Add ASoC driver bindings documentation. Describe the required properties for each of the hardware IPs drivers.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++++++++++++++++++++ 1 file changed, 155 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt

diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt new file mode 100644 index 0000000..028fa1c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -0,0 +1,155 @@ +STMicroelectronics sti ASoC cards + +The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas +codec or external codecs. + +sti sound drivers allows to expose sti SoC audio interface through the +generic ASoC simple card. For details about sound card declaration please refer to +Documentation/devicetree/bindings/sound/simple-card.txt. + +1) sti-uniperiph-dai: audio dai device. +--------------------------------------- + +Required properties: + - compatible: "st,sti-uni-player" or "st,sti-uni-reader" + + - st,syscfg: phandle to boot-device system configuration registers + + - clock-names: name of the clocks listed in clocks property in the same order + + - reg: CPU DAI IP Base address and size entries, listed in same + order than the CPU_DAI properties. + + - reg-names: names of the mapped memory regions listed in regs property in + the same order. + + - interrupts: CPU_DAI interrupt line, listed in the same order than the + CPU_DAI properties. + + - dma: CPU_DAI DMA controller phandle and DMA request line, listed in the same + order than the CPU_DAI properties. + + - dma-names: identifier string for each DMA request line in the dmas property. + "tx" for "st,sti-uni-player" compatibility + "rx" for "st,sti-uni-reader" compatibility + + - version: IP version integrated in SOC. + + - dai-name: DAI name that describes the IP. + +Required properties ("st,sti-uni-player" compatibility only): + - clocks: CPU_DAI IP clock source, listed in the same order than the + CPU_DAI properties. + + - uniperiph-id: internal SOC IP instance ID. + + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec IP and IEC HDMI formats. + "SPDIF"connected to SPDIF codec and support SPDIF formats. + "PCM" PCM standard mode for I2S or TDM bus. + +Optional properties: + - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for + external codecs connection. + + - pinctrl-names: should contain only one value - "default". + +Example: + + sti_uni_player2: sti-uni-player@2 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_2>; + reg = <0x8D82000 0x158>; + interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>; + dmas = <&fdma0 4 0 1>; + dai-name = "Uni Player #1 (DAC)"; + dma-names = "tx"; + uniperiph-id = <2>; + version = <5>; + mode = "PCM"; + }; + + sti_uni_player3: sti-uni-player@3 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; + reg = <0x8D85000 0x158>; + interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>; + dmas = <&fdma0 7 0 1>; + dma-names = "tx"; + dai-name = "Uni Player #1 (PIO)"; + uniperiph-id = <3>; + version = <5>; + mode = "SPDIF"; + }; + + sti_uni_reader1: sti-uni-reader@1 { + compatible = "st,sti-uni-reader"; + status = "disabled"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + reg = <0x8D84000 0x158>; + interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>; + dmas = <&fdma0 6 0 1>; + dma-names = "rx"; + dai-name = "Uni Reader #1 (HDMI RX)"; + version = <3>; + }; + +2) sti-sas-codec: internal audio codec IPs driver +------------------------------------------------- + +Required properties: + - compatible: "st,sti<chip>-sas-codec" . + Should be chip "st,stih416-sas-codec" or "st,stih407-sas-codec" + + - st,syscfg: phandle to boot-device system configuration registers. + + - pinctrl-0: SPDIF PIO description. + + - pinctrl-names: should contain only one value - "default". + +Example: + sti_sas_codec: sti-sas-codec { + compatible = "st,stih407-sas-codec"; + #sound-dai-cells = <1>; + st,reg_audio = <&syscfg_core>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdif_out >; + }; + +Example of audio card declaration: + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sti audio card"; + status = "okay"; + + simple-audio-card,dai-link@0 { + /* DAC */ + format = "i2s"; + dai-tdm-slot-width = <32>; + cpu { + sound-dai = <&sti_uni_player2>; + }; + + codec { + sound-dai = <&sti_sasg_codec 1>; + }; + }; + simple-audio-card,dai-link@1 { + /* SPDIF */ + format = "left_j"; + cpu { + sound-dai = <&sti_uni_player3>; + }; + + codec { + sound-dai = <&sti_sasg_codec 0>; + }; + }; + };

Add code to manage Uniperipheral player IP instances. These DAIs are dedicated to playback and support I2S and IEC modes.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- sound/soc/sti/uniperif.h | 116 ++++++ sound/soc/sti/uniperif_player.c | 890 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 1006 insertions(+) create mode 100644 sound/soc/sti/uniperif_player.c

diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 043853e..a8d9e94 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -10,6 +10,8 @@

#include <linux/regmap.h>

+#include <sound/dmaengine_pcm.h> + /* * Register access macros */ @@ -1097,3 +1099,117 @@ UNIPERIF_DBG_STANDBY_LEFT_SP_OFFSET(ip), \ UNIPERIF_DBG_STANDBY_LEFT_SP_SHIFT(ip), \ UNIPERIF_DBG_STANDBY_LEFT_SP_MASK(ip), value) + +/* + * uniperipheral IP capabilities + */ + +#define UNIPERIF_FIFO_SIZE 70 /* FIFO is 70 cells deep */ +#define UNIPERIF_FIFO_FRAMES 4 /* FDMA trigger limit in frames */ + +/* + * Uniperipheral IP revisions + */ +enum uniperif_version { + SND_ST_UNIPERIF_VERSION_UNKNOWN, + /* SASG1 (Orly), Newman */ + SND_ST_UNIPERIF_VERSION_C6AUD0_UNI_1_0, + /* SASC1, SASG2 (Orly2) */ + SND_ST_UNIPERIF_VERSION_UNI_PLR_1_0, + /* SASC1, SASG2 (Orly2), TELSS, Cannes */ + SND_ST_UNIPERIF_VERSION_UNI_RDR_1_0, + /* TELSS (SASC1) */ + SND_ST_UNIPERIF_VERSION_TDM_PLR_1_0, + /* Cannes/Monaco */ + SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 +}; + +enum uniperif_type { + SND_ST_UNIPERIF_PLAYER_TYPE_NONE, + SND_ST_UNIPERIF_PLAYER_TYPE_HDMI, + SND_ST_UNIPERIF_PLAYER_TYPE_PCM, + SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF +}; + +enum uniperif_state { + UNIPERIF_STATE_STOPPED, + UNIPERIF_STATE_STARTED, + UNIPERIF_STATE_STANDBY, + UNIPERIF_STATE_UNDERFLOW, + UNIPERIF_STATE_OVERFLOW = UNIPERIF_STATE_UNDERFLOW, + UNIPERIF_STATE_XRUN +}; + +enum uniperif_iec958_encoding_mode { + UNIPERIF_IEC958_ENCODING_MODE_PCM, + UNIPERIF_IEC958_ENCODING_MODE_ENCODED +}; + +struct uniperif_info { + int id; /* instance value of the uniperipheral IP */ + enum uniperif_type player_type; + int underflow_enabled; /* Underflow recovery mode */ +}; + +struct uniperif_iec958_settings { + enum uniperif_iec958_encoding_mode encoding_mode; + struct snd_aes_iec958 iec958; +}; + +struct uniperif { + /* System information */ + struct uniperif_info *info; + struct device *dev; + int ver; /* IP version, used by register access macros */ + struct regmap_field *clk_sel; + + /* capabilities */ + const struct snd_pcm_hardware *hw; + + /* Resources */ + struct resource *mem_region; + void *base; + unsigned long fifo_phys_address; + int irq; + + /* Clocks */ + struct clk *clk; + int mclk; + + /* Runtime data */ + enum uniperif_state state; + + struct snd_pcm_substream *substream; + + /* Specific to IEC958 player */ + struct uniperif_iec958_settings stream_settings; + + /*alsa ctrl*/ + struct snd_kcontrol_new *snd_ctrls; + int num_ctrls; + + /* dai properties */ + unsigned int daifmt; + + /* DAI callbacks */ + const struct snd_soc_dai_ops *dai_ops; +}; + +struct sti_uniperiph_dai { + int stream; + struct uniperif *uni; + struct snd_dmaengine_dai_dma_data dma_data; +}; + +struct sti_uniperiph_data { + struct platform_device *pdev; + struct snd_soc_dai_driver *dai; + struct sti_uniperiph_dai dai_data; +}; + +/* uniperiph player*/ +int uni_player_init(struct platform_device *pdev, + struct uniperif *uni_player); +int uni_player_resume(struct uniperif *player); + +#endif diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c new file mode 100644 index 0000000..d66d633 --- /dev/null +++ b/sound/soc/sti/uniperif_player.c @@ -0,0 +1,890 @@ +/* + * Copyright (C) STMicroelectronics SA 2015 + * Authors: Arnaud Pouliquen arnaud.pouliquen@st.com + * for STMicroelectronics. + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/syscon.h> + +#include <sound/asoundef.h> +#include <sound/soc.h> + +#include "uniperif.h" + +/* + * Some hardware-related definitions + */ + +/* sys config registers definitions */ +#define SYS_CFG_AUDIO_GLUE 0xA4 +#define SYS_CFG_AUDI0_GLUE_PCM_CLKX 8 + +/* + * Driver specific types. + */ +#define UNIPERIF_PLAYER_TYPE_IS_HDMI(p) \ + ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_HDMI) +#define UNIPERIF_PLAYER_TYPE_IS_PCM(p) \ + ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_PCM) +#define UNIPERIF_PLAYER_TYPE_IS_SPDIF(p) \ + ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF) +#define UNIPERIF_PLAYER_TYPE_IS_IEC958(p) \ + (UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \ + UNIPERIF_PLAYER_TYPE_IS_SPDIF(p)) + +/* + * Note: snd_pcm_hardware is linked to DMA controller but is declared here to + * integrate DAI_CPU capability in term of rate and supported channels + */ +const struct snd_pcm_hardware uni_player_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE, + + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 192000, + + .channels_min = 2, + .channels_max = 8, + + .periods_min = 2, + .periods_max = 48, + + .period_bytes_min = 128, + .period_bytes_max = 64 * PAGE_SIZE, + .buffer_bytes_max = 256 * PAGE_SIZE +}; + +static inline int reset_player(struct uniperif *player) +{ + int count = 10; + + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { + while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) { + udelay(5); + count--; + } + } + + if (!count) { + dev_err(player->dev, "Failed to reset uniperif"); + return -EIO; + } + + return 0; +} + +/* + * uni_player_irq_handler + * In case of error audio stream is stopped; stop action is protected via PCM + * stream lock to avoid race condition with trigger callback. + */ +static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) +{ + irqreturn_t ret = IRQ_NONE; + struct uniperif *player = dev_id; + unsigned int status; + unsigned int tmp; + + if (player->state == UNIPERIF_STATE_STOPPED) { + /* Unexpected IRQ: do nothing */ + return IRQ_NONE; + } + + /* Get interrupt status & clear them immediately */ + status = GET_UNIPERIF_ITS(player); + SET_UNIPERIF_ITS_BCLR(player, status); + + /* Check for fifo error (underrun) */ + if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) { + dev_err(player->dev, "FIFO underflow error detected"); + + /* Interrupt is just for information when underflow recovery */ + if (player->info->underflow_enabled) { + /* Update state to underflow */ + player->state = UNIPERIF_STATE_UNDERFLOW; + + } else { + /* Disable interrupt so doesn't continually fire */ + SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); + + /* Stop the player */ + snd_pcm_stream_lock(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(player->substream); + } + + ret = IRQ_HANDLED; + } + + /* Check for dma error (overrun) */ + if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) { + dev_err(player->dev, "DMA error detected"); + + /* Disable interrupt so doesn't continually fire */ + SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); + + /* Stop the player */ + snd_pcm_stream_lock(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(player->substream); + + ret = IRQ_HANDLED; + } + + /* Check for underflow recovery done */ + if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) { + if (!player->info->underflow_enabled) { + dev_err(player->dev, "unexpected Underflow recovering"); + return -EPERM; + } + /* Read the underflow recovery duration */ + tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player); + + /* Clear the underflow recovery duration */ + SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player); + + /* Update state to started */ + player->state = UNIPERIF_STATE_STARTED; + + ret = IRQ_HANDLED; + } + + /* Check if underflow recovery failed */ + if (unlikely(status & + UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) { + dev_err(player->dev, "Underflow recovery failed"); + + /* Stop the player */ + snd_pcm_stream_lock(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(player->substream); + + ret = IRQ_HANDLED; + } + + return ret; +} + +static void uni_player_set_channel_status(struct uniperif *player, + struct snd_pcm_runtime *runtime) +{ + int n; + unsigned int status; + + /* + * Some AVRs and TVs require the channel status to contain a correct + * sampling frequency. If no sample rate is already specified, then + * set one. + */ + if (runtime && (player->stream_settings.iec958.status[3] + == IEC958_AES3_CON_FS_NOTID)) { + switch (runtime->rate) { + case 22050: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_22050; + break; + case 44100: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_44100; + break; + case 88200: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_88200; + break; + case 176400: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_176400; + break; + case 24000: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_24000; + break; + case 48000: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_48000; + break; + case 96000: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_96000; + break; + case 192000: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_192000; + break; + case 32000: + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_32000; + break; + default: + /* Mark as sampling frequency not indicated */ + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_NOTID; + break; + } + } + + /* Audio mode: + * Use audio mode status to select PCM or encoded mode + */ + if (player->stream_settings.iec958.status[0] & IEC958_AES0_NONAUDIO) + player->stream_settings.encoding_mode = + UNIPERIF_IEC958_ENCODING_MODE_ENCODED; + else + player->stream_settings.encoding_mode = + UNIPERIF_IEC958_ENCODING_MODE_PCM; + + if (player->stream_settings.encoding_mode == + UNIPERIF_IEC958_ENCODING_MODE_PCM) + /* Clear user validity bits */ + SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); + else + /* Set user validity bits */ + SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 1); + + /* Program the new channel status */ + for (n = 0; n < 6; ++n) { + status = + player->stream_settings.iec958.status[0 + (n * 4)] & 0xf; + status |= + player->stream_settings.iec958.status[1 + (n * 4)] << 8; + status |= + player->stream_settings.iec958.status[2 + (n * 4)] << 16; + status |= + player->stream_settings.iec958.status[3 + (n * 4)] << 24; + SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); + } + + /* Update the channel status */ + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + SET_UNIPERIF_CONFIG_CHL_STS_UPDATE(player); + else + SET_UNIPERIF_BIT_CONTROL_CHL_STS_UPDATE(player); +} + +static int uni_player_prepare_iec958(struct uniperif *player, + struct snd_pcm_runtime *runtime) +{ + int clk_div; + + clk_div = player->mclk / runtime->rate; + + /* Oversampling must be multiple of 128 as iec958 frame is 32-bits */ + if ((clk_div % 128) || (clk_div <= 0)) { + dev_err(player->dev, "%s: invalid clk_div %d", + __func__, clk_div); + return -EINVAL; + } + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + /* 16/16 memory format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_16(player); + /* 16-bits per sub-frame */ + SET_UNIPERIF_I2S_FMT_NBIT_32(player); + /* Set 16-bit sample precision */ + SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player); + break; + case SNDRV_PCM_FORMAT_S32_LE: + /* 16/0 memory format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player); + /* 32-bits per sub-frame */ + SET_UNIPERIF_I2S_FMT_NBIT_32(player); + /* Set 24-bit sample precision */ + SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player); + break; + default: + dev_err(player->dev, "format not supported"); + return -EINVAL; + } + + /* Set parity to be calculated by the hardware */ + SET_UNIPERIF_CONFIG_PARITY_CNTR_BY_HW(player); + + /* Set channel status bits to be inserted by the hardware */ + SET_UNIPERIF_CONFIG_CHANNEL_STA_CNTR_BY_HW(player); + + /* Set user data bits to be inserted by the hardware */ + SET_UNIPERIF_CONFIG_USER_DAT_CNTR_BY_HW(player); + + /* Set validity bits to be inserted by the hardware */ + SET_UNIPERIF_CONFIG_VALIDITY_DAT_CNTR_BY_HW(player); + + /* Set full software control to disabled */ + SET_UNIPERIF_CONFIG_SPDIF_SW_CTRL_DISABLE(player); + + SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + + /* Update the channel status */ + uni_player_set_channel_status(player, runtime); + + /* Clear the user validity user bits */ + SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); + + /* Disable one-bit audio mode */ + SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player); + + /* Enable consecutive frames repetition of Z preamble (not for HBRA) */ + SET_UNIPERIF_CONFIG_REPEAT_CHL_STS_ENABLE(player); + + /* Change to SUF0_SUBF1 and left/right channels swap! */ + SET_UNIPERIF_CONFIG_SUBFRAME_SEL_SUBF1_SUBF0(player); + + /* Set data output as MSB first */ + SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); + + if (player->stream_settings.encoding_mode == + UNIPERIF_IEC958_ENCODING_MODE_ENCODED) + SET_UNIPERIF_CTRL_EXIT_STBY_ON_EOBLOCK_ON(player); + else + SET_UNIPERIF_CTRL_EXIT_STBY_ON_EOBLOCK_OFF(player); + + SET_UNIPERIF_I2S_FMT_NUM_CH(player, runtime->channels / 2); + + /* Set rounding to off */ + SET_UNIPERIF_CTRL_ROUNDING_OFF(player); + + /* Set clock divisor */ + SET_UNIPERIF_CTRL_DIVIDER(player, clk_div / 128); + + /* Set the spdif latency to not wait before starting player */ + SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player); + + /* + * Ensure iec958 formatting is off. It will be enabled in function + * uni_player_start() at the same time as the operation + * mode is set to work around a silicon issue. + */ + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player); + else + SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); + + return 0; +} + +static int uni_player_prepare_pcm(struct uniperif *player, + struct snd_pcm_runtime *runtime) +{ + int output_frame_size, slot_width, clk_div; + + /* Force slot width to 32 in I2S mode (HW constraint) */ + if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_I2S) { + slot_width = 32; + } else { + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 16; + break; + default: + slot_width = 32; + break; + } + } + output_frame_size = slot_width * runtime->channels; + + clk_div = player->mclk / runtime->rate; + /* + * For 32 bits subframe clk_div must be a multiple of 128, + * for 16 bits must be a multiple of 64 + */ + if ((slot_width == 32) && (clk_div % 128)) { + dev_err(player->dev, "%s: invalid clk_div", __func__); + return -EINVAL; + } + + if ((slot_width == 16) && (clk_div % 64)) { + dev_err(player->dev, "%s: invalid clk_div", __func__); + return -EINVAL; + } + + /* + * Number of bits per subframe (which is one channel sample) + * on output - Transfer 16 or 32 bits from FIFO + */ + switch (slot_width) { + case 32: + SET_UNIPERIF_I2S_FMT_NBIT_32(player); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player); + break; + case 16: + SET_UNIPERIF_I2S_FMT_NBIT_16(player); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player); + break; + default: + dev_err(player->dev, "subframe format not supported"); + return -EINVAL; + } + + /* Configure data memory format */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + /* One data word contains two samples */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_16(player); + break; + + case SNDRV_PCM_FORMAT_S32_LE: + /* + * Actually "16 bits/0 bits" means "32/28/24/20/18/16 bits + * on the left than zeros (if less than 32 bytes)"... ;-) + */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player); + break; + + default: + dev_err(player->dev, "format not supported"); + return -EINVAL; + } + + /* Set rounding to off */ + SET_UNIPERIF_CTRL_ROUNDING_OFF(player); + + /* Set clock divisor */ + SET_UNIPERIF_CTRL_DIVIDER(player, clk_div / (2 * output_frame_size)); + + /* Number of channelsmust be even*/ + if ((runtime->channels % 2) || (runtime->channels < 2) || + (runtime->channels > 10)) { + dev_err(player->dev, "%s: invalid nb of channels", __func__); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_NUM_CH(player, runtime->channels / 2); + + /* Set 1-bit audio format to disabled */ + SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player); + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player); + + /* No iec958 formatting as outputting to DAC */ + SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player); + + return 0; +} + +static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + if (clk_id != 0) + return -EINVAL; + + player->mclk = freq; + + return clk_set_rate(player->clk, freq); +} + +static int uni_player_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + struct snd_pcm_runtime *runtime = substream->runtime; + int transfer_size, trigger_limit; + int ret; + + /* The player should be stopped */ + if (player->state != UNIPERIF_STATE_STOPPED) { + dev_err(player->dev, "%s: invalid player state %d", __func__, + player->state); + return -EINVAL; + } + + /* Calculate transfer size (in fifo cells and bytes) for frame count */ + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + + /* Calculate number of empty cells available before asserting DREQ */ + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { + trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; + } else { + /* + * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 + * FDMA_TRIGGER_LIMIT also controls when the state switches + * from OFF or STANDBY to AUDIO DATA. + */ + trigger_limit = transfer_size; + } + + /* Trigger limit must be an even number */ + if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) || + (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) { + dev_err(player->dev, "invalid trigger limit %d", trigger_limit); + return -EINVAL; + } + + SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(player, trigger_limit); + + /* Uniperipheral setup depends on player type */ + switch (player->info->player_type) { + case SND_ST_UNIPERIF_PLAYER_TYPE_HDMI: + ret = uni_player_prepare_iec958(player, runtime); + break; + case SND_ST_UNIPERIF_PLAYER_TYPE_PCM: + ret = uni_player_prepare_pcm(player, runtime); + break; + case SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF: + ret = uni_player_prepare_iec958(player, runtime); + break; + default: + dev_err(player->dev, "invalid player type"); + return -EINVAL; + } + + if (ret) + return ret; + + switch (player->daifmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(player); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(player); + break; + case SND_SOC_DAIFMT_NB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(player); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(player); + break; + case SND_SOC_DAIFMT_IB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(player); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player); + break; + case SND_SOC_DAIFMT_IB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(player); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player); + } + + switch (player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player); + SET_UNIPERIF_I2S_FMT_PADDING_I2S_MODE(player); + break; + case SND_SOC_DAIFMT_LEFT_J: + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player); + SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player); + break; + case SND_SOC_DAIFMT_RIGHT_J: + SET_UNIPERIF_I2S_FMT_ALIGN_RIGHT(player); + SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player); + break; + default: + dev_err(player->dev, "format not supported"); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0); + + /* Reset uniperipheral player */ + SET_UNIPERIF_SOFT_RST_SOFT_RST(player); + + return reset_player(player); +} + +static int uni_player_start(struct uniperif *player) +{ + int ret; + + /* The player should be stopped */ + if (player->state != UNIPERIF_STATE_STOPPED) { + dev_err(player->dev, "%s: invalid player state", __func__); + return -EINVAL; + } + + ret = clk_prepare_enable(player->clk); + if (ret) { + dev_err(player->dev, "%s: Failed to enable clock", __func__); + return ret; + } + + /* Clear any pending interrupts */ + SET_UNIPERIF_ITS_BCLR(player, GET_UNIPERIF_ITS(player)); + + /* Set the interrupt mask */ + SET_UNIPERIF_ITM_BSET_DMA_ERROR(player); + SET_UNIPERIF_ITM_BSET_FIFO_ERROR(player); + + /* Enable underflow recovery interrupts */ + if (player->info->underflow_enabled) { + SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_DONE(player); + SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player); + } + + /* Reset uniperipheral player */ + SET_UNIPERIF_SOFT_RST_SOFT_RST(player); + + ret = reset_player(player); + if (ret < 0) + return ret; + + /* + * Does not use IEC61937 features of the uniperipheral hardware. + * Instead it performs IEC61937 in software and inserts it directly + * into the audio data stream. As such, when encoded mode is selected, + * linear pcm mode is still used, but with the differences of the + * channel status bits set for encoded mode and the validity bits set. + */ + SET_UNIPERIF_CTRL_OPERATION_PCM_DATA(player); + + /* + * If iec958 formatting is required for hdmi or spdif, then it must be + * enabled after the operation mode is set. If set prior to this, it + * will not take affect and hang the player. + */ + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) + SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); + + /* Force channel status update (no update if clk disable) */ + if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + SET_UNIPERIF_CONFIG_CHL_STS_UPDATE(player); + else + SET_UNIPERIF_BIT_CONTROL_CHL_STS_UPDATE(player); + + /* Update state to started */ + player->state = UNIPERIF_STATE_STARTED; + + return 0; +} + +static int uni_player_stop(struct uniperif *player) +{ + int ret; + + /* The player should not be in stopped state */ + if (player->state == UNIPERIF_STATE_STOPPED) { + dev_err(player->dev, "%s: invalid player state", __func__); + return -EINVAL; + } + + /* Turn the player off */ + SET_UNIPERIF_CTRL_OPERATION_OFF(player); + + /* Soft reset the player */ + SET_UNIPERIF_SOFT_RST_SOFT_RST(player); + + ret = reset_player(player); + if (ret < 0) + return ret; + + /* Disable interrupts */ + SET_UNIPERIF_ITM_BCLR(player, GET_UNIPERIF_ITM(player)); + + /* Disable clock */ + clk_disable_unprepare(player->clk); + + /* Update state to stopped and return */ + player->state = UNIPERIF_STATE_STOPPED; + + return 0; +} + +int uni_player_resume(struct uniperif *player) +{ + int ret; + + /* Select the frequency synthesizer clock */ + if (player->clk_sel) { + ret = regmap_field_write(player->clk_sel, 1); + if (ret) { + dev_err(player->dev, + "%s: Failed to select freq synth clock", + __func__); + return ret; + } + } + + SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(player); + SET_UNIPERIF_CTRL_ROUNDING_OFF(player); + SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player); + SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player); + + return 0; +} +EXPORT_SYMBOL_GPL(uni_player_resume); + +static int uni_player_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + return uni_player_start(player); + case SNDRV_PCM_TRIGGER_STOP: + return uni_player_stop(player); + case SNDRV_PCM_TRIGGER_RESUME: + return uni_player_resume(player); + default: + return -EINVAL; + } +} + +static void uni_player_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + + if (player->state != UNIPERIF_STATE_STOPPED) + /* Stop the player */ + uni_player_stop(player); +} + +static int uni_player_parse_dt_clk_glue(struct platform_device *pdev, + struct uniperif *player) +{ + int bit_offset; + struct device_node *node = pdev->dev.of_node; + struct regmap *regmap; + + bit_offset = SYS_CFG_AUDI0_GLUE_PCM_CLKX + player->info->id; + + regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg"); + + if (regmap) { + struct reg_field regfield = + REG_FIELD(SYS_CFG_AUDIO_GLUE, bit_offset, bit_offset); + + player->clk_sel = regmap_field_alloc(regmap, regfield); + } else { + dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n"); + return -EINVAL; + } + + return 0; +} + +static int uni_player_parse_dt(struct platform_device *pdev, + struct uniperif *player) +{ + struct uniperif_info *info; + struct device *dev = &pdev->dev; + struct device_node *pnode = pdev->dev.of_node; + const char *mode; + + /* Allocate memory for the info structure */ + info = devm_kzalloc(dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + of_property_read_u32(pnode, "version", &player->ver); + if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + dev_err(dev, "Unknown uniperipheral version "); + return -EINVAL; + } + /* Underflow recovery is only supported on later ip revisions */ + if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + info->underflow_enabled = 1; + + of_property_read_u32(pnode, "uniperiph-id", &info->id); + + /* Read the device mode property */ + of_property_read_string(pnode, "mode", &mode); + + if (strcasecmp(mode, "hdmi") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; + else if (strcasecmp(mode, "pcm") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; + else if (strcasecmp(mode, "spdif") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; + else + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + + /* Save the info structure */ + player->info = info; + + /* Get the PCM_CLK_SEL bit from audio-glue-ctrl SoC register */ + if (uni_player_parse_dt_clk_glue(pdev, player)) + return -EINVAL; + + return 0; +} + +const struct snd_soc_dai_ops uni_player_dai_ops = { + .shutdown = uni_player_shutdown, + .prepare = uni_player_prepare, + .trigger = uni_player_trigger, + .hw_params = sti_uniperiph_dai_hw_params, + .set_fmt = sti_uniperiph_dai_set_fmt, + .set_sysclk = uni_player_set_sysclk +}; + +int uni_player_init(struct platform_device *pdev, + struct uniperif *player) +{ + int ret = 0; + + player->dev = &pdev->dev; + player->state = UNIPERIF_STATE_STOPPED; + player->hw = &uni_player_pcm_hw; + player->dai_ops = &uni_player_dai_ops; + + ret = uni_player_parse_dt(pdev, player); + + if (ret < 0) { + dev_err(player->dev, "Failed to parse DeviceTree"); + return ret; + } + + /* Get uniperif resource */ + player->clk = of_clk_get(pdev->dev.of_node, 0); + if (IS_ERR(player->clk)) + ret = (int)PTR_ERR(player->clk); + + /* Select the frequency synthesizer clock */ + if (player->clk_sel) { + ret = regmap_field_write(player->clk_sel, 1); + if (ret) { + dev_err(player->dev, + "%s: Failed to select freq synth clock", + __func__); + return ret; + } + } + + ret = devm_request_irq(&pdev->dev, player->irq, + uni_player_irq_handler, IRQF_SHARED, + dev_name(&pdev->dev), player); + if (ret < 0) + return ret; + + /* Ensure that disabled by default */ + SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(player); + SET_UNIPERIF_CTRL_ROUNDING_OFF(player); + SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player); + SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player); + + if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) { + /* Set default iec958 status bits */ + + /* Consumer, PCM, copyright, 2ch, mode 0 */ + player->stream_settings.iec958.status[0] = 0x00; + /* Broadcast reception category */ + player->stream_settings.iec958.status[1] = + IEC958_AES1_CON_GENERAL; + /* Do not take into account source or channel number */ + player->stream_settings.iec958.status[2] = + IEC958_AES2_CON_SOURCE_UNSPEC; + /* Sampling frequency not indicated */ + player->stream_settings.iec958.status[3] = + IEC958_AES3_CON_FS_NOTID; + /* Max sample word 24-bit, sample word length not indicated */ + player->stream_settings.iec958.status[4] = + IEC958_AES4_CON_MAX_WORDLEN_24 | + IEC958_AES4_CON_WORDLEN_24_20; + } + + return 0; +} +EXPORT_SYMBOL_GPL(uni_player_init);

Add code to manage Uniperipheral reader IP instances. These DAIs are dedicated to capture and support I2S and IEC modes.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- sound/soc/sti/uniperif.h | 3 + sound/soc/sti/uniperif_reader.c | 364 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 367 insertions(+) create mode 100644 sound/soc/sti/uniperif_reader.c

diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index a8d9e94..f396958 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1212,4 +1212,7 @@ int uni_player_init(struct platform_device *pdev, struct uniperif *uni_player); int uni_player_resume(struct uniperif *player);

+/* uniperiph reader */ +int uni_reader_init(struct platform_device *pdev, + struct uniperif *uni_reader); #endif diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c new file mode 100644 index 0000000..b098c92 --- /dev/null +++ b/sound/soc/sti/uniperif_reader.c @@ -0,0 +1,364 @@ +/* + * Copyright (C) STMicroelectronics SA 2015 + * Authors: Arnaud Pouliquen arnaud.pouliquen@st.com + * for STMicroelectronics. + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> + +#include <sound/soc.h> + +#include "uniperif.h" + +/* + * Note: snd_pcm_hardware is linked to DMA controller but is declared here to + * integrate unireader capability in term of rate and supported channels + */ +const struct snd_pcm_hardware uni_reader_pcm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE, + + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 96000, + + .channels_min = 2, + .channels_max = 8, + + .periods_min = 2, + .periods_max = 48, + + .period_bytes_min = 128, + .period_bytes_max = 64 * PAGE_SIZE, + .buffer_bytes_max = 256 * PAGE_SIZE +}; + +/* + * uni_reader_irq_handler + * In case of error audio stream is stopped; stop action is protected via PCM + * stream lock to avoid race condition with trigger callback. + */ +static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) +{ + irqreturn_t ret = IRQ_NONE; + struct uniperif *reader = dev_id; + unsigned int status; + + if (reader->state == UNIPERIF_STATE_STOPPED) { + /* Unexpected IRQ: do nothing */ + dev_warn(reader->dev, "unexpected IRQ "); + return IRQ_HANDLED; + } + + /* Get interrupt status & clear them immediately */ + status = GET_UNIPERIF_ITS(reader); + SET_UNIPERIF_ITS_BCLR(reader, status); + + /* Check for fifo overflow error */ + if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { + dev_err(reader->dev, "FIFO error detected"); + + snd_pcm_stream_lock(reader->substream); + snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(reader->substream); + + return IRQ_HANDLED; + } + + return ret; +} + +static int uni_reader_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + struct snd_pcm_runtime *runtime = substream->runtime; + int transfer_size, trigger_limit; + int slot_width; + int count = 10; + + /* The reader should be stopped */ + if (reader->state != UNIPERIF_STATE_STOPPED) { + dev_err(reader->dev, "%s: invalid reader state %d", __func__, + reader->state); + return -EINVAL; + } + + /* Calculate transfer size (in fifo cells and bytes) for frame count */ + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + + /* Calculate number of empty cells available before asserting DREQ */ + if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; + else + /* + * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 + * FDMA_TRIGGER_LIMIT also controls when the state switches + * from OFF or STANDBY to AUDIO DATA. + */ + trigger_limit = transfer_size; + + /* Trigger limit must be an even number */ + if ((!trigger_limit % 2) || + (trigger_limit != 1 && transfer_size % 2) || + (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { + dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); + return -EINVAL; + } + + SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit); + + switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_NB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); + break; + default: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); + } + + /* Force slot width to 32 in I2S mode */ + if ((reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_I2S) { + slot_width = 32; + } else { + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 16; + break; + default: + slot_width = 32; + break; + } + } + + /* Number of bits per subframe (i.e one channel sample) on input. */ + switch (slot_width) { + case 32: + SET_UNIPERIF_I2S_FMT_NBIT_32(reader); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(reader); + break; + case 16: + SET_UNIPERIF_I2S_FMT_NBIT_16(reader); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader); + break; + default: + dev_err(reader->dev, "subframe format not supported"); + return -EINVAL; + } + + /* Configure data memory format */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + /* One data word contains two samples */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_16(reader); + break; + + case SNDRV_PCM_FORMAT_S32_LE: + /* + * Actually "16 bits/0 bits" means "32/28/24/20/18/16 bits + * on the MSB then zeros (if less than 32 bytes)"... + */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(reader); + break; + + default: + dev_err(reader->dev, "format not supported"); + return -EINVAL; + } + + switch (reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); + SET_UNIPERIF_I2S_FMT_PADDING_I2S_MODE(reader); + break; + case SND_SOC_DAIFMT_LEFT_J: + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); + SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader); + break; + case SND_SOC_DAIFMT_RIGHT_J: + SET_UNIPERIF_I2S_FMT_ALIGN_RIGHT(reader); + SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader); + break; + default: + dev_err(reader->dev, "format not supported"); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); + + /* Data clocking (changing) on the rising edge */ + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); + + /* Number of channels must be even */ + + if ((runtime->channels % 2) || (runtime->channels < 2) || + (runtime->channels > 10)) { + dev_err(reader->dev, "%s: invalid nb of channels", __func__); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2); + + /* Clear any pending interrupts */ + SET_UNIPERIF_ITS_BCLR(reader, GET_UNIPERIF_ITS(reader)); + + SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(reader, 0); + + /* Set the interrupt mask */ + SET_UNIPERIF_ITM_BSET_DMA_ERROR(reader); + SET_UNIPERIF_ITM_BSET_FIFO_ERROR(reader); + SET_UNIPERIF_ITM_BSET_MEM_BLK_READ(reader); + + /* Enable underflow recovery interrupts */ + if (reader->info->underflow_enabled) { + SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_DONE(reader); + SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(reader); + } + + /* Reset uniperipheral reader */ + SET_UNIPERIF_SOFT_RST_SOFT_RST(reader); + + while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) { + udelay(5); + count--; + } + if (!count) { + dev_err(reader->dev, "Failed to reset uniperif"); + return -EIO; + } + + return 0; +} + +static int uni_reader_start(struct uniperif *reader) +{ + /* The reader should be stopped */ + if (reader->state != UNIPERIF_STATE_STOPPED) { + dev_err(reader->dev, "%s: invalid reader state", __func__); + return -EINVAL; + } + + /* Enable reader interrupts (and clear possible stalled ones) */ + enable_irq(reader->irq); + SET_UNIPERIF_ITS_BCLR_FIFO_ERROR(reader); + SET_UNIPERIF_ITM_BSET_FIFO_ERROR(reader); + + /* Launch the reader */ + SET_UNIPERIF_CTRL_OPERATION_PCM_DATA(reader); + + /* Update state to started */ + reader->state = UNIPERIF_STATE_STARTED; + return 0; +} + +static int uni_reader_stop(struct uniperif *reader) +{ + /* The reader should not be in stopped state */ + if (reader->state == UNIPERIF_STATE_STOPPED) { + dev_err(reader->dev, "%s: invalid reader state", __func__); + return -EINVAL; + } + + /* Turn the reader off */ + SET_UNIPERIF_CTRL_OPERATION_OFF(reader); + + /* Disable interrupts */ + SET_UNIPERIF_ITM_BCLR(reader, GET_UNIPERIF_ITM(reader)); + disable_irq_nosync(reader->irq); + + /* Update state to stopped and return */ + reader->state = UNIPERIF_STATE_STOPPED; + + return 0; +} + +static int uni_reader_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + return uni_reader_start(reader); + case SNDRV_PCM_TRIGGER_STOP: + return uni_reader_stop(reader); + default: + return -EINVAL; + } +} + +static void uni_reader_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + + if (reader->state != UNIPERIF_STATE_STOPPED) { + /* Stop the reader */ + uni_reader_stop(reader); + } +} + +static int uni_reader_parse_dt(struct platform_device *pdev, + struct uniperif *reader) +{ + struct uniperif_info *info; + struct device_node *node = pdev->dev.of_node; + + /* Allocate memory for the info structure */ + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + of_property_read_u32(node, "version", &reader->ver); + + /* Save the info structure */ + reader->info = info; + + return 0; +} + +const struct snd_soc_dai_ops uni_reader_dai_ops = { + .shutdown = uni_reader_shutdown, + .prepare = uni_reader_prepare, + .trigger = uni_reader_trigger, + .hw_params = sti_uniperiph_dai_hw_params, + .set_fmt = sti_uniperiph_dai_set_fmt, +}; + +int uni_reader_init(struct platform_device *pdev, + struct uniperif *reader) +{ + int ret = 0; + + reader->dev = &pdev->dev; + reader->state = UNIPERIF_STATE_STOPPED; + reader->hw = &uni_reader_pcm_hw; + reader->dai_ops = &uni_reader_dai_ops; + + dev_err(reader->dev, "%s: enter\n", __func__); + ret = uni_reader_parse_dt(pdev, reader); + if (ret < 0) { + dev_err(reader->dev, "Failed to parse DeviceTree"); + return ret; + } + + ret = devm_request_irq(&pdev->dev, reader->irq, + uni_reader_irq_handler, IRQF_SHARED, + dev_name(&pdev->dev), reader); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to request IRQ"); + return -EBUSY; + } + + return 0; +} +EXPORT_SYMBOL_GPL(uni_reader_init);

Define the platform and codec drivers, and how to build them.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sti/Kconfig | 11 +++++++++++ sound/soc/sti/Makefile | 4 ++++ 4 files changed, 17 insertions(+) create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3ba52da..791f2a6 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -53,6 +53,7 @@ source "sound/soc/samsung/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" +source "sound/soc/sti/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 974ba70..30b2d6d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -34,6 +34,7 @@ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ +obj-$(CONFIG_SND_SOC) += sti/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/sti/Kconfig b/sound/soc/sti/Kconfig new file mode 100644 index 0000000..64a6900 --- /dev/null +++ b/sound/soc/sti/Kconfig @@ -0,0 +1,11 @@ +# +# STM SoC audio configuration +# +menuconfig SND_SOC_STI + tristate "SoC Audio support for STI System-On-Chip" + depends on SND_SOC + depends on ARCH_STI || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y if you want to enable ASoC-support for + any of the STI platforms (e.g. STIH416). diff --git a/sound/soc/sti/Makefile b/sound/soc/sti/Makefile new file mode 100644 index 0000000..4b188d2 --- /dev/null +++ b/sound/soc/sti/Makefile @@ -0,0 +1,4 @@ +# STI platform support +snd-soc-sti-objs := sti_uniperif.o uniperif_player.o uniperif_reader.o + +obj-$(CONFIG_SND_SOC_STI) += snd-soc-sti.o

Codec part of the sti platform that supports codec IPs. This first version does not support HDMI, but only DAC and SPDIF out.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 627 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 634 insertions(+) create mode 100644 sound/soc/codecs/sti-sas.c

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 186fe7f..7434a20 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -102,6 +102,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -604,6 +605,10 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate

+config SND_SOC_STI_SAS + tristate "codec Audio support for STI SAS codec" + depends on SND_SOC_STI + config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3dcf5ac..1744e22 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -105,6 +105,7 @@ snd-soc-sta32x-objs := sta32x.o snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o snd-soc-tfa9879-objs := tfa9879.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c new file mode 100644 index 0000000..32db2c2 --- /dev/null +++ b/sound/soc/codecs/sti-sas.c @@ -0,0 +1,627 @@ +/* + * Copyright (C) STMicroelectronics SA 2015 + * Authors: Arnaud Pouliquen arnaud.pouliquen@st.com + * for STMicroelectronics. + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/io.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/reset.h> +#include <linux/mfd/syscon.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +/* chipID supported */ +#define CHIPID_STIH416 0 +#define CHIPID_STIH407 1 + +/* DAC definitions */ + +/* stih416 DAC registers */ +/* sysconf 2517: Audio-DAC-Control */ +#define STIH416_AUDIO_DAC_CTRL 0x00000814 +/* sysconf 2519: Audio-Gue-Control */ +#define STIH416_AUDIO_GLUE_CTRL 0x0000081C + +#define STIH416_DAC_NOT_STANDBY 0x3 +#define STIH416_DAC_SOFTMUTE 0x4 +#define STIH416_DAC_ANA_NOT_PWR 0x5 +#define STIH416_DAC_NOT_PNDBG 0x6 + +#define STIH416_DAC_NOT_STANDBY_MASK BIT(STIH416_DAC_NOT_STANDBY) +#define STIH416_DAC_SOFTMUTE_MASK BIT(STIH416_DAC_SOFTMUTE) +#define STIH416_DAC_ANA_NOT_PWR_MASK BIT(STIH416_DAC_ANA_NOT_PWR) +#define STIH416_DAC_NOT_PNDBG_MASK BIT(STIH416_DAC_NOT_PNDBG) + +/* stih407 DAC registers */ +/* sysconf 5041: Audio-Gue-Control */ +#define STIH407_AUDIO_GLUE_CTRL 0x000000A4 +/* sysconf 5042: Audio-DAC-Control */ +#define STIH407_AUDIO_DAC_CTRL 0x000000A8 + +/* DAC definitions */ +#define STIH407_DAC_SOFTMUTE 0x0 +#define STIH407_DAC_STANDBY_ANA 0x1 +#define STIH407_DAC_STANDBY 0x2 + +#define STIH407_DAC_SOFTMUTE_MASK BIT(STIH407_DAC_SOFTMUTE) +#define STIH407_DAC_STANDBY_ANA_MASK BIT(STIH407_DAC_STANDBY_ANA) +#define STIH407_DAC_STANDBY_MASK BIT(STIH407_DAC_STANDBY) + +/* SPDIF definitions */ +#define SPDIF_BIPHASE_ENABLE 0x6 +#define SPDIF_BIPHASE_IDLE 0x7 + +#define SPDIF_BIPHASE_ENABLE_MASK BIT(SPDIF_BIPHASE_ENABLE) +#define SPDIF_BIPHASE_IDLE_MASK BIT(SPDIF_BIPHASE_IDLE) + +enum { + STI_SAS_DAI_SPDIF_OUT, + STI_SAS_DAI_ANALOG_OUT, +}; + +static const struct reg_default stih416_sas_reg_defaults[] = { + { STIH407_AUDIO_GLUE_CTRL, 0x00000040 }, + { STIH407_AUDIO_DAC_CTRL, 0x000000000 }, +}; + +static const struct reg_default stih407_sas_reg_defaults[] = { + { STIH416_AUDIO_DAC_CTRL, 0x000000000 }, + { STIH416_AUDIO_GLUE_CTRL, 0x00000040 }, +}; + +struct sti_dac_audio { + struct regmap *regmap; + struct regmap *virt_regmap; + struct regmap_field **field; + struct reset_control *rst; + int mclk; +}; + +struct sti_spdif_audio { + struct regmap *regmap; + struct regmap_field **field; + int mclk; +}; + +/* device data structure */ +struct sti_sas_dev_data { + const int chipid; /* IC version */ + const struct regmap_config *regmap; + const struct snd_soc_dai_ops *dac_ops; /* DAC function callbacks */ + const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */ + const int num_dapm_widgets; /* dapms declaration */ + const struct snd_soc_dapm_route *dapm_routes; /* route declaration */ + const int num_dapm_routes; /* route declaration */ +}; + +/* driver data structure */ +struct sti_sas_data { + struct device *dev; + const struct sti_sas_dev_data *dev_data; + struct sti_dac_audio dac; + struct sti_spdif_audio spdif; +}; + +/* Read a register from the sysconf reg bank */ +static int sti_sas_read_reg(void *context, unsigned int reg, + unsigned int *value) +{ + struct sti_sas_data *drvdata = context; + int status; + u32 val; + + status = regmap_read(drvdata->dac.regmap, reg, &val); + *value = (unsigned int)val; + + return status; +} + +/* Read a register from the sysconf reg bank */ +static int sti_sas_write_reg(void *context, unsigned int reg, + unsigned int value) +{ + struct sti_sas_data *drvdata = context; + int status; + + status = regmap_write(drvdata->dac.regmap, reg, value); + + return status; +} + +static int sti_sas_init_sas_registers(struct snd_soc_codec *codec, + struct sti_sas_data *data) +{ + int ret; + /* + * DAC and SPDIF are activated by default + * put them in IDLE to save power + */ + + /* Initialise bi-phase formatter to disabled */ + ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL, + SPDIF_BIPHASE_ENABLE_MASK, 0); + + if (!ret) + /* Initialise bi-phase formatter idle value to 0 */ + ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL, + SPDIF_BIPHASE_IDLE_MASK, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to update SPDIF registers"); + return ret; + } + + /* Init DAC configuration */ + switch (data->dev_data->chipid) { + case CHIPID_STIH407: + /* init configuration */ + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY_MASK, + STIH407_DAC_STANDBY_MASK); + + if (!ret) + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY_ANA_MASK, + STIH407_DAC_STANDBY_ANA_MASK); + if (!ret) + ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_SOFTMUTE_MASK, + STIH407_DAC_SOFTMUTE_MASK); + break; + case CHIPID_STIH416: + ret = snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_NOT_STANDBY_MASK, 0); + if (!ret) + ret = snd_soc_update_bits(codec, + STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_ANA_NOT_PWR, 0); + if (!ret) + ret = snd_soc_update_bits(codec, + STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_NOT_PNDBG_MASK, + 0); + if (!ret) + ret = snd_soc_update_bits(codec, + STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_SOFTMUTE_MASK, + STIH416_DAC_SOFTMUTE_MASK); + break; + default: + return -EINVAL; + } + + if (ret < 0) { + dev_err(codec->dev, "Failed to update DAC registers"); + return ret; + } + + return ret; +} + +/* + * DAC + */ +static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + /* Sanity check only */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(dai->codec->dev, + "%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + return 0; +} + +static int stih416_dac_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); + struct sti_dac_audio *dac = &drvdata->dac; + + /* Get reset control */ + dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); + if (IS_ERR(dac->rst)) { + dev_err(dai->codec->dev, + "%s: ERROR: DAC reset control not defined (%d)!\n", + __func__, (int)dac->rst); + dac->rst = NULL; + return -EFAULT; + } + /* Put the DAC into reset */ + reset_control_assert(dac->rst); + + return 0; +} + +const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = { + SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC standby", "dac_p", STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_NOT_STANDBY, 0), + SND_SOC_DAPM_OUTPUT("DAC Output"), +}; + +const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY_ANA, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC standby", "dac_p", STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_STANDBY, 1), + SND_SOC_DAPM_OUTPUT("DAC Output"), +}; + +const struct snd_soc_dapm_route stih416_sas_route[] = { + {"DAC Output", NULL, "DAC bandgap"}, + {"DAC Output", NULL, "DAC standby ana"}, + {"DAC standby ana", NULL, "DAC standby"}, +}; + +const struct snd_soc_dapm_route stih407_sas_route[] = { + {"DAC Output", NULL, "DAC standby ana"}, + {"DAC standby ana", NULL, "DAC standby"}, +}; + +static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_SOFTMUTE_MASK, + STIH416_DAC_SOFTMUTE_MASK); + } else { + return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL, + STIH416_DAC_SOFTMUTE_MASK, 0); + } +} + +static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + return snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_SOFTMUTE_MASK, + STIH407_DAC_SOFTMUTE_MASK); + } else { + return snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL, + STIH407_DAC_SOFTMUTE_MASK, + 0); + } +} + +/* + * SPDIF + */ +static int sti_sas_spdif_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(dai->codec->dev, + "%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + return 0; +} + +/* + * sti_sas_spdif_trigger: + * Trigger function is used to ensure that BiPhase Formater is disabled + * before CPU dai is stopped. + * This is mandatory to avoid that BPF is stalled + */ +static int sti_sas_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + return snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL, + SPDIF_BIPHASE_ENABLE_MASK, + SPDIF_BIPHASE_ENABLE_MASK); + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + return snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL, + SPDIF_BIPHASE_ENABLE_MASK, + 0); + default: + return -EINVAL; + } +} + +static bool sti_sas_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == STIH407_AUDIO_GLUE_CTRL) + return true; + + return false; +} + +/* + * CODEC DAIS + */ + +/* + * sti_sas_set_sysclk: + * get MCLK input frequency to check that MCLK-FS ratio is coherent + */ +static int sti_sas_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); + + if (dir == SND_SOC_CLOCK_OUT) + return 0; + + if (clk_id != 0) + return -EINVAL; + + switch (dai->id) { + case STI_SAS_DAI_SPDIF_OUT: + drvdata->spdif.mclk = freq; + break; + + case STI_SAS_DAI_ANALOG_OUT: + drvdata->dac.mclk = freq; + break; + } + + return 0; +} + +static int sti_sas_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); + struct snd_pcm_runtime *runtime = substream->runtime; + + switch (dai->id) { + case STI_SAS_DAI_SPDIF_OUT: + if ((drvdata->spdif.mclk / runtime->rate) != 128) { + dev_err(codec->dev, "unexpected mclk-fs ratio"); + return -EINVAL; + } + break; + case STI_SAS_DAI_ANALOG_OUT: + if ((drvdata->dac.mclk / runtime->rate) != 256) { + dev_err(codec->dev, "unexpected mclk-fs ratio"); + return -EINVAL; + } + break; + } + + return 0; +} + +const struct snd_soc_dai_ops stih416_dac_ops = { + .set_fmt = sti_sas_dac_set_fmt, + .mute_stream = stih416_sas_dac_mute, + .prepare = sti_sas_prepare, + .set_sysclk = sti_sas_set_sysclk, +}; + +const struct snd_soc_dai_ops stih407_dac_ops = { + .set_fmt = sti_sas_dac_set_fmt, + .mute_stream = stih407_sas_dac_mute, + .prepare = sti_sas_prepare, + .set_sysclk = sti_sas_set_sysclk, +}; + +const struct regmap_config stih407_sas_regmap = { + .reg_bits = 32, + .val_bits = 32, + + .max_register = STIH407_AUDIO_DAC_CTRL, + .reg_defaults = stih407_sas_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(stih407_sas_reg_defaults), + .volatile_reg = sti_sas_volatile_register, + .cache_type = REGCACHE_RBTREE, + .reg_read = sti_sas_read_reg, + .reg_write = sti_sas_write_reg, +}; + +const struct regmap_config stih416_sas_regmap = { + .reg_bits = 32, + .val_bits = 32, + + .max_register = STIH416_AUDIO_DAC_CTRL, + .reg_defaults = stih416_sas_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults), + .volatile_reg = sti_sas_volatile_register, + .cache_type = REGCACHE_RBTREE, + .reg_read = sti_sas_read_reg, + .reg_write = sti_sas_write_reg, +}; + +const struct sti_sas_dev_data stih416_data = { + .chipid = CHIPID_STIH416, + .regmap = &stih416_sas_regmap, + .dac_ops = &stih416_dac_ops, + .dapm_widgets = stih416_sas_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets), + .dapm_routes = stih416_sas_route, + .num_dapm_routes = ARRAY_SIZE(stih416_sas_route), +}; + +const struct sti_sas_dev_data stih407_data = { + .chipid = CHIPID_STIH407, + .regmap = &stih407_sas_regmap, + .dac_ops = &stih407_dac_ops, + .dapm_widgets = stih407_sas_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(stih407_sas_dapm_widgets), + .dapm_routes = stih407_sas_route, + .num_dapm_routes = ARRAY_SIZE(stih407_sas_route), +}; + +static struct snd_soc_dai_driver sti_sas_dai[] = { + { + .name = "sas-dai-spdif-out", + .id = STI_SAS_DAI_SPDIF_OUT, + .playback = { + .stream_name = "spdif_p", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_fmt = sti_sas_spdif_set_fmt, + .trigger = sti_sas_spdif_trigger, + .set_sysclk = sti_sas_set_sysclk, + .prepare = sti_sas_prepare, + } + }, + }, + { + .name = "sas-dai-dac", + .id = STI_SAS_DAI_ANALOG_OUT, + .playback = { + .stream_name = "dac_p", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + }, +}; + +#ifdef CONFIG_PM_SLEEP +static int sti_sas_resume(struct snd_soc_codec *codec) +{ + struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); + + return sti_sas_init_sas_registers(codec, drvdata); +} +#else +#define sti_sas_resume NULL +#endif + +static int sti_sas_codec_probe(struct snd_soc_codec *codec) +{ + struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); + int ret; + + ret = sti_sas_init_sas_registers(codec, drvdata); + + return ret; +} + +static struct snd_soc_codec_driver sti_sas_driver = { + .probe = sti_sas_codec_probe, + .resume = sti_sas_resume, +}; + +static const struct of_device_id sti_sas_dev_match[] = { + { + .compatible = "st,stih416-sas-codec", + .data = &stih416_data, + }, + { + .compatible = "st,stih407-sas-codec", + .data = &stih407_data, + }, + {}, +}; + +static int sti_sas_driver_probe(struct platform_device *pdev) +{ + struct device_node *pnode = pdev->dev.of_node; + struct sti_sas_data *drvdata; + + /* Allocate device structure */ + drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data), + GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + + /* Populate data structure depending on compatibility */ + if (!of_match_node(sti_sas_dev_match, pnode)->data) { + dev_err(&pdev->dev, "data associated to device is missing"); + return -EINVAL; + } + + drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; + + /* Initialise device structure */ + drvdata->dev = &pdev->dev; + + /* Request the DAC & SPDIF registers memory region */ + drvdata->dac.virt_regmap = devm_regmap_init(&pdev->dev, NULL, drvdata, + drvdata->dev_data->regmap); + if (!drvdata->dac.virt_regmap) { + dev_err(&pdev->dev, "audio registers not enabled\n"); + return -EFAULT; + } + + /* Request the syscon region */ + drvdata->dac.regmap = + syscon_regmap_lookup_by_phandle(pnode, "st,syscfg"); + if (!drvdata->dac.regmap) { + dev_err(&pdev->dev, "syscon registers not available\n"); + return -EFAULT; + } + drvdata->spdif.regmap = drvdata->dac.regmap; + + /* Set DAC dai probe */ + if (drvdata->dev_data->chipid == CHIPID_STIH416) + sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe; + + sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops; + + /* Set dapms*/ + sti_sas_driver.dapm_widgets = drvdata->dev_data->dapm_widgets; + sti_sas_driver.num_dapm_widgets = drvdata->dev_data->num_dapm_widgets; + + sti_sas_driver.dapm_routes = drvdata->dev_data->dapm_routes; + sti_sas_driver.num_dapm_routes = drvdata->dev_data->num_dapm_routes; + + /* Store context */ + dev_set_drvdata(&pdev->dev, drvdata); + + return snd_soc_register_codec(&pdev->dev, &sti_sas_driver, + sti_sas_dai, + ARRAY_SIZE(sti_sas_dai)); +} + +static int sti_sas_driver_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver sti_sas_platform_driver = { + .driver = { + .name = "sti-sas-codec", + .owner = THIS_MODULE, + .of_match_table = sti_sas_dev_match, + }, + .probe = sti_sas_driver_probe, + .remove = sti_sas_driver_remove, +}; + +module_platform_driver(sti_sas_platform_driver); + +MODULE_DESCRIPTION("audio codec for STMicroelectronics sti platforms"); +MODULE_AUTHOR("Arnaud.pouliquen@st.com"); +MODULE_LICENSE("GPL v2");

Add capability to adjust player clock, for clocks drift management.

Signed-off-by: Arnaud Pouliquen arnaud.pouliquen@st.com --- sound/soc/sti/uniperif.h | 2 + sound/soc/sti/uniperif_player.c | 152 +++++++++++++++++++++++++++++++++++++++- 2 files changed, 153 insertions(+), 1 deletion(-)

diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 9c4f75c..210f115 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1175,6 +1175,7 @@ struct uniperif { /* Clocks */ struct clk *clk; int mclk; + int clk_adj;

/* Runtime data */ enum uniperif_state state; @@ -1183,6 +1184,7 @@ struct uniperif {

/* Specific to IEC958 player */ struct uniperif_iec958_settings stream_settings; + struct mutex ctrl_lock; /* For resource updated by stream and controls*/

/*alsa ctrl*/ struct snd_kcontrol_new *snd_ctrls; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d66d633..a1b0135 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -36,6 +36,9 @@ (UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \ UNIPERIF_PLAYER_TYPE_IS_SPDIF(p))

+#define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999 +#define UNIPERIF_PLAYER_CLK_ADJ_MAX 1000000 + /* * Note: snd_pcm_hardware is linked to DMA controller but is declared here to * integrate DAI_CPU capability in term of rate and supported channels @@ -172,6 +175,70 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) return ret; }

+int uni_player_clk_set_rate(struct uniperif *player, unsigned long rate) +{ + int rate_adjusted, rate_achieved, delta, ret; + int adjustment = player->clk_adj; + + /* + * a + * F = f + --------- * f = f + d + * 1000000 + * + * a + * d = --------- * f + * 1000000 + * + * where: + * f - nominal rate + * a - adjustment in ppm (parts per milion) + * F - rate to be set in synthesizer + * d - delta (difference) between f and F + */ + if (adjustment < 0) { + /* div64_64 operates on unsigned values... */ + delta = -1; + adjustment = -adjustment; + } else { + delta = 1; + } + /* 500000 ppm is 0.5, which is used to round up values */ + delta *= (int)div64_u64((uint64_t)rate * + (uint64_t)adjustment + 500000, 1000000); + rate_adjusted = rate + delta; + + /* Adjusted rate should never be == 0 */ + if (!rate_adjusted) + return -EINVAL; + + ret = clk_set_rate(player->clk, rate_adjusted); + if (ret < 0) + return ret; + + rate_achieved = clk_get_rate(player->clk); + if (!rate_achieved) + /* If value is 0 means that clock or parent not valid */ + return -EINVAL; + + /* + * Using ALSA's adjustment control, we can modify the rate to be up + * to twice as much as requested, but no more + */ + delta = rate_achieved - rate; + if (delta < 0) { + /* div64_64 operates on unsigned values... */ + delta = -delta; + adjustment = -1; + } else { + adjustment = 1; + } + /* Frequency/2 is added to round up result */ + adjustment *= (int)div64_u64((uint64_t)delta * 1000000 + rate / 2, + rate); + player->clk_adj = adjustment; + return 0; +} + static void uni_player_set_channel_status(struct uniperif *player, struct snd_pcm_runtime *runtime) { @@ -470,11 +537,84 @@ static int uni_player_prepare_pcm(struct uniperif *player, return 0; }

+/* + * uniperif rate adjustement control + */ +static int snd_sti_clk_adjustment_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = UNIPERIF_PLAYER_CLK_ADJ_MIN; + uinfo->value.integer.max = UNIPERIF_PLAYER_CLK_ADJ_MAX; + uinfo->value.integer.step = 1; + + return 0; +} + +static int snd_sti_clk_adjustment_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + + ucontrol->value.integer.value[0] = player->clk_adj; + + return 0; +} + +static int snd_sti_clk_adjustment_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + int ret = 0; + + if ((ucontrol->value.integer.value[0] < UNIPERIF_PLAYER_CLK_ADJ_MIN) || + (ucontrol->value.integer.value[0] > UNIPERIF_PLAYER_CLK_ADJ_MAX)) + return -EINVAL; + + mutex_lock(&player->ctrl_lock); + player->clk_adj = ucontrol->value.integer.value[0]; + + if (player->mclk) + ret = uni_player_clk_set_rate(player, player->mclk); + mutex_unlock(&player->ctrl_lock); + + return ret; +} + +static struct snd_kcontrol_new uni_player_clk_adj_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Playback Oversampling Freq. Adjustment", + .info = snd_sti_clk_adjustment_info, + .get = snd_sti_clk_adjustment_get, + .put = snd_sti_clk_adjustment_put, +}; + +static struct snd_kcontrol_new *snd_sti_ctl[] = { + &uni_player_clk_adj_ctl, +}; + +static int uni_player_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *player = priv->dai_data.uni; + + player->clk_adj = 0; + + return 0; +} + static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + int ret;

if (dir == SND_SOC_CLOCK_IN) return 0; @@ -484,7 +624,11 @@ static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id,

player->mclk = freq;

- return clk_set_rate(player->clk, freq); + mutex_lock(&player->ctrl_lock); + ret = uni_player_clk_set_rate(player, freq); + mutex_unlock(&player->ctrl_lock); + + return ret; }

static int uni_player_prepare(struct snd_pcm_substream *substream, @@ -812,6 +956,7 @@ static int uni_player_parse_dt(struct platform_device *pdev, }

const struct snd_soc_dai_ops uni_player_dai_ops = { + .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, .trigger = uni_player_trigger, @@ -859,6 +1004,8 @@ int uni_player_init(struct platform_device *pdev, if (ret < 0) return ret;

+ mutex_init(&player->ctrl_lock); + /* Ensure that disabled by default */ SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(player); SET_UNIPERIF_CTRL_ROUNDING_OFF(player); @@ -885,6 +1032,9 @@ int uni_player_init(struct platform_device *pdev, IEC958_AES4_CON_WORDLEN_24_20; }

+ player->num_ctrls = ARRAY_SIZE(snd_sti_ctl); + player->snd_ctrls = snd_sti_ctl[0]; + return 0; } EXPORT_SYMBOL_GPL(uni_player_init);