +132,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; -- cgit v1.2.3 From 02024756e6ab3a3fcdc3b203552b16b345ebd97d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 2 Sep 2014 18:05:56 +0530 Subject: ASoC: mfld: pcm: Replace pr_ with dev_ Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 44 +++++++++++++++++---------------- 1 file changed, 23 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 8e1e9bc27642..85deecd82b92 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -43,12 +43,12 @@ int sst_register_dsp(struct sst_device *dev) return -ENODEV; mutex_lock(&sst_lock); if (sst) { - pr_err("we already have a device %s\n", sst->name); + dev_err(dev->dev, "we already have a device %s\n", sst->name); module_put(dev->dev->driver->owner); mutex_unlock(&sst_lock); return -EEXIST; } - pr_debug("registering device %s\n", dev->name); + dev_dbg(dev->dev, "registering device %s\n", dev->name); sst = dev; mutex_unlock(&sst_lock); return 0; @@ -70,7 +70,7 @@ int sst_unregister_dsp(struct sst_device *dev) } module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); + dev_dbg(dev->dev, "unreg %s\n", sst->name); sst = NULL; mutex_unlock(&sst_lock); return 0; @@ -306,9 +306,10 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret_val; - pr_debug("setting buffer ptr param\n"); + dev_dbg(rtd->dev, "setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; stream->stream_info.arg = substream; @@ -316,7 +317,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info); if (ret_val) - pr_err("control_set ret error %d\n", ret_val); + dev_err(rtd->dev, "control_set ret error %d\n", ret_val); return ret_val; } @@ -337,7 +338,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, mutex_lock(&sst_lock); if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); + dev_err(dai->dev, "no device available to run\n"); ret_val = -ENODEV; goto out_ops; } @@ -385,10 +386,11 @@ static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform substream->runtime->private_data; u32 str_id = stream->stream_info.str_id; unsigned int pipe_id; + pipe_id = map[str_id].device_id; - pr_debug("%s: got pipe_id = %#x for str_id = %d\n", - __func__, pipe_id, str_id); + dev_dbg(platform->dev, "got pipe_id = %#x for str_id = %d\n", + pipe_id, str_id); return pipe_id; } @@ -459,29 +461,30 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int ret_val = 0, str_id; struct sst_runtime_stream *stream; int status; + struct snd_soc_pcm_runtime *rtd = substream->private_data; - pr_debug("sst_platform_pcm_trigger called\n"); + dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); + dev_dbg(rtd->dev, "sst: Trigger Start\n"); status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); + dev_dbg(rtd->dev, "sst: in stop\n"); status = SST_PLATFORM_DROPPED; ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); + dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); + dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; @@ -502,6 +505,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer struct sst_runtime_stream *stream; int ret_val, status; struct pcm_stream_info *str_info; + struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); @@ -510,7 +514,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer str_info = &stream->stream_info; ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { - pr_err("sst: error code = %d\n", ret_val); + dev_err(rtd->dev, "sst: error code = %d\n", ret_val); return ret_val; } substream->runtime->delay = str_info->pcm_delay; @@ -526,7 +530,7 @@ static struct snd_pcm_ops sst_platform_ops = { static void sst_pcm_free(struct snd_pcm *pcm) { - pr_debug("sst_pcm_free called\n"); + dev_dbg(pcm->dev, "sst_pcm_free called\n"); snd_pcm_lib_preallocate_free_for_all(pcm); } @@ -543,7 +547,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - pr_err("dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocationf fail\n"); return retval; } } @@ -576,13 +580,11 @@ static int sst_platform_probe(struct platform_device *pdev) drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (drv == NULL) { - pr_err("kzalloc failed\n"); return -ENOMEM; } pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) { - pr_err("kzalloc failed for pdata\n"); return -ENOMEM; } @@ -594,14 +596,14 @@ static int sst_platform_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { - pr_err("registering soc platform failed\n"); + dev_err(&pdev->dev, "registering soc platform failed\n"); return ret; } ret = snd_soc_register_component(&pdev->dev, &sst_component, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { - pr_err("registering cpu dais failed\n"); + dev_err(&pdev->dev, "registering cpu dais failed\n"); snd_soc_unregister_platform(&pdev->dev); } return ret; @@ -612,7 +614,7 @@ static int sst_platform_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); + dev_dbg(&pdev->dev, "sst_platform_remove success\n"); return 0; } -- cgit v1.2.3 From d2b16b8fa1b6352757cd0a58234591e1496a82ad Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 9 Sep 2014 15:11:24 +0530 Subject: ASoC: Intel: mfld-pcm: don't call trigger ops to DSP for internal streams For internal stream i.e. BE we have don't need trigger ops as that would be handled by DAPM for us in subsequent patches Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 85deecd82b92..9906b7c1c2e1 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -464,6 +464,8 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n"); + if (substream->pcm->internal) + return 0; stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; switch (cmd) { -- cgit v1.2.3 From 10615a5c49721803ed258316280858142a24e72a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 9 Sep 2014 15:11:25 +0530 Subject: ASoC: Intel: mrfld: add bytes control for modules This patch add support for various modules like eq etc for mrfld DSP. All these modules will be exposed to usermode as bytes controls. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 179 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 130 ++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 2 +- 3 files changed, 310 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index ace3c4a59b14..7104a34181a9 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -25,6 +25,179 @@ #include "sst-mfld-platform.h" #include "sst-atom-controls.h" +static int sst_fill_byte_control(struct sst_data *drv, + u8 ipc_msg, u8 block, + u8 task_id, u8 pipe_id, + u16 len, void *cmd_data) +{ + struct snd_sst_bytes_v2 *byte_data = drv->byte_stream; + + byte_data->type = SST_CMD_BYTES_SET; + byte_data->ipc_msg = ipc_msg; + byte_data->block = block; + byte_data->task_id = task_id; + byte_data->pipe_id = pipe_id; + + if (len > SST_MAX_BIN_BYTES - sizeof(*byte_data)) { + dev_err(&drv->pdev->dev, "command length too big (%u)", len); + return -EINVAL; + } + byte_data->len = len; + memcpy(byte_data->bytes, cmd_data, len); + print_hex_dump_bytes("writing to lpe: ", DUMP_PREFIX_OFFSET, + byte_data, len + sizeof(*byte_data)); + return 0; +} + +static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret = 0; + + ret = sst_fill_byte_control(drv, ipc_msg, + block, task_id, pipe_id, len, cmd_data); + if (ret < 0) + return ret; + return sst->ops->send_byte_stream(sst->dev, drv->byte_stream); +} + +/** + * sst_fill_and_send_cmd - generate the IPC message and send it to the FW + * @ipc_msg: type of IPC (CMD, SET_PARAMS, GET_PARAMS) + * @cmd_data: the IPC payload + */ +static int sst_fill_and_send_cmd(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret; + + mutex_lock(&drv->lock); + ret = sst_fill_and_send_cmd_unlocked(drv, ipc_msg, block, + task_id, pipe_id, cmd_data, len); + mutex_unlock(&drv->lock); + + return ret; +} + +static int sst_send_algo_cmd(struct sst_data *drv, + struct sst_algo_control *bc) +{ + int len, ret = 0; + struct sst_cmd_set_params *cmd; + + /*bc->max includes sizeof algos + length field*/ + len = sizeof(cmd->dst) + sizeof(cmd->command_id) + bc->max; + + cmd = kzalloc(len, GFP_KERNEL); + if (cmd == NULL) + return -ENOMEM; + + SST_FILL_DESTINATION(2, cmd->dst, bc->pipe_id, bc->module_id); + cmd->command_id = bc->cmd_id; + memcpy(cmd->params, bc->params, bc->max); + + ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS, + SST_FLAG_BLOCKED, bc->task_id, 0, cmd, len); + kfree(cmd); + return ret; +} + +static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = bc->max; + + return 0; +} + +static int sst_algo_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(ucontrol->value.bytes.data, bc->params, bc->max); + break; + default: + dev_err(component->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + + } + return 0; +} + +static int sst_algo_control_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt); + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + dev_dbg(cmpnt->dev, "control_name=%s\n", kcontrol->id.name); + mutex_lock(&drv->lock); + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(bc->params, ucontrol->value.bytes.data, bc->max); + break; + default: + mutex_unlock(&drv->lock); + dev_err(cmpnt->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + } + /*if pipe is enabled, need to send the algo params from here*/ + if (bc->w && bc->w->power) + ret = sst_send_algo_cmd(drv, bc); + mutex_unlock(&drv->lock); + + return ret; +} + +static const struct snd_kcontrol_new sst_algo_controls[] = { + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("sprot_loop_out", "lpro", 192, SST_MODULE_ID_SPROT, + SST_PATH_INDEX_SPROT_LOOP_OUT, 0, SST_TASK_SBA, SBA_VB_LPRO), + SST_ALGO_KCONTROL_BYTES("codec_in0", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN0, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("codec_in1", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN1, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + +}; + +static int sst_algo_control_init(struct device *dev) +{ + int i = 0; + struct sst_algo_control *bc; + /*allocate space to cache the algo parameters in the driver*/ + for (i = 0; i < ARRAY_SIZE(sst_algo_controls); i++) { + bc = (struct sst_algo_control *)sst_algo_controls[i].private_value; + bc->params = devm_kzalloc(dev, bc->max, GFP_KERNEL); + if (bc->params == NULL) + return -ENOMEM; + } + return 0; +} + int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) { int ret = 0; @@ -35,5 +208,11 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) if (!drv->byte_stream) return -ENOMEM; + /*Initialize algo control params*/ + ret = sst_algo_control_init(platform->dev); + if (ret) + return ret; + ret = snd_soc_add_platform_controls(platform, sst_algo_controls, + ARRAY_SIZE(sst_algo_controls)); return ret; } diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 8554889c0694..a73e894b175c 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -309,4 +309,134 @@ enum sst_swm_state { SST_SWM_ON = 3, }; +#define SST_FILL_LOCATION_IDS(dst, cell_idx, pipe_id) do { \ + dst.location_id.p.cell_nbr_idx = (cell_idx); \ + dst.location_id.p.path_id = (pipe_id); \ + } while (0) +#define SST_FILL_LOCATION_ID(dst, loc_id) (\ + dst.location_id.f = (loc_id)) +#define SST_FILL_MODULE_ID(dst, mod_id) (\ + dst.module_id = (mod_id)) + +#define SST_FILL_DESTINATION1(dst, id) do { \ + SST_FILL_LOCATION_ID(dst, (id) & 0xFFFF); \ + SST_FILL_MODULE_ID(dst, ((id) & 0xFFFF0000) >> 16); \ + } while (0) +#define SST_FILL_DESTINATION2(dst, loc_id, mod_id) do { \ + SST_FILL_LOCATION_ID(dst, loc_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) +#define SST_FILL_DESTINATION3(dst, cell_idx, path_id, mod_id) do { \ + SST_FILL_LOCATION_IDS(dst, cell_idx, path_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) + +#define SST_FILL_DESTINATION(level, dst, ...) \ + SST_FILL_DESTINATION##level(dst, __VA_ARGS__) +#define SST_FILL_DEFAULT_DESTINATION(dst) \ + SST_FILL_DESTINATION(2, dst, SST_DEFAULT_LOCATION_ID, SST_DEFAULT_MODULE_ID) + +struct sst_destination_id { + union sst_location_id { + struct { + u8 cell_nbr_idx; /* module index */ + u8 path_id; /* pipe_id */ + } __packed p; /* part */ + u16 f; /* full */ + } __packed location_id; + u16 module_id; +} __packed; +struct sst_dsp_header { + struct sst_destination_id dst; + u16 command_id; + u16 length; +} __packed; + +/* + * + * Common Commands + * + */ +struct sst_cmd_generic { + struct sst_dsp_header header; +} __packed; +struct sst_cmd_set_params { + struct sst_destination_id dst; + u16 command_id; + char params[0]; +} __packed; +#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \ + xpname " " xmname " " #xinstance " " xtype + +#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \ + xpname " " xmname " " #xinstance " " xtype " " xsubmodule +enum sst_algo_kcontrol_type { + SST_ALGO_PARAMS, + SST_ALGO_BYPASS, +}; + +struct sst_algo_control { + enum sst_algo_kcontrol_type type; + int max; + u16 module_id; + u16 pipe_id; + u16 task_id; + u16 cmd_id; + bool bypass; + unsigned char *params; + struct snd_soc_dapm_widget *w; +}; + +/* size of the control = size of params + size of length field */ +#define SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, xmod, xtask, xcmd) \ + (struct sst_algo_control){ \ + .max = xcount + sizeof(u16), .type = xtype, .module_id = xmod, \ + .pipe_id = xpipe, .task_id = xtask, .cmd_id = xcmd, \ + } + +#define SST_ALGO_KCONTROL(xname, xcount, xmod, xpipe, \ + xtask, xcmd, xtype, xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = xinfo, .get = xget, .put = xput, \ + .private_value = (unsigned long)& \ + SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, \ + xmod, xtask, xcmd), \ +} + +#define SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "params"), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "bypass"), \ + 0, xmod, xpipe, xtask, 0, SST_ALGO_BYPASS, \ + snd_soc_info_bool_ext, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_BYPASS_PARAMS(xpname, xmname, xcount, xmod, xpipe, \ + xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask), \ + SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, xpipe, xinstance, xtask, xcmd) + +#define SST_COMBO_ALGO_KCONTROL_BYTES(xpname, xmname, xsubmod, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, "params", \ + xsubmod), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + + +struct sst_enum { + bool tx; + unsigned short reg; + unsigned int max; + const char * const *texts; + struct snd_soc_dapm_widget *w; +}; + #endif diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 0c5b943daff3..7092ee3e96a3 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -166,7 +166,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; - char *byte_stream; + struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); -- cgit v1.2.3 From 6df5d768050f31d810dd3ba0ad8210922c3e9b6d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 9 Sep 2014 15:11:32 +0530 Subject: ASoC: Intel: mrfld: Use snd_soc_dai_get_drvdata to derive drv data Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 9906b7c1c2e1..8e1b2c14291c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -252,7 +252,7 @@ int sst_fill_stream_params(void *substream, } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, - struct snd_soc_platform *platform) + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream = substream->runtime->private_data; @@ -260,7 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; - struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); @@ -377,10 +377,10 @@ static void sst_media_close(struct snd_pcm_substream *substream, kfree(stream); } -static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, +static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_data *sst = snd_soc_dai_get_drvdata(dai); struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; struct sst_runtime_stream *stream = substream->runtime->private_data; @@ -389,7 +389,7 @@ static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform pipe_id = map[str_id].device_id; - dev_dbg(platform->dev, "got pipe_id = %#x for str_id = %d\n", + dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n", pipe_id, str_id); return pipe_id; } @@ -407,7 +407,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; } - ret_val = sst_platform_alloc_stream(substream, dai->platform); + ret_val = sst_platform_alloc_stream(substream, dai); if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), -- cgit v1.2.3 From f4a43caba7d495699f98532b4faee90fd9980732 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 16 Sep 2014 10:13:16 +0800 Subject: ASoC: fsl_ssi: refine ipg clock usage in this module Check if ipg clock is in clock-names property, then we can move the ipg clock enable and disable operation to startup and shutdown, that is only enable ipg clock when ssi is working and keep clock is disabled when ssi is in idle. But when the checking is failed, remain the clock control as before. Tested-by: Markus Pargmann Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 53 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 45 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2fc3e6683e4f..16a1361b68b3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -169,6 +169,7 @@ struct fsl_ssi_private { u8 i2s_mode; bool use_dma; bool use_dual_fifo; + bool has_ipg_clk_name; unsigned int fifo_depth; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; @@ -530,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; /* When using dual fifo mode, it is safer to ensure an even period * size. If appearing to an odd number while DMA always starts its @@ -543,6 +549,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, return 0; } +/** + * fsl_ssi_shutdown: shutdown the SSI + * + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + clk_disable_unprepare(ssi_private->clk); + +} + /** * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock * @@ -1043,6 +1064,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, @@ -1168,17 +1190,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, u32 dmas[4]; int ret; - ssi_private->clk = devm_clk_get(&pdev->dev, NULL); + if (ssi_private->has_ipg_clk_name) + ssi_private->clk = devm_clk_get(&pdev->dev, "ipg"); + else + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); return ret; } - ret = clk_prepare_enable(ssi_private->clk); - if (ret) { - dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); - return ret; + if (!ssi_private->has_ipg_clk_name) { + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } } /* For those SLAVE implementations, we ingore non-baudclk cases @@ -1236,8 +1263,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return 0; error_pcm: - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); return ret; } @@ -1246,7 +1274,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev, { if (!ssi_private->use_dma) imx_pcm_fiq_exit(pdev); - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); } static int fsl_ssi_probe(struct platform_device *pdev) @@ -1321,8 +1350,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENOMEM; } - ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, + ret = of_property_match_string(np, "clock-names", "ipg"); + if (ret < 0) { + ssi_private->has_ipg_clk_name = false; + ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, &fsl_ssi_regconfig); + } else { + ssi_private->has_ipg_clk_name = true; + ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, + "ipg", iomem, &fsl_ssi_regconfig); + } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); return PTR_ERR(ssi_private->regs); -- cgit v1.2.3 From 0121327c1a68bc8c80f240c2794e682722b69051 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 19 Sep 2014 16:46:03 +0530 Subject: ASoC: Intel: mfld-pcm: add control for powering up/down dsp When we have PCM (FE/BE) opened or DAPM widgets triggered we need power up/down DSP accordingly. The DSP will do ref count of these requests i.e. link these runtime_get/put calls of DSP Also fix some preexisting spacing error. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 16 ++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 17 +++++++++-------- 2 files changed, 25 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 8e1b2c14291c..aa9b600dfc9b 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -322,6 +322,16 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } +static int power_up_sst(struct sst_runtime_stream *stream) +{ + return stream->ops->power(sst->dev, true); +} + +static void power_down_sst(struct sst_runtime_stream *stream) +{ + stream->ops->power(sst->dev, false); +} + static int sst_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -351,6 +361,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, /* allocate memory for SST API set */ runtime->private_data = stream; + ret_val = power_up_sst(stream); + if (ret_val < 0) + return ret_val; + /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, 2); @@ -370,6 +384,8 @@ static void sst_media_close(struct snd_pcm_substream *substream, int ret_val = 0, str_id; stream = substream->runtime->private_data; + power_down_sst(stream); + str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(sst->dev, str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 7092ee3e96a3..19f83ec51613 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -120,15 +120,16 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct device *dev, struct snd_sst_params *str_param); - int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info); - int (*stream_start) (struct device *dev, int str_id); - int (*stream_drop) (struct device *dev, int str_id); - int (*stream_pause) (struct device *dev, int str_id); - int (*stream_pause_release) (struct device *dev, int str_id); - int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info); + int (*open)(struct device *dev, struct snd_sst_params *str_param); + int (*stream_init)(struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start)(struct device *dev, int str_id); + int (*stream_drop)(struct device *dev, int str_id); + int (*stream_pause)(struct device *dev, int str_id); + int (*stream_pause_release)(struct device *dev, int str_id); + int (*stream_read_tstamp)(struct device *dev, struct pcm_stream_info *str_info); int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); - int (*close) (struct device *dev, unsigned int str_id); + int (*close)(struct device *dev, unsigned int str_id); + int (*power)(struct device *dev, bool state); }; struct sst_runtime_stream { -- cgit v1.2.3 From 969168e2e9f4a5bfd6a49344f46b820437cd9163 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 26 Sep 2014 16:25:38 +0300 Subject: ASoC: Intel: byt-max98090: Set card as fully routed All byt-max98090 audio connections are known and described in DAPM routing table. Set the fully_routed flag in order to disable unused codec pins. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index b8b8af571ef1..d52681e7225e 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -139,6 +139,7 @@ static struct snd_soc_card byt_max98090_card = { .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), .controls = byt_max98090_controls, .num_controls = ARRAY_SIZE(byt_max98090_controls), + .fully_routed = true, }; static int byt_max98090_probe(struct platform_device *pdev) -- cgit v1.2.3 From ac06dd8df6e13591524f0e1bedf36af4ca0e967b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 29 Sep 2014 16:58:15 +0300 Subject: ASoC: Intel: byt-rt5640: Remove IN2N pin from DAPM route table I tested couple byt-rt5640 based platforms and they have single-ended headset microphone connection to IN2P only. I guess IN2N was either defined by accident or some early platform had floating ground for headset. It's better to remove IN2N and add a custom route for such a platform if needed. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 234a58de3c53..d6d8b19c22dc 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -36,7 +36,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, -- cgit v1.2.3 From f8a770c2c67f28956f8f4601feb99e9bd02a16c8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 29 Sep 2014 16:58:16 +0300 Subject: ASoC: Intel: byt-rt5640: Add quirk for Asus T100 Asus T100 internal microphone is not digital but analogue connected to IN1P pin of the RT564x codec with shared bias between internal and headset microphones. Because of this there is need to have machine specific DAPM routes in byt-rt5640. Add handling for them with the help of DMI quirk that is used to add custom routes in addition to common. Because "Internal Mic" connected to DMIC1 is not common to all move it as a default custom route when there is no match in quirk table. Custom "Internal Mic" -> "IN1P" with MICBIAS1 route is added for Asus T100. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 53 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index d6d8b19c22dc..c323a101214e 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -36,7 +37,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "SPOLP"}, @@ -45,6 +45,22 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Speaker", NULL, "SPORN"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5640_DMIC1_MAP, + BYT_RT5640_IN1_MAP, +}; + +static unsigned long byt_rt5640_custom_map = BYT_RT5640_DMIC1_MAP; + static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -76,12 +92,32 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, return 0; } +static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) +{ + byt_rt5640_custom_map = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), + }, + .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, + }, + {} +}; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; card->dapm.idle_bias_off = true; @@ -92,6 +128,21 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } + dmi_check_system(byt_rt5640_quirk_table); + switch (byt_rt5640_custom_map) { + case BYT_RT5640_IN1_MAP: + custom_map = byt_rt5640_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); + break; + default: + custom_map = byt_rt5640_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); + }; + + ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); + if (ret) + return ret; + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); snd_soc_dapm_ignore_suspend(dapm, "HPOR"); -- cgit v1.2.3 From ece1e4999606fc323aee96a1cdb9b7991c01dd09 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 29 Sep 2014 23:25:29 -0300 Subject: ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property There is no need to use 'i2s-slave' property, since master/slave configuration are passed via machine layer. This change does not break existing users because they do check for slave mode inside sound/soc/fsl/mpc8610_hpcd.c/p1022_ds.c/p1022_rdk.c Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 16a1361b68b3..f19224ee5b03 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1314,9 +1314,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; - else if (!strcmp(sprop, "i2s-slave")) - ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM; } ssi_private->use_dma = !of_property_read_bool(np, -- cgit v1.2.3 From cd69dc8868d64cfa2993944607d9e97927d95987 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 1 Oct 2014 15:08:14 +0300 Subject: ASoC: rt5640: Add function for enabling DMIC from ACPI probed machine There is no code enabling DMIC clock in systems that don't provide platform data for rt5640 after commit 71d97a794301 ("ASoC: rt5640: Use the platform data for DMIC settings"). I think it's worth to keep this static DMIC clock and alternative data pin setting during probe time. For making possible to use DMIC from ACPI probed machine (prior ACPI 5.1 with _DSD) this patch moves DMIC configuration to new exported rt5640_dmic_enable() that machine drivers can call. Please note, this patch moves DMIC configuration from i2c probe to codec probe in case platform data for rt5640 is set. Signed-off-by: Jarkko Nikula Cc: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 49 +++++++++++++++++++++++++++++------------------ sound/soc/codecs/rt5640.h | 3 +++ 2 files changed, 33 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efdec550..2fdcbb8e8a2a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1906,6 +1906,32 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, return 0; } +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5640_dmic_enable); + static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1945,6 +1971,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) return -ENODEV; } + if (rt5640->pdata.dmic_en) + rt5640_dmic_enable(codec, rt5640->pdata.dmic1_data_pin, + rt5640->pdata.dmic2_data_pin); + return 0; } @@ -2194,25 +2224,6 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); - if (rt5640->pdata.dmic_en) { - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); - - if (rt5640->pdata.dmic1_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); - } - - if (rt5640->pdata.dmic2_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); - } - } - rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96b86da..3deb8babeabb 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2097,4 +2097,7 @@ struct rt5640_priv { bool hp_mute; }; +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin); + #endif -- cgit v1.2.3 From a5f0ab05b67213ef33107b716e8596a480b5875f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 1 Oct 2014 15:08:15 +0300 Subject: ASoC: Intel: byt-rt5640: Enable DMIC interface for default DAPM route It turned out DMIC interface wasn't enabled/disabled runtime for active DMIC route in the rt5640 codec driver anymore after commit 71d97a794301 ("ASoC: rt5640: Use the platform data for DMIC settings"). Since DMIC interface must be enabled explicitly either by passing platform data to rt5640 codec driver or by calling new rt5640_dmic_enable() this patch adds a DMI quirk flag that is used to conditionally enable DMIC interface during sound card init time. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index c323a101214e..8392c160d9e2 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -59,7 +59,11 @@ enum { BYT_RT5640_IN1_MAP, }; -static unsigned long byt_rt5640_custom_map = BYT_RT5640_DMIC1_MAP; +#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5640_DMIC_EN BIT(16) + +static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP | + BYT_RT5640_DMIC_EN; static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), @@ -94,7 +98,7 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) { - byt_rt5640_custom_map = (unsigned long)id->driver_data; + byt_rt5640_quirk = (unsigned long)id->driver_data; return 1; } @@ -129,7 +133,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) } dmi_check_system(byt_rt5640_quirk_table); - switch (byt_rt5640_custom_map) { + switch (BYT_RT5640_MAP(byt_rt5640_quirk)) { case BYT_RT5640_IN1_MAP: custom_map = byt_rt5640_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); @@ -143,6 +147,12 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + ret = rt5640_dmic_enable(codec, 0, 0); + if (ret) + return ret; + } + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); snd_soc_dapm_ignore_suspend(dapm, "HPOR"); -- cgit v1.2.3 From 24221dcc8be736a2b0b83ecaeb60b99bd7e9334c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 2 Oct 2014 13:29:08 +0300 Subject: ASoC: Intel: byt-rt5640: Add quirk for Dell Venue 8 Pro tablet It was found with help of Jan-Michael Brummer that Dell Venue 8 Pro tablet has a digital microphone connected to DMIC2 interface of the RT564x. This patch adds a DAPM route to DMIC2 and a quirk using it for that tablet. Signed-off-by: Jarkko Nikula Reported-by: Jan-Michael Brummer Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 8392c160d9e2..a9619b4201f9 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -49,6 +49,10 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { {"DMIC1", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, @@ -56,6 +60,7 @@ static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { enum { BYT_RT5640_DMIC1_MAP, + BYT_RT5640_DMIC2_MAP, BYT_RT5640_IN1_MAP, }; @@ -111,6 +116,15 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, }, + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), + }, + .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | + BYT_RT5640_DMIC_EN), + }, {} }; @@ -138,6 +152,10 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5640_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); break; + case BYT_RT5640_DMIC2_MAP: + custom_map = byt_rt5640_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); + break; default: custom_map = byt_rt5640_intmic_dmic1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); -- cgit v1.2.3 From c47a39a6806d756c34eb01b1081866845fb76dc3 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 2 Oct 2014 13:29:09 +0300 Subject: ASoC: Intel: byt-rt5640: Set card as fully routed Although it's not known does current version of byt-rt5640 cover all possible variants it is better to set the fully_routed flag on in order to disable unused codecs pins in known machines and get regression from machines that use different routing than the default one. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index a9619b4201f9..88ad57fc58b2 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -209,6 +209,7 @@ static struct snd_soc_card byt_rt5640_card = { .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), .dapm_routes = byt_rt5640_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), + .fully_routed = true, }; static int byt_rt5640_probe(struct platform_device *pdev) -- cgit v1.2.3 From bb78cdd4914df22bdf233a9cd4b554a1f6e39804 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Sat, 4 Oct 2014 19:09:33 +0100 Subject: ASoC: Intel: byt-rt5640: fix coccinelle warnings sound/soc/intel/byt-rt5640.c:140:2-3: Unneeded semicolon Removes unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 88ad57fc58b2..e03abdf21c1b 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -159,7 +159,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) default: custom_map = byt_rt5640_intmic_dmic1_map; num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); - }; + } ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); if (ret) -- cgit v1.2.3