diff options
| author | Linaro CI <ci_notify@linaro.org> | 2020-07-17 22:30:31 +0000 |
|---|---|---|
| committer | Linaro CI <ci_notify@linaro.org> | 2020-07-17 22:30:31 +0000 |
| commit | 8e232e6850e07e01e6f88806620c96fa5571bf3e (patch) | |
| tree | 7220964dbd1f6e99d3a481dd413a25765f72190f | |
| parent | d225b13dd5ae0b3c118ee9260c8d0767153edb1b (diff) | |
| parent | b13b91b4b0c241162a87c8eaf99ea8a550e18f7d (diff) | |
Merge remote-tracking branch 'audio/tracking-qcomlt-audio' into integration-linux-qcomlt
25 files changed, 1733 insertions, 213 deletions
diff --git a/Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt b/Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt index e94a2ad3a710..1c615b622d07 100644 --- a/Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt +++ b/Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt @@ -14,6 +14,7 @@ Please refer to slimbus/bus.txt for details of the common SLIMBus bindings. must be one of the following. "qcom,slim-ngd-v1.5.0" for MSM8996 "qcom,slim-ngd-v2.1.0" for SDM845 + "qcom,slim-ngd-v2.2.0" for SM8250 - reg: Usage: required diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst index ad4bfbdacc83..935f325dbc77 100644 --- a/Documentation/sound/designs/compress-offload.rst +++ b/Documentation/sound/designs/compress-offload.rst @@ -151,6 +151,57 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +State Machine +============= + +The compressed audio stream state machine is described below :: + + +----------+ + | | + | OPEN | + | | + +----------+ + | + | + | compr_set_params() + | + v + compr_free() +----------+ + +------------------------------------| | + | | SETUP | + | +-------------------------| |<-------------------------+ + | | compr_write() +----------+ | + | | ^ | + | | | compr_drain_notify() | + | | | or | + | | | compr_stop() | + | | | | + | | +----------+ | + | | | | | + | | | DRAIN | | + | | | | | + | | +----------+ | + | | ^ | + | | | | + | | | compr_drain() | + | | | | + | v | | + | +----------+ +----------+ | + | | | compr_start() | | compr_stop() | + | | PREPARE |------------------->| RUNNING |--------------------------+ + | | | | | | + | +----------+ +----------+ | + | | | ^ | + | |compr_free() | | | + | | compr_pause() | | compr_resume() | + | | | | | + | v v | | + | +----------+ +----------+ | + | | | | | compr_stop() | + +--->| FREE | | PAUSE |---------------------------+ + | | | | + +----------+ +----------+ + Gapless Playback ================ @@ -199,6 +250,38 @@ Sequence flow for gapless would be: (note: order for partial_drain and write for next track can be reversed as well) +Gapless Playback SM +=================== + +For Gapless, we move from running state to partial drain and back, along +with setting of meta_data and signalling for next track :: + + + +----------+ + compr_drain_notify() | | + +------------------------>| RUNNING | + | | | + | +----------+ + | | + | | + | | compr_next_track() + | | + | V + | +----------+ + | | | + | |NEXT_TRACK| + | | | + | +----------+ + | | + | | + | | compr_partial_drain() + | | + | V + | +----------+ + | | | + +------------------------ | PARTIAL_ | + | DRAIN | + +----------+ Not supported ============= diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 743ee7b4e63f..e643338678a6 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1323,6 +1323,9 @@ static const struct of_device_id qcom_slim_ngd_dt_match[] = { },{ .compatible = "qcom,slim-ngd-v2.1.0", .data = &ngd_v1_5_offset_info, + },{ + .compatible = "qcom,slim-ngd-v2.2.0", + .data = &ngd_v1_5_offset_info, }, {} }; diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index fa2b4ab92ed9..d121cf739090 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL config SOUNDWIRE_QCOM tristate "Qualcomm SoundWire Master driver" - depends on SLIMBUS depends on SND_SOC help SoundWire Qualcomm Master driver. diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index a1c2a44a3b4d..58ffb46e0d64 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -34,6 +34,7 @@ #define SWRM_INTERRUPT_STATUS_SPECIAL_CMD_ID_FINISHED BIT(10) #define SWRM_INTERRUPT_MASK_ADDR 0x204 #define SWRM_INTERRUPT_CLEAR 0x208 +#define SWRM_INTERRUPT_CPU_EN 0x210 #define SWRM_CMD_FIFO_WR_CMD 0x300 #define SWRM_CMD_FIFO_RD_CMD 0x304 #define SWRM_CMD_FIFO_CMD 0x308 @@ -90,6 +91,7 @@ struct qcom_swrm_ctrl { struct sdw_bus bus; struct device *dev; struct regmap *regmap; + void __iomem *mmio; struct completion *comp; struct work_struct slave_work; /* read/write lock */ @@ -114,7 +116,7 @@ struct qcom_swrm_ctrl { #define to_qcom_sdw(b) container_of(b, struct qcom_swrm_ctrl, bus) -static int qcom_swrm_abh_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, +static int qcom_swrm_ahb_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val) { struct regmap *wcd_regmap = ctrl->regmap; @@ -154,6 +156,20 @@ static int qcom_swrm_ahb_reg_write(struct qcom_swrm_ctrl *ctrl, return SDW_CMD_OK; } +static int qcom_swrm_cpu_reg_read(struct qcom_swrm_ctrl *ctrl, int reg, + u32 *val) +{ + *val = readl(ctrl->mmio + reg); + return SDW_CMD_OK; +} + +static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int reg, + int val) +{ + writel(val, ctrl->mmio + reg); + return SDW_CMD_OK; +} + static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *ctrl, u8 cmd_data, u8 dev_addr, u16 reg_addr) { @@ -310,6 +326,12 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) ctrl->reg_write(ctrl, SWRM_COMP_CFG_ADDR, SWRM_COMP_CFG_IRQ_LEVEL_OR_PULSE_MSK | SWRM_COMP_CFG_ENABLE_MSK); + + /* enable CPU IRQs */ + if (ctrl->mmio) { + ctrl->reg_write(ctrl, SWRM_INTERRUPT_CPU_EN, + SWRM_INTERRUPT_STATUS_RMSK); + } return 0; } @@ -746,6 +768,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) struct sdw_master_prop *prop; struct sdw_bus_params *params; struct qcom_swrm_ctrl *ctrl; + struct resource *res; int ret; u32 val; @@ -753,15 +776,25 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (!ctrl) return -ENOMEM; +#ifdef CONFIG_SLIMBUS if (dev->parent->bus == &slimbus_bus) { - ctrl->reg_read = qcom_swrm_abh_reg_read; +#else + if (false) { +#endif + ctrl->reg_read = qcom_swrm_ahb_reg_read; ctrl->reg_write = qcom_swrm_ahb_reg_write; ctrl->regmap = dev_get_regmap(dev->parent, NULL); if (!ctrl->regmap) return -EINVAL; } else { - /* Only WCD based SoundWire controller is supported */ - return -ENOTSUPP; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + ctrl->reg_read = qcom_swrm_cpu_reg_read; + ctrl->reg_write = qcom_swrm_cpu_reg_write; + ctrl->mmio = devm_ioremap_resource(dev, res); + if (IS_ERR(ctrl->mmio)) + return PTR_ERR(ctrl->mmio); } ctrl->irq = of_irq_get(dev->of_node, 0); @@ -859,6 +892,7 @@ static int qcom_swrm_remove(struct platform_device *pdev) static const struct of_device_id qcom_swrm_of_match[] = { { .compatible = "qcom,soundwire-v1.3.0", }, + { .compatible = "qcom,soundwire-v1.5.1", }, {/* sentinel */}, }; diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index a9a72574b34a..cfdc5b95b63d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -703,9 +703,12 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) } if (!multi_link) { - kfree(wr_msg); - kfree(wbuf); - bus->defer_msg.msg = NULL; + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + bus->defer_msg.msg = NULL; + } + bus->params.curr_bank = !bus->params.curr_bank; bus->params.next_bank = !bus->params.next_bank; } @@ -715,7 +718,11 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) error: kfree(wbuf); error_1: - kfree(wr_msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg); + bus->defer_msg.msg = NULL; + } + return ret; } @@ -748,6 +755,7 @@ static int sdw_ml_sync_bank_switch(struct sdw_bus *bus) if (bus->defer_msg.msg) { kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); + bus->defer_msg.msg = NULL; } return 0; @@ -839,9 +847,11 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + bus->defer_msg.msg = NULL; + } } msg_unlock: diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index 1df06f8ad5c3..7207ab2b57bf 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -107,6 +107,28 @@ #define QUINARY_TDM_RX_7 102 #define QUINARY_TDM_TX_7 103 #define DISPLAY_PORT_RX 104 +#define WSA_CODEC_DMA_RX_0 105 +#define WSA_CODEC_DMA_TX_0 106 +#define WSA_CODEC_DMA_RX_1 107 +#define WSA_CODEC_DMA_TX_1 108 +#define WSA_CODEC_DMA_TX_2 109 +#define VA_CODEC_DMA_TX_0 110 +#define VA_CODEC_DMA_TX_1 111 +#define VA_CODEC_DMA_TX_2 112 +#define RX_CODEC_DMA_RX_0 113 +#define TX_CODEC_DMA_TX_0 114 +#define RX_CODEC_DMA_RX_1 115 +#define TX_CODEC_DMA_TX_1 116 +#define RX_CODEC_DMA_RX_2 117 +#define TX_CODEC_DMA_TX_2 118 +#define RX_CODEC_DMA_RX_3 119 +#define TX_CODEC_DMA_TX_3 120 +#define RX_CODEC_DMA_RX_4 121 +#define TX_CODEC_DMA_TX_4 122 +#define RX_CODEC_DMA_RX_5 123 +#define TX_CODEC_DMA_TX_5 124 +#define RX_CODEC_DMA_RX_6 125 +#define RX_CODEC_DMA_RX_7 126 #endif /* __DT_BINDINGS_Q6_AFE_H__ */ diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..8d23351f7ad7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -93,6 +93,9 @@ struct snd_compr_stream { * @set_params: Sets the compressed stream parameters, mandatory * This can be called in during stream creation only to set codec params * and the stream properties + * @set_codec_params: Sets the compressed stream codec parameters, mandatory + * This can be called in during gapless next track codec change only to set + * codec params * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream * @get_metadata: retrieves the requested metadata values from stream @@ -112,6 +115,8 @@ struct snd_compr_ops { int (*free)(struct snd_compr_stream *stream); int (*set_params)(struct snd_compr_stream *stream, struct snd_compr_params *params); + int (*set_codec_params)(struct snd_compr_stream *stream, + struct snd_codec *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); int (*set_metadata)(struct snd_compr_stream *stream, diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5663891148e3..1e69c54ed0b9 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -36,6 +36,9 @@ struct snd_compress_ops { int (*get_params)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_codec_params)(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *params); int (*set_metadata)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_compr_metadata *metadata); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 212257e84fac..526794ee555b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -192,6 +192,9 @@ int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai, int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_params *params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec); int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_codec *params); @@ -292,6 +295,8 @@ struct snd_soc_cdai_ops { struct snd_soc_dai *); int (*set_params)(struct snd_compr_stream *, struct snd_compr_params *, struct snd_soc_dai *); + int (*set_codec_params)(struct snd_compr_stream *, + struct snd_codec *, struct snd_soc_dai *); int (*get_params)(struct snd_compr_stream *, struct snd_codec *, struct snd_soc_dai *); int (*set_metadata)(struct snd_compr_stream *, diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..c46286113a4b 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -172,6 +172,7 @@ struct snd_compr_metadata { struct snd_compr_metadata) #define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ struct snd_compr_metadata) +#define SNDRV_COMPRESS_SET_CODEC_PARAMS _IOW('C', 0x16, struct snd_codec) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..1c4b2cf450a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -566,6 +566,18 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_codec_params(struct snd_codec *codec) +{ + /* now codec parameters */ + if (codec->id == 0 || codec->id > SND_AUDIOCODEC_MAX) + return -EINVAL; + + if (codec->ch_in == 0 || codec->ch_out == 0) + return -EINVAL; + + return 0; +} + static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ @@ -574,14 +586,41 @@ static int snd_compress_check_input(struct snd_compr_params *params) params->buffer.fragments == 0) return -EINVAL; - /* now codec parameters */ - if (params->codec.id == 0 || params->codec.id > SND_AUDIOCODEC_MAX) - return -EINVAL; + return snd_compress_check_codec_params(¶ms->codec); - if (params->codec.ch_in == 0 || params->codec.ch_out == 0) - return -EINVAL; +} - return 0; +static int snd_compr_set_codec_params(struct snd_compr_stream *stream, + unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->set_codec_params) + return -EPERM; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* codec params can be only set when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + params = memdup_user((void __user *)arg, sizeof(*params)); + if (IS_ERR(params)) + return PTR_ERR(params); + + retval = snd_compress_check_codec_params(params); + if (retval) + goto out; + + retval = stream->ops->set_codec_params(stream, params); + if (retval) + goto out; + +out: + kfree(params); + return retval; } static int @@ -964,6 +1003,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): retval = snd_compr_set_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_CODEC_PARAMS): + retval = snd_compr_set_codec_params(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..493f04e472f9 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -109,3 +109,12 @@ config SND_SOC_SDM845 To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. Say Y if you want to use audio device on this SoCs. +config SND_SOC_SM8250 + tristate "SoC Machine driver for SM8250 boards" + depends on QCOM_APR + select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON + help + To add support for audio on Qualcomm Technologies Inc. + SM8250 SoC-based systems. + Say Y if you want to use audio device on this SoCs. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 41b2c7a23a4d..8236eed36c22 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -15,12 +15,14 @@ snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o snd-soc-apq8096-objs := apq8096.o snd-soc-sdm845-objs := sdm845.o +snd-soc-sm8250-objs := sm8250.o snd-soc-qcom-common-objs := common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o +obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o #DSP lib diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 287ad2aa27f3..690edb332d94 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -5,14 +5,21 @@ #include <linux/platform_device.h> #include <linux/of_device.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #define SLIM_MAX_TX_PORTS 16 #define SLIM_MAX_RX_PORTS 16 #define WCD9335_DEFAULT_MCLK_RATE 9600000 +struct apq8096_card_data { + struct snd_soc_jack jack; + bool jack_setup; +}; + static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -67,6 +74,7 @@ static struct snd_soc_ops apq8096_ops = { static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct apq8096_card_data *data = snd_soc_card_get_drvdata(rtd->card); /* * Codec SLIMBUS configuration @@ -79,6 +87,8 @@ static int apq8096_init(struct snd_soc_pcm_runtime *rtd) unsigned int tx_ch[SLIM_MAX_TX_PORTS] = {128, 129, 130, 131, 132, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143}; + struct snd_soc_card *card = rtd->card; + int rval; snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, ARRAY_SIZE(rx_ch), rx_ch); @@ -86,6 +96,38 @@ static int apq8096_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dai_set_sysclk(codec_dai, 0, WCD9335_DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + if (!data->jack_setup) { + struct snd_jack *jack; + + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4, + &data->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = data->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + data->jack_setup = true; + } + + rval = snd_soc_component_set_jack(codec_dai->component, + &data->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } + return 0; } @@ -105,6 +147,7 @@ static void apq8096_add_be_ops(struct snd_soc_card *card) static int apq8096_platform_probe(struct platform_device *pdev) { + struct apq8096_card_data *data; struct snd_soc_card *card; struct device *dev = &pdev->dev; int ret; @@ -113,8 +156,15 @@ static int apq8096_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) { + kfree(card); + return -ENOMEM; + } + card->dev = dev; dev_set_drvdata(dev, card); + snd_soc_card_set_drvdata(card, data); ret = qcom_snd_parse_of(card); if (ret) goto err; @@ -130,16 +180,19 @@ err_card_register: kfree(card->dai_link); err: kfree(card); + kfree(data); return ret; } static int apq8096_platform_remove(struct platform_device *pdev) { struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + struct apq8096_card_data *data = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); kfree(card->dai_link); kfree(card); + kfree(data); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 2a5302f1db98..d0d20178d3e4 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -55,6 +55,48 @@ .remove = msm_dai_q6_dai_remove, \ } +#define Q6AFE_CDC_DMA_RX_DAI(did) { \ + .playback = { \ + .stream_name = #did" Playback", \ + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_176400, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ + .channels_min = 1, \ + .channels_max = 8, \ + .rate_min = 8000, \ + .rate_max = 176400, \ + }, \ + .name = #did, \ + .ops = &q6dma_ops, \ + .id = did, \ + .probe = msm_dai_q6_dai_probe, \ + .remove = msm_dai_q6_dai_remove, \ + } + +#define Q6AFE_CDC_DMA_TX_DAI(did) { \ + .capture = { \ + .stream_name = #did" Capture", \ + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_176400, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ + .channels_min = 1, \ + .channels_max = 8, \ + .rate_min = 8000, \ + .rate_max = 176400, \ + }, \ + .name = #did, \ + .ops = &q6dma_ops, \ + .id = did, \ + .probe = msm_dai_q6_dai_probe, \ + .remove = msm_dai_q6_dai_remove, \ + } + struct q6afe_dai_priv_data { uint32_t sd_line_mask; uint32_t sync_mode; @@ -307,6 +349,90 @@ static int q6tdm_hw_params(struct snd_pcm_substream *substream, return 0; } + +static int q6dma_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_ch_mask, + unsigned int rx_num, unsigned int *rx_ch_mask) +{ + + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_cdc_dma_cfg *cfg = &dai_data->port_config[dai->id].dma_cfg; + int ch_mask; + int rc = 0; + + switch (dai->id) { + case WSA_CODEC_DMA_TX_0: + case WSA_CODEC_DMA_TX_1: + case WSA_CODEC_DMA_TX_2: + case VA_CODEC_DMA_TX_0: + case VA_CODEC_DMA_TX_1: + case VA_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + case TX_CODEC_DMA_TX_4: + case TX_CODEC_DMA_TX_5: + if (!tx_ch_mask) { + dev_err(dai->dev, "tx slot not found\n"); + return -EINVAL; + } + + if (tx_num > AFE_PORT_MAX_AUDIO_CHAN_CNT) { + dev_err(dai->dev, "invalid tx num %d\n", + tx_num); + return -EINVAL; + } + ch_mask = *tx_ch_mask; + + break; + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_2: + case RX_CODEC_DMA_RX_3: + case RX_CODEC_DMA_RX_4: + case RX_CODEC_DMA_RX_5: + case RX_CODEC_DMA_RX_6: + case RX_CODEC_DMA_RX_7: + /* rx */ + if (!rx_ch_mask) { + dev_err(dai->dev, "rx slot not found\n"); + return -EINVAL; + } + if (rx_num > AFE_PORT_MAX_AUDIO_CHAN_CNT) { + dev_err(dai->dev, "invalid rx num %d\n", + rx_num); + return -EINVAL; + } + ch_mask = *rx_ch_mask; + + break; + default: + dev_err(dai->dev, "%s: invalid dai id 0x%x\n", + __func__, dai->id); + return -EINVAL; + } + + cfg->active_channels_mask = ch_mask; + + return rc; +} + +static int q6dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_cdc_dma_cfg *cfg = &dai_data->port_config[dai->id].dma_cfg; + + cfg->bit_width = params_width(params); + cfg->sample_rate = params_rate(params); + cfg->num_channels = params_channels(params); + + return 0; +} static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -430,6 +556,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR: case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1: + case Q6AFE_CLK_ID_WSA_CORE_MCLK ... Q6AFE_CLK_ID_VA_CORE_2X_MCLK: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO, Q6AFE_LPASS_CLK_ROOT_DEFAULT, @@ -562,6 +689,29 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"PRI_MI2S_TX", NULL, "Primary MI2S Capture"}, {"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"}, {"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"}, + + {"WSA_CODEC_DMA_RX_0 Playback", NULL, "WSA_CODEC_DMA_RX_0"}, + {"WSA_CODEC_DMA_TX_0", NULL, "WSA_CODEC_DMA_TX_0 Capture"}, + {"WSA_CODEC_DMA_RX_1 Playback", NULL, "WSA_CODEC_DMA_RX_1"}, + {"WSA_CODEC_DMA_TX_1", NULL, "WSA_CODEC_DMA_TX_1 Capture"}, + {"WSA_CODEC_DMA_TX_2", NULL, "WSA_CODEC_DMA_TX_2 Capture"}, + {"VA_CODEC_DMA_TX_0", NULL, "VA_CODEC_DMA_TX_0 Capture"}, + {"VA_CODEC_DMA_TX_1", NULL, "VA_CODEC_DMA_TX_1 Capture"}, + {"VA_CODEC_DMA_TX_2", NULL, "VA_CODEC_DMA_TX_2 Capture"}, + {"RX_CODEC_DMA_RX_0 Playback", NULL, "RX_CODEC_DMA_RX_0"}, + {"TX_CODEC_DMA_TX_0", NULL, "TX_CODEC_DMA_TX_0 Capture"}, + {"RX_CODEC_DMA_RX_1 Playback", NULL, "RX_CODEC_DMA_RX_1"}, + {"TX_CODEC_DMA_TX_1", NULL, "TX_CODEC_DMA_TX_1 Capture"}, + {"RX_CODEC_DMA_RX_2 Playback", NULL, "RX_CODEC_DMA_RX_2"}, + {"TX_CODEC_DMA_TX_2", NULL, "TX_CODEC_DMA_TX_2 Capture"}, + {"RX_CODEC_DMA_RX_3 Playback", NULL, "RX_CODEC_DMA_RX_3"}, + {"TX_CODEC_DMA_TX_3", NULL, "TX_CODEC_DMA_TX_3 Capture"}, + {"RX_CODEC_DMA_RX_4 Playback", NULL, "RX_CODEC_DMA_RX_4"}, + {"TX_CODEC_DMA_TX_4", NULL, "TX_CODEC_DMA_TX_4 Capture"}, + {"RX_CODEC_DMA_RX_5 Playback", NULL, "RX_CODEC_DMA_RX_5"}, + {"TX_CODEC_DMA_TX_5", NULL, "TX_CODEC_DMA_TX_5 Capture"}, + {"RX_CODEC_DMA_RX_6 Playback", NULL, "RX_CODEC_DMA_RX_6"}, + {"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"}, }; static const struct snd_soc_dai_ops q6hdmi_ops = { @@ -594,6 +744,14 @@ static const struct snd_soc_dai_ops q6tdm_ops = { .hw_params = q6tdm_hw_params, }; +static const struct snd_soc_dai_ops q6dma_ops = { + .prepare = q6afe_dai_prepare, + .shutdown = q6afe_dai_shutdown, + .set_sysclk = q6afe_mi2s_set_sysclk, + .set_channel_map = q6dma_set_channel_map, + .hw_params = q6dma_hw_params, +}; + static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai) { struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); @@ -1128,6 +1286,28 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, }, + Q6AFE_CDC_DMA_RX_DAI(WSA_CODEC_DMA_RX_0), + Q6AFE_CDC_DMA_TX_DAI(WSA_CODEC_DMA_TX_0), + Q6AFE_CDC_DMA_RX_DAI(WSA_CODEC_DMA_RX_1), + Q6AFE_CDC_DMA_TX_DAI(WSA_CODEC_DMA_TX_1), + Q6AFE_CDC_DMA_TX_DAI(WSA_CODEC_DMA_TX_2), + Q6AFE_CDC_DMA_TX_DAI(VA_CODEC_DMA_TX_0), + Q6AFE_CDC_DMA_TX_DAI(VA_CODEC_DMA_TX_1), + Q6AFE_CDC_DMA_TX_DAI(VA_CODEC_DMA_TX_2), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_0), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_0), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_1), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_1), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_2), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_2), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_3), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_3), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_4), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_4), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_5), + Q6AFE_CDC_DMA_TX_DAI(TX_CODEC_DMA_TX_5), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_6), + Q6AFE_CDC_DMA_RX_DAI(RX_CODEC_DMA_RX_7), }; static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, @@ -1350,6 +1530,51 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL, 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0), + + SND_SOC_DAPM_AIF_IN("WSA_CODEC_DMA_RX_0", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("WSA_CODEC_DMA_TX_0", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("WSA_CODEC_DMA_RX_1", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("WSA_CODEC_DMA_TX_1", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("WSA_CODEC_DMA_TX_2", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("VA_CODEC_DMA_TX_0", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("VA_CODEC_DMA_TX_1", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("VA_CODEC_DMA_TX_2", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_0", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_0", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_1", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_1", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_2", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_2", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_3", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_3", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_4", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_4", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_5", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX_CODEC_DMA_TX_5", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_6", "NULL", + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_7", "NULL", + 0, 0, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 0ce4eb60f984..c2c6649bc630 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -42,6 +42,9 @@ #define AFE_PARAM_ID_I2S_CONFIG 0x0001020D #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 +#define AFE_PARAM_ID_CODEC_DMA_CONFIG 0x000102B8 +#define AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f4 +#define AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST 0x000100f6 /* I2S config specific */ #define AFE_API_VERSION_I2S_CONFIG 0x1 @@ -299,22 +302,72 @@ #define AFE_PORT_ID_QUINARY_TDM_TX_7 \ (AFE_PORT_ID_QUINARY_TDM_TX + 0x0E) +/* AFE WSA Codec DMA Rx port 0 */ +#define AFE_PORT_ID_WSA_CODEC_DMA_RX_0 0xB000 +/* AFE WSA Codec DMA Tx port 0 */ +#define AFE_PORT_ID_WSA_CODEC_DMA_TX_0 0xB001 +/* AFE WSA Codec DMA Rx port 1 */ +#define AFE_PORT_ID_WSA_CODEC_DMA_RX_1 0xB002 +/* AFE WSA Codec DMA Tx port 1 */ +#define AFE_PORT_ID_WSA_CODEC_DMA_TX_1 0xB003 +/* AFE WSA Codec DMA Tx port 2 */ +#define AFE_PORT_ID_WSA_CODEC_DMA_TX_2 0xB005 +/* AFE VA Codec DMA Tx port 0 */ +#define AFE_PORT_ID_VA_CODEC_DMA_TX_0 0xB021 +/* AFE VA Codec DMA Tx port 1 */ +#define AFE_PORT_ID_VA_CODEC_DMA_TX_1 0xB023 +/* AFE VA Codec DMA Tx port 2 */ +#define AFE_PORT_ID_VA_CODEC_DMA_TX_2 0xB025 +/* AFE Rx Codec DMA Rx port 0 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_0 0xB030 +/* AFE Tx Codec DMA Tx port 0 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_0 0xB031 +/* AFE Rx Codec DMA Rx port 1 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_1 0xB032 +/* AFE Tx Codec DMA Tx port 1 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_1 0xB033 +/* AFE Rx Codec DMA Rx port 2 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_2 0xB034 +/* AFE Tx Codec DMA Tx port 2 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_2 0xB035 +/* AFE Rx Codec DMA Rx port 3 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_3 0xB036 +/* AFE Tx Codec DMA Tx port 3 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_3 0xB037 +/* AFE Rx Codec DMA Rx port 4 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_4 0xB038 +/* AFE Tx Codec DMA Tx port 4 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_4 0xB039 +/* AFE Rx Codec DMA Rx port 5 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_5 0xB03A +/* AFE Tx Codec DMA Tx port 5 */ +#define AFE_PORT_ID_TX_CODEC_DMA_TX_5 0xB03B +/* AFE Rx Codec DMA Rx port 6 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_6 0xB03C +/* AFE Rx Codec DMA Rx port 7 */ +#define AFE_PORT_ID_RX_CODEC_DMA_RX_7 0xB03E + #define Q6AFE_LPASS_MODE_CLK1_VALID 1 #define Q6AFE_LPASS_MODE_CLK2_VALID 2 #define Q6AFE_LPASS_CLK_SRC_INTERNAL 1 #define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0 #define AFE_API_VERSION_TDM_CONFIG 1 #define AFE_API_VERSION_SLOT_MAPPING_CONFIG 1 +#define AFE_API_VERSION_CODEC_DMA_CONFIG 1 #define TIMEOUT_MS 1000 #define AFE_CMD_RESP_AVAIL 0 #define AFE_CMD_RESP_NONE 1 +struct aprv2_ibasic_rsp_result_t; + struct q6afe { struct apr_device *apr; struct device *dev; struct q6core_svc_api_info ainfo; struct mutex lock; + struct aprv2_ibasic_rsp_result_t result; + wait_queue_head_t wait; struct list_head port_list; spinlock_t port_list_lock; }; @@ -448,11 +501,21 @@ struct afe_param_id_tdm_cfg { u32 slot_mask; } __packed; +struct afe_param_id_cdc_dma_cfg { + u32 cdc_dma_cfg_minor_version; + u16 sample_rate; + u16 bit_width; + u16 data_format; + u16 num_channels; + u16 active_channels_mask; +} __packed; + union afe_port_config { struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; struct afe_param_id_slimbus_cfg slim_cfg; struct afe_param_id_i2s_cfg i2s_cfg; struct afe_param_id_tdm_cfg tdm_cfg; + struct afe_param_id_cdc_dma_cfg dma_cfg; } __packed; @@ -486,6 +549,18 @@ struct q6afe_port { struct list_head node; }; +struct afe_cmd_remote_lpass_core_hw_vote_request { + uint32_t hw_block_id; + char client_name[8]; +} __packed; + +struct afe_cmd_remote_lpass_core_hw_devote_request { + uint32_t hw_block_id; + uint32_t client_handle; +} __packed; + + + struct afe_port_map { int port_id; int token; @@ -707,6 +782,50 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { QUINARY_TDM_TX_7, 0, 1}, [DISPLAY_PORT_RX] = { AFE_PORT_ID_HDMI_OVER_DP_RX, DISPLAY_PORT_RX, 1, 1}, + [WSA_CODEC_DMA_RX_0] = { AFE_PORT_ID_WSA_CODEC_DMA_RX_0, + WSA_CODEC_DMA_RX_0, 1, 1}, + [WSA_CODEC_DMA_TX_0] = { AFE_PORT_ID_WSA_CODEC_DMA_TX_0, + WSA_CODEC_DMA_TX_0, 0, 1}, + [WSA_CODEC_DMA_RX_1] = { AFE_PORT_ID_WSA_CODEC_DMA_RX_1, + WSA_CODEC_DMA_RX_1, 1, 1}, + [WSA_CODEC_DMA_TX_1] = { AFE_PORT_ID_WSA_CODEC_DMA_TX_1, + WSA_CODEC_DMA_TX_1, 0, 1}, + [WSA_CODEC_DMA_TX_2] = { AFE_PORT_ID_WSA_CODEC_DMA_TX_2, + WSA_CODEC_DMA_TX_2, 0, 1}, + [VA_CODEC_DMA_TX_0] = { AFE_PORT_ID_VA_CODEC_DMA_TX_0, + VA_CODEC_DMA_TX_0, 0, 1}, + [VA_CODEC_DMA_TX_1] = { AFE_PORT_ID_VA_CODEC_DMA_TX_1, + VA_CODEC_DMA_TX_1, 0, 1}, + [VA_CODEC_DMA_TX_2] = { AFE_PORT_ID_VA_CODEC_DMA_TX_2, + VA_CODEC_DMA_TX_2, 0, 1}, + [RX_CODEC_DMA_RX_0] = { AFE_PORT_ID_RX_CODEC_DMA_RX_0, + RX_CODEC_DMA_RX_0, 1, 1}, + [TX_CODEC_DMA_TX_0] = { AFE_PORT_ID_TX_CODEC_DMA_TX_0, + TX_CODEC_DMA_TX_0, 0, 1}, + [RX_CODEC_DMA_RX_1] = { AFE_PORT_ID_RX_CODEC_DMA_RX_1, + RX_CODEC_DMA_RX_1, 1, 1}, + [TX_CODEC_DMA_TX_1] = { AFE_PORT_ID_TX_CODEC_DMA_TX_1, + TX_CODEC_DMA_TX_1, 0, 1}, + [RX_CODEC_DMA_RX_2] = { AFE_PORT_ID_RX_CODEC_DMA_RX_2, + RX_CODEC_DMA_RX_2, 1, 1}, + [TX_CODEC_DMA_TX_2] = { AFE_PORT_ID_TX_CODEC_DMA_TX_2, + TX_CODEC_DMA_TX_2, 0, 1}, + [RX_CODEC_DMA_RX_3] = { AFE_PORT_ID_RX_CODEC_DMA_RX_3, + RX_CODEC_DMA_RX_3, 1, 1}, + [TX_CODEC_DMA_TX_3] = { AFE_PORT_ID_TX_CODEC_DMA_TX_3, + TX_CODEC_DMA_TX_3, 0, 1}, + [RX_CODEC_DMA_RX_4] = { AFE_PORT_ID_RX_CODEC_DMA_RX_4, + RX_CODEC_DMA_RX_4, 1, 1}, + [TX_CODEC_DMA_TX_4] = { AFE_PORT_ID_TX_CODEC_DMA_TX_4, + TX_CODEC_DMA_TX_4, 0, 1}, + [RX_CODEC_DMA_RX_5] = { AFE_PORT_ID_RX_CODEC_DMA_RX_5, + RX_CODEC_DMA_RX_5, 1, 1}, + [TX_CODEC_DMA_TX_5] = { AFE_PORT_ID_TX_CODEC_DMA_TX_5, + TX_CODEC_DMA_TX_5, 0, 1}, + [RX_CODEC_DMA_RX_6] = { AFE_PORT_ID_RX_CODEC_DMA_RX_6, + RX_CODEC_DMA_RX_6, 1, 1}, + [RX_CODEC_DMA_RX_7] = { AFE_PORT_ID_RX_CODEC_DMA_RX_7, + RX_CODEC_DMA_RX_7, 1, 1}, }; static void q6afe_port_free(struct kref *ref) @@ -771,6 +890,11 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) kref_put(&port->refcount, q6afe_port_free); } break; + case AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST: + case AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST: + afe->result = *res; + wake_up(&afe->wait); + break; default: dev_err(afe->dev, "Unknown cmd 0x%x\n", res->opcode); break; @@ -808,16 +932,26 @@ int q6afe_is_rx_port(int index) return port_maps[index].is_rx; } EXPORT_SYMBOL_GPL(q6afe_is_rx_port); + static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, struct q6afe_port *port) { wait_queue_head_t *wait = &port->wait; struct apr_hdr *hdr = &pkt->hdr; + struct aprv2_ibasic_rsp_result_t *result; int ret; mutex_lock(&afe->lock); - port->result.opcode = 0; - port->result.status = 0; + if (port) { + wait = &port->wait; + result = &port->result; + } else { + result = &afe->result; + wait = &afe->wait; + } + + result->opcode = 0; + result->status = 0; ret = apr_send_pkt(afe->apr, pkt); if (ret < 0) { @@ -826,13 +960,13 @@ static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, goto err; } - ret = wait_event_timeout(*wait, (port->result.opcode == hdr->opcode), + ret = wait_event_timeout(*wait, (result->opcode == hdr->opcode), msecs_to_jiffies(TIMEOUT_MS)); if (!ret) { ret = -ETIMEDOUT; - } else if (port->result.status > 0) { + } else if (result->status > 0) { dev_err(afe->dev, "DSP returned error[%x]\n", - port->result.status); + result->status); ret = -EINVAL; } else { ret = 0; @@ -1002,6 +1136,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR: case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1: case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: + case Q6AFE_CLK_ID_WSA_CORE_MCLK ... Q6AFE_CLK_ID_VA_CORE_2X_MCLK: cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET; cset.clk_id = clk_id; cset.clk_freq_in_hz = freq; @@ -1297,6 +1432,27 @@ int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg) EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare); /** + * q6afe_dam_port_prepare() - Prepare dma afe port. + * + * @port: Instance of afe port + * @cfg: DMA configuration for the afe port + * + */ +void q6afe_cdc_dma_port_prepare(struct q6afe_port *port, + struct q6afe_cdc_dma_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + struct afe_param_id_cdc_dma_cfg *dma_cfg = &pcfg->dma_cfg; + + dma_cfg->cdc_dma_cfg_minor_version = AFE_API_VERSION_CODEC_DMA_CONFIG; + dma_cfg->sample_rate = cfg->sample_rate; + dma_cfg->bit_width = cfg->bit_width; + dma_cfg->data_format = cfg->data_format; + dma_cfg->num_channels = cfg->num_channels; + dma_cfg->active_channels_mask = cfg->active_channels_mask; +} +EXPORT_SYMBOL_GPL(q6afe_cdc_dma_port_prepare); +/** * q6afe_port_start() - Start a afe port * * @port: Instance of port to start @@ -1428,7 +1584,9 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_PRIMARY_TDM_RX ... AFE_PORT_ID_QUINARY_TDM_TX_7: cfg_type = AFE_PARAM_ID_TDM_CONFIG; break; - + case AFE_PORT_ID_WSA_CODEC_DMA_RX_0 ... AFE_PORT_ID_RX_CODEC_DMA_RX_7: + cfg_type = AFE_PARAM_ID_CODEC_DMA_CONFIG; + break; default: dev_err(dev, "Invalid port id 0x%x\n", port_id); return ERR_PTR(-EINVAL); @@ -1466,6 +1624,81 @@ void q6afe_port_put(struct q6afe_port *port) } EXPORT_SYMBOL_GPL(q6afe_port_put); +int q6afe_unvote_lpass_core_hw(struct q6afe *afe, uint32_t hw_block_id, + uint32_t client_handle) +{ + struct afe_cmd_remote_lpass_core_hw_devote_request *vote_cfg; + struct apr_pkt *pkt; + int ret = 0; + int pkt_size; + void *p; + + pkt_size = APR_HDR_SIZE + sizeof(*vote_cfg); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + vote_cfg = p + APR_HDR_SIZE; + + pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + pkt->hdr.pkt_size = pkt_size; + pkt->hdr.src_port = 0; + pkt->hdr.dest_port = 0; + pkt->hdr.token = hw_block_id; + pkt->hdr.opcode = AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST; + vote_cfg->hw_block_id = hw_block_id; + vote_cfg->client_handle = client_handle; + + ret = afe_apr_send_pkt(afe, pkt, NULL); + if (ret) + dev_err(afe->dev, "AFE close failed %d\n", ret); + + kfree(pkt); + return ret; +} +EXPORT_SYMBOL(q6afe_unvote_lpass_core_hw); + +int q6afe_vote_lpass_core_hw(struct q6afe *afe, uint32_t hw_block_id, + char *client_name, uint32_t *client_handle) +{ + struct afe_cmd_remote_lpass_core_hw_vote_request *vote_cfg; + struct apr_pkt *pkt; + int ret = 0; + int pkt_size; + void *p; + + pkt_size = APR_HDR_SIZE + sizeof(*vote_cfg); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + vote_cfg = p + APR_HDR_SIZE; + + pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + pkt->hdr.pkt_size = pkt_size; + pkt->hdr.src_port = 0; + pkt->hdr.dest_port = 0; + pkt->hdr.token = hw_block_id; + pkt->hdr.opcode = AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST; + vote_cfg->hw_block_id = hw_block_id; + strlcpy(vote_cfg->client_name, client_name, + sizeof(vote_cfg->client_name)); + + ret = afe_apr_send_pkt(afe, pkt, NULL); + if (ret) + dev_err(afe->dev, "AFE close failed %d\n", ret); + + kfree(pkt); + return ret; +} +EXPORT_SYMBOL(q6afe_vote_lpass_core_hw); + static int q6afe_probe(struct apr_device *adev) { struct q6afe *afe; @@ -1478,6 +1711,7 @@ static int q6afe_probe(struct apr_device *adev) q6core_get_svc_api_info(adev->svc_id, &afe->ainfo); afe->apr = adev; mutex_init(&afe->lock); + init_waitqueue_head(&afe->wait); afe->dev = dev; INIT_LIST_HEAD(&afe->port_list); spin_lock_init(&afe->port_list_lock); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 1a0f80a14afe..f06140488fdb 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -5,7 +5,7 @@ #include <dt-bindings/sound/qcom,q6afe.h> -#define AFE_PORT_MAX 105 +#define AFE_PORT_MAX 127 #define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -132,6 +132,17 @@ #define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305 /* Clock ID for INT MCLK1 */ #define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306 +/* Clock ID for MCLK5 */ +#define Q6AFE_LPASS_CLK_ID_MCLK_5 0x308 + +#define Q6AFE_CLK_ID_WSA_CORE_MCLK 0x309 +#define Q6AFE_CLK_ID_WSA_CORE_NPL_MCLK 0x30a +#define Q6AFE_CLK_ID_VA_CORE_MCLK 0x30b +#define Q6AFE_CLK_ID_TX_CORE_MCLK 0x30c +#define Q6AFE_CLK_ID_TX_CORE_NPL_MCLK 0x30d +#define Q6AFE_CLK_ID_RX_CORE_MCLK 0x30e +#define Q6AFE_CLK_ID_RX_CORE_NPL_MCLK 0x30f +#define Q6AFE_CLK_ID_VA_CORE_2X_MCLK 0x310 /* Clock attribute for invalid use (reserved for internal usage) */ #define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0 @@ -184,11 +195,21 @@ struct q6afe_tdm_cfg { u16 ch_mapping[AFE_MAX_CHAN_COUNT]; }; +struct q6afe_cdc_dma_cfg { + u16 sample_rate; + u16 bit_width; + u16 data_format; + u16 num_channels; + u16 active_channels_mask; +}; + + struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; struct q6afe_slim_cfg slim; struct q6afe_i2s_cfg i2s_cfg; struct q6afe_tdm_cfg tdm; + struct q6afe_cdc_dma_cfg dma_cfg; }; struct q6afe_port; @@ -205,8 +226,15 @@ void q6afe_slim_port_prepare(struct q6afe_port *port, struct q6afe_slim_cfg *cfg); int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg); void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg); +void q6afe_cdc_dma_port_prepare(struct q6afe_port *port, + struct q6afe_cdc_dma_cfg *cfg); int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, int clk_src, int clk_root, unsigned int freq, int dir); +struct q6afe; +int q6afe_vote_lpass_core_hw(struct q6afe *afe, uint32_t hw_block_id, + char *client_name, uint32_t *client_handle); +int q6afe_unvote_lpass_core_hw(struct q6afe *afe, uint32_t hw_block_id, + uint32_t client_handle); #endif /* __Q6AFE_H__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index aff57052a735..e637fe958c23 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -53,7 +53,7 @@ enum stream_state { struct q6asm_dai_rtd { struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; - struct snd_compr_params codec_param; + struct snd_codec codec; struct snd_dma_buffer dma_buffer; spinlock_t lock; phys_addr_t phys; @@ -67,8 +67,15 @@ struct q6asm_dai_rtd { uint16_t bits_per_sample; uint16_t source; /* Encoding source bit mask */ struct audio_client *audio_client; + uint32_t next_track_stream_id; + bool next_track; + /* Active */ + uint32_t stream_id; uint16_t session_id; enum stream_state state; + uint32_t initial_samples_drop; + uint32_t trailing_samples_drop; + bool notify_on_drain; }; struct q6asm_dai_data { @@ -184,8 +191,8 @@ static void event_handler(uint32_t opcode, uint32_t token, switch (opcode) { case ASM_CLIENT_EVENT_CMD_RUN_DONE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - q6asm_write_async(prtd->audio_client, - prtd->pcm_count, 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); break; case ASM_CLIENT_EVENT_CMD_EOS_DONE: prtd->state = Q6ASM_STREAM_STOPPED; @@ -194,8 +201,8 @@ static void event_handler(uint32_t opcode, uint32_t token, prtd->pcm_irq_pos += prtd->pcm_count; snd_pcm_period_elapsed(substream); if (prtd->state == Q6ASM_STREAM_RUNNING) - q6asm_write_async(prtd->audio_client, - prtd->pcm_count, 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); break; } @@ -203,7 +210,7 @@ static void event_handler(uint32_t opcode, uint32_t token, prtd->pcm_irq_pos += prtd->pcm_count; snd_pcm_period_elapsed(substream); if (prtd->state == Q6ASM_STREAM_RUNNING) - q6asm_read(prtd->audio_client); + q6asm_read(prtd->audio_client, prtd->stream_id); break; default: @@ -218,6 +225,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; struct q6asm_dai_rtd *prtd = runtime->private_data; struct q6asm_dai_data *pdata; + struct device *dev = component->dev; int ret, i; pdata = snd_soc_component_get_drvdata(component); @@ -225,7 +233,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, return -EINVAL; if (!prtd || !prtd->audio_client) { - pr_err("%s: private data null or audio client freed\n", + dev_err(dev, "%s: private data null or audio client freed\n", __func__); return -EINVAL; } @@ -235,7 +243,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, /* rate and channels are sent to audio driver */ if (prtd->state) { /* clear the previous setup if any */ - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6routing_stream_close(soc_prtd->dai_link->id, @@ -248,21 +256,23 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, prtd->periods); if (ret < 0) { - pr_err("Audio Start: Buffer Allocation failed rc = %d\n", + dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n", ret); return -ENOMEM; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, - 0, prtd->bits_per_sample); + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, + 0, prtd->bits_per_sample, false); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, + prtd->bits_per_sample); } if (ret < 0) { - pr_err("%s: q6asm_open_write failed\n", __func__); + dev_err(dev, "%s: q6asm_open_write failed\n", __func__); q6asm_audio_client_free(prtd->audio_client); prtd->audio_client = NULL; return -ENOMEM; @@ -272,27 +282,29 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE, prtd->session_id, substream->stream); if (ret) { - pr_err("%s: stream reg failed ret:%d\n", __func__, ret); + dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret); return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_media_format_block_multi_ch_pcm( - prtd->audio_client, runtime->rate, - runtime->channels, NULL, + prtd->audio_client, prtd->stream_id, + runtime->rate, runtime->channels, NULL, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client, - runtime->rate, runtime->channels, - prtd->bits_per_sample); + prtd->stream_id, + runtime->rate, + runtime->channels, + prtd->bits_per_sample); /* Queue the buffers */ for (i = 0; i < runtime->periods; i++) - q6asm_read(prtd->audio_client); + q6asm_read(prtd->audio_client, prtd->stream_id); } if (ret < 0) - pr_info("%s: CMD Format block failed\n", __func__); + dev_info(dev, "%s: CMD Format block failed\n", __func__); prtd->state = Q6ASM_STREAM_RUNNING; @@ -310,15 +322,18 @@ static int q6asm_dai_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: prtd->state = Q6ASM_STREAM_STOPPED; - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); break; default: ret = -EINVAL; @@ -344,7 +359,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, pdata = snd_soc_component_get_drvdata(component); if (!pdata) { - pr_err("Drv data not found ..\n"); + dev_err(dev, "Drv data not found ..\n"); return -EINVAL; } @@ -357,12 +372,15 @@ static int q6asm_dai_open(struct snd_soc_component *component, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); if (IS_ERR(prtd->audio_client)) { - pr_info("%s: Could not allocate memory\n", __func__); + dev_info(dev, "%s: Could not allocate memory\n", __func__); ret = PTR_ERR(prtd->audio_client); kfree(prtd); return ret; } + /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = q6asm_dai_hardware_playback; else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -372,12 +390,12 @@ static int q6asm_dai_open(struct snd_soc_component *component, SNDRV_PCM_HW_PARAM_RATE, &constraints_sample_rates); if (ret < 0) - pr_info("snd_pcm_hw_constraint_list failed\n"); + dev_info(dev, "snd_pcm_hw_constraint_list failed\n"); /* Ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - pr_info("snd_pcm_hw_constraint_integer failed\n"); + dev_info(dev, "snd_pcm_hw_constraint_integer failed\n"); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = snd_pcm_hw_constraint_minmax(runtime, @@ -385,21 +403,21 @@ static int q6asm_dai_open(struct snd_soc_component *component, PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE, PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE); if (ret < 0) { - pr_err("constraint for buffer bytes min max ret = %d\n", - ret); + dev_err(dev, "constraint for buffer bytes min max ret = %d\n", + ret); } } ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret < 0) { - pr_err("constraint for period bytes step ret = %d\n", + dev_err(dev, "constraint for period bytes step ret = %d\n", ret); } ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret < 0) { - pr_err("constraint for buffer bytes step ret = %d\n", + dev_err(dev, "constraint for buffer bytes step ret = %d\n", ret); } @@ -429,7 +447,8 @@ static int q6asm_dai_close(struct snd_soc_component *component, if (prtd->audio_client) { if (prtd->state) - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE); q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); @@ -493,16 +512,24 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; - struct snd_compr_stream *substream = prtd->cstream; - unsigned long flags; + struct snd_compr_stream *stream = prtd->cstream; + unsigned long flags = 0; + u32 wflags = 0; uint64_t avail; + uint32_t bytes_written, bytes_to_write; + bool is_last_buffer = false; switch (opcode) { case ASM_CLIENT_EVENT_CMD_RUN_DONE: spin_lock_irqsave(&prtd->lock, flags); if (!prtd->bytes_sent) { - q6asm_write_async(prtd->audio_client, prtd->pcm_count, - 0, 0, NO_TIMESTAMP); + q6asm_stream_remove_initial_silence(prtd->audio_client, + prtd->stream_id, + prtd->initial_samples_drop); + avail = prtd->bytes_received - prtd->bytes_sent; + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); prtd->bytes_sent += prtd->pcm_count; } @@ -510,14 +537,32 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, break; case ASM_CLIENT_EVENT_CMD_EOS_DONE: - prtd->state = Q6ASM_STREAM_STOPPED; + if (prtd->notify_on_drain) { + if (stream->partial_drain && prtd->next_track_stream_id) { + /* Close old stream and make it stale, switch + * the active stream now! */ + q6asm_cmd_nowait(prtd->audio_client, + prtd->stream_id, + CMD_CLOSE); + prtd->stream_id = prtd->next_track_stream_id; + prtd->next_track_stream_id = 0; + } + + snd_compr_drain_notify(prtd->cstream); + prtd->notify_on_drain = false; + + } else { + prtd->state = Q6ASM_STREAM_STOPPED; + } break; case ASM_CLIENT_EVENT_DATA_WRITE_DONE: spin_lock_irqsave(&prtd->lock, flags); - prtd->copied_total += prtd->pcm_count; - snd_compr_fragment_elapsed(substream); + bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT; + prtd->copied_total += bytes_written; + + snd_compr_fragment_elapsed(stream); if (prtd->state != Q6ASM_STREAM_RUNNING) { spin_unlock_irqrestore(&prtd->lock, flags); @@ -525,13 +570,32 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, } avail = prtd->bytes_received - prtd->bytes_sent; + if (avail > prtd->pcm_count) { + bytes_to_write = prtd->pcm_count; + } else { + if (stream->partial_drain || prtd->notify_on_drain) + is_last_buffer = true; + bytes_to_write = avail; + } - if (avail >= prtd->pcm_count) { - q6asm_write_async(prtd->audio_client, - prtd->pcm_count, 0, 0, NO_TIMESTAMP); - prtd->bytes_sent += prtd->pcm_count; + if (bytes_to_write) { + if (stream->partial_drain && is_last_buffer) { + wflags |= ASM_LAST_BUFFER_FLAG; + q6asm_stream_remove_trailing_silence(prtd->audio_client, + prtd->stream_id, + prtd->trailing_samples_drop); + } + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + bytes_to_write, 0, 0, wflags); + + prtd->bytes_sent += bytes_to_write; } + if (prtd->notify_on_drain && is_last_buffer) + q6asm_cmd_nowait(prtd->audio_client, + prtd->stream_id, CMD_EOS); + spin_unlock_irqrestore(&prtd->lock, flags); break; @@ -562,6 +626,9 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component, if (!prtd) return -ENOMEM; + /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + prtd->cstream = stream; prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)compress_event_handler, @@ -586,7 +653,6 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component, else prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32); - snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer); spin_lock_init(&prtd->lock); runtime->private_data = prtd; @@ -608,8 +674,15 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = stream->private_data; if (prtd->audio_client) { - if (prtd->state) - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + if (prtd->state) { + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE); + if (prtd->next_track_stream_id) { + q6asm_cmd(prtd->audio_client, + prtd->next_track_stream_id, + CMD_CLOSE); + } + } snd_dma_free_pages(&prtd->dma_buffer); q6asm_unmap_memory_regions(stream->direction, @@ -623,21 +696,20 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component, return 0; } -static int q6asm_dai_compr_set_params(struct snd_soc_component *component, - struct snd_compr_stream *stream, - struct snd_compr_params *params) +static int __q6asm_dai_compr_set_codec_params(struct snd_compr_stream *stream, + struct snd_codec *codec, + int stream_id) { struct snd_compr_runtime *runtime = stream->runtime; struct q6asm_dai_rtd *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = stream->private_data; - int dir = stream->direction; - struct q6asm_dai_data *pdata; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct q6asm_flac_cfg flac_cfg; struct q6asm_wma_cfg wma_cfg; struct q6asm_alac_cfg alac_cfg; struct q6asm_ape_cfg ape_cfg; unsigned int wma_v9 = 0; - struct device *dev = component->dev; + struct device *dev = c->dev; int ret; union snd_codec_options *codec_options; struct snd_dec_flac *flac; @@ -645,52 +717,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_dec_alac *alac; struct snd_dec_ape *ape; - codec_options = &(prtd->codec_param.codec.options); - - - memcpy(&prtd->codec_param, params, sizeof(*params)); - - pdata = snd_soc_component_get_drvdata(component); - if (!pdata) - return -EINVAL; - - if (!prtd || !prtd->audio_client) { - dev_err(dev, "private data null or audio client freed\n"); - return -EINVAL; - } - - prtd->periods = runtime->fragments; - prtd->pcm_count = runtime->fragment_size; - prtd->pcm_size = runtime->fragments * runtime->fragment_size; - prtd->bits_per_sample = 16; - if (dir == SND_COMPRESS_PLAYBACK) { - ret = q6asm_open_write(prtd->audio_client, params->codec.id, - params->codec.profile, prtd->bits_per_sample); - - if (ret < 0) { - dev_err(dev, "q6asm_open_write failed\n"); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return ret; - } - } + codec_options = &(prtd->codec.options); - prtd->session_id = q6asm_get_session_id(prtd->audio_client); - ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, - prtd->session_id, dir); - if (ret) { - dev_err(dev, "Stream reg failed ret:%d\n", ret); - return ret; - } + memcpy(&prtd->codec, codec, sizeof(*codec)); - switch (params->codec.id) { + switch (codec->id) { case SND_AUDIOCODEC_FLAC: memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); flac = &codec_options->flac_d; - flac_cfg.ch_cfg = params->codec.ch_in; - flac_cfg.sample_rate = params->codec.sample_rate; + flac_cfg.ch_cfg = codec->ch_in; + flac_cfg.sample_rate = codec->sample_rate; flac_cfg.stream_info_present = 1; flac_cfg.sample_size = flac->sample_size; flac_cfg.min_blk_size = flac->min_blk_size; @@ -699,6 +737,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, flac_cfg.min_frame_size = flac->min_frame_size; ret = q6asm_stream_media_format_block_flac(prtd->audio_client, + stream_id, &flac_cfg); if (ret < 0) { dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); @@ -711,10 +750,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); - wma_cfg.sample_rate = params->codec.sample_rate; - wma_cfg.num_channels = params->codec.ch_in; - wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; - wma_cfg.block_align = params->codec.align; + wma_cfg.sample_rate = codec->sample_rate; + wma_cfg.num_channels = codec->ch_in; + wma_cfg.bytes_per_sec = codec->bit_rate / 8; + wma_cfg.block_align = codec->align; wma_cfg.bits_per_sample = prtd->bits_per_sample; wma_cfg.enc_options = wma->encoder_option; wma_cfg.adv_enc_options = wma->adv_encoder_option; @@ -728,7 +767,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, return -EINVAL; /* check the codec profile */ - switch (params->codec.profile) { + switch (codec->profile) { case SND_AUDIOPROFILE_WMA9: wma_cfg.fmtag = 0x161; wma_v9 = 1; @@ -752,16 +791,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, default: dev_err(dev, "Unknown WMA profile:%x\n", - params->codec.profile); + codec->profile); return -EIO; } if (wma_v9) ret = q6asm_stream_media_format_block_wma_v9( - prtd->audio_client, &wma_cfg); + prtd->audio_client, stream_id, + &wma_cfg); else ret = q6asm_stream_media_format_block_wma_v10( - prtd->audio_client, &wma_cfg); + prtd->audio_client, stream_id, + &wma_cfg); if (ret < 0) { dev_err(dev, "WMA9 CMD failed:%d\n", ret); return -EIO; @@ -772,10 +813,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&alac_cfg, 0x0, sizeof(alac_cfg)); alac = &codec_options->alac_d; - alac_cfg.sample_rate = params->codec.sample_rate; - alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.sample_rate = codec->sample_rate; + alac_cfg.avg_bit_rate = codec->bit_rate; alac_cfg.bit_depth = prtd->bits_per_sample; - alac_cfg.num_channels = params->codec.ch_in; + alac_cfg.num_channels = codec->ch_in; alac_cfg.frame_length = alac->frame_length; alac_cfg.pb = alac->pb; @@ -785,7 +826,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, alac_cfg.compatible_version = alac->compatible_version; alac_cfg.max_frame_bytes = alac->max_frame_bytes; - switch (params->codec.ch_in) { + switch (codec->ch_in) { case 1: alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; break; @@ -794,6 +835,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, break; } ret = q6asm_stream_media_format_block_alac(prtd->audio_client, + stream_id, &alac_cfg); if (ret < 0) { dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); @@ -805,8 +847,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&ape_cfg, 0x0, sizeof(ape_cfg)); ape = &codec_options->ape_d; - ape_cfg.sample_rate = params->codec.sample_rate; - ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.sample_rate = codec->sample_rate; + ape_cfg.num_channels = codec->ch_in; ape_cfg.bits_per_sample = prtd->bits_per_sample; ape_cfg.compatible_version = ape->compatible_version; @@ -818,6 +860,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, ape_cfg.seek_table_present = ape->seek_table_present; ret = q6asm_stream_media_format_block_ape(prtd->audio_client, + stream_id, &ape_cfg); if (ret < 0) { dev_err(dev, "APE CMD Format block failed:%d\n", ret); @@ -829,6 +872,98 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, break; } + return 0; +} + +static int q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *codec) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret; + + ret = q6asm_open_write(prtd->audio_client, prtd->next_track_stream_id, + codec->id, codec->profile, prtd->bits_per_sample, + true); + if (ret < 0) { + dev_err(component->dev, "q6asm_open_write failed\n"); + return ret; + } + + ret = __q6asm_dai_compr_set_codec_params(stream, codec, + prtd->next_track_stream_id); + if (ret < 0) { + dev_err(component->dev, "q6asm_open_write failed\n"); + return ret; + } + + return q6asm_stream_remove_initial_silence(prtd->audio_client, + prtd->next_track_stream_id, + prtd->initial_samples_drop); +} + +static int q6asm_dai_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + int dir = stream->direction; + struct q6asm_dai_data *pdata; + struct device *dev = c->dev; + int ret; + union snd_codec_options *codec_options; + + codec_options = &(prtd->codec.options); + + + memcpy(&prtd->codec, ¶ms->codec, sizeof(params->codec)); + + pdata = snd_soc_component_get_drvdata(c); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + dev_err(dev, "private data null or audio client freed\n"); + return -EINVAL; + } + + prtd->periods = runtime->fragments; + prtd->pcm_count = runtime->fragment_size; + prtd->pcm_size = runtime->fragments * runtime->fragment_size; + prtd->bits_per_sample = 16; + + if (dir == SND_COMPRESS_PLAYBACK) { + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id, + params->codec.profile, prtd->bits_per_sample, + true); + + if (ret < 0) { + dev_err(dev, "q6asm_open_write failed\n"); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; + } + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, dir); + if (ret) { + dev_err(dev, "Stream reg failed ret:%d\n", ret); + return ret; + } + + ret = __q6asm_dai_compr_set_codec_params(stream, &prtd->codec, + prtd->stream_id); + if (ret) { + dev_err(dev, "codec param setup failed ret:%d\n", ret); + return ret; + } + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, (prtd->pcm_size / prtd->periods), prtd->periods); @@ -843,6 +978,29 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, return 0; } +static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret = 0; + + switch (metadata->key) { + case SNDRV_COMPRESS_ENCODER_PADDING: + prtd->trailing_samples_drop = metadata->value[0]; + break; + case SNDRV_COMPRESS_ENCODER_DELAY: + prtd->initial_samples_drop = metadata->value[0]; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + static int q6asm_dai_compr_trigger(struct snd_soc_component *component, struct snd_compr_stream *stream, int cmd) { @@ -854,15 +1012,29 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: prtd->state = Q6ASM_STREAM_STOPPED; - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); + break; + case SND_COMPR_TRIGGER_NEXT_TRACK: + /* Get Next stream id + open it + */ + prtd->next_track = true; + prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1); + break; + case SND_COMPR_TRIGGER_DRAIN: + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + prtd->notify_on_drain = true; break; default: ret = -EINVAL; @@ -890,16 +1062,76 @@ static int q6asm_dai_compr_pointer(struct snd_soc_component *component, return 0; } -static int q6asm_dai_compr_ack(struct snd_soc_component *component, - struct snd_compr_stream *stream, - size_t count) +static int q6asm_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *stream, char __user *buf, + size_t count) { struct snd_compr_runtime *runtime = stream->runtime; struct q6asm_dai_rtd *prtd = runtime->private_data; unsigned long flags; + u32 wflags = 0; + int avail, bytes_in_flight = 0; + void *dstn; + size_t copy; + u32 app_pointer; + u32 bytes_received; + + + bytes_received = prtd->bytes_received; + + /** + * Make sure that next track data pointer is aligned at 32 bit boundary + * This is a Mandatory requirement from DSP data buffers alignment + */ + if (prtd->next_track) + bytes_received= ALIGN(prtd->bytes_received, prtd->pcm_count); + + app_pointer = bytes_received/prtd->pcm_size; + app_pointer = bytes_received - (app_pointer * prtd->pcm_size); + dstn = prtd->dma_buffer.area + app_pointer; + + if (count < prtd->pcm_size - app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + } else { + copy = prtd->pcm_size - app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(prtd->dma_buffer.area, buf + copy, + count - copy)) + return -EFAULT; + } spin_lock_irqsave(&prtd->lock, flags); + + bytes_in_flight = prtd->bytes_received - prtd->copied_total; + + if (prtd->next_track) { + /* Adjust the bytes sent and copied as per new aligment */ + prtd->next_track = false; + prtd->bytes_received= ALIGN(prtd->bytes_received, prtd->pcm_count); + prtd->copied_total= ALIGN(prtd->copied_total, prtd->pcm_count); + prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count); + } prtd->bytes_received += count; + + /* Kick off the data to dsp if its starving!! */ + if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) { + uint32_t bytes_to_write = prtd->pcm_count; + + avail = prtd->bytes_received - prtd->bytes_sent; + + if (avail > prtd->pcm_count) { + bytes_to_write = prtd->pcm_count; + } else { + bytes_to_write = avail; + } + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + bytes_to_write, 0, 0, wflags); + prtd->bytes_sent += bytes_to_write; + } + spin_unlock_irqrestore(&prtd->lock, flags); return count; @@ -956,12 +1188,14 @@ static struct snd_compress_ops q6asm_dai_compress_ops = { .open = q6asm_dai_compr_open, .free = q6asm_dai_compr_free, .set_params = q6asm_dai_compr_set_params, + .set_codec_params = q6asm_dai_compr_set_codec_params, + .set_metadata = q6asm_dai_compr_set_metadata, .pointer = q6asm_dai_compr_pointer, .trigger = q6asm_dai_compr_trigger, .get_caps = q6asm_dai_compr_get_caps, .get_codec_caps = q6asm_dai_compr_get_codec_caps, .mmap = q6asm_dai_compr_mmap, - .ack = q6asm_dai_compr_ack, + .copy = q6asm_compr_copy, }; static int q6asm_dai_pcm_new(struct snd_soc_component *component, @@ -1026,7 +1260,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = { .mmap = q6asm_dai_mmap, .pcm_construct = q6asm_dai_pcm_new, .pcm_destruct = q6asm_dai_pcm_free, - .compress_ops = &q6asm_dai_compress_ops, + .compress_ops =&q6asm_dai_compress_ops, }; static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index ae4b2cabdf2d..c663b9308144 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -51,6 +51,8 @@ #define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D #define ASM_MEDIA_FMT_ALAC 0x00012f31 #define ASM_MEDIA_FMT_APE 0x00012f32 +#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67 +#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68 #define ASM_LEGACY_STREAM_SESSION 0 @@ -62,7 +64,7 @@ #define ASM_ASYNC_IO_MODE 0x0002 #define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ #define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */ -#define ASM_SHIFT_GAPLESS_MODE_FLAG 31 +#define ASM_SHIFT_GAPLESS_MODE_FLAG BIT(31) #define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3 struct avs_cmd_shared_mem_map_regions { @@ -270,7 +272,6 @@ struct audio_client { wait_queue_head_t cmd_wait; struct aprv2_ibasic_rsp_result_t result; int perf_mode; - int stream_id; struct q6asm *q6asm; struct device *dev; }; @@ -311,7 +312,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac, 5 * HZ); if (!rc) { - dev_err(a->dev, "CMD timeout\n"); + dev_err(a->dev, "CMD %x timeout\n", hdr->opcode); rc = -ETIMEDOUT; } else if (ac->result.status > 0) { dev_err(a->dev, "DSP returned error[%x]\n", @@ -640,6 +641,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, case ASM_STREAM_CMD_OPEN_READWRITE_V2: case ASM_STREAM_CMD_SET_ENCDEC_PARAM: case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2: + case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE: + case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE: if (result->status != 0) { dev_err(ac->dev, "cmd = 0x%x returned error = 0x%x\n", @@ -651,8 +654,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, } break; default: - dev_err(ac->dev, "command[0x%x] not expecting rsp\n", - result->opcode); + dev_err(ac->dev, "command[0x%x] not expecting rsp status [0x%x]\n", + result->opcode, result->status); break; } @@ -671,6 +674,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, if (ac->io_mode & ASM_SYNC_IO_MODE) { phys_addr_t phys; unsigned long flags; + int token = hdr->token & ASM_WRITE_TOKEN_MASK; + struct audio_buffer *ab; spin_lock_irqsave(&ac->lock, flags); @@ -682,12 +687,13 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, goto done; } - phys = port->buf[hdr->token].phys; + ab = &port->buf[token]; + phys = ab->phys; - if (lower_32_bits(phys) != result->opcode || + if (lower_32_bits(phys) != (result->opcode) || upper_32_bits(phys) != result->status) { dev_err(ac->dev, "Expected addr %pa\n", - &port->buf[hdr->token].phys); + &phys); spin_unlock_irqrestore(&ac->lock, flags); ret = -EINVAL; goto done; @@ -828,21 +834,21 @@ EXPORT_SYMBOL_GPL(q6asm_get_session_id); * @dev: Pointer to asm child device. * @cb: event callback. * @priv: private data associated with this client. - * @stream_id: stream id + * @session_id: session id * @perf_mode: performace mode for this client * * Return: Will be an error pointer on error or a valid audio client * on success. */ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, - void *priv, int stream_id, + void *priv, int session_id, int perf_mode) { struct q6asm *a = dev_get_drvdata(dev->parent); struct audio_client *ac; unsigned long flags; - ac = q6asm_get_audio_client(a, stream_id + 1); + ac = q6asm_get_audio_client(a, session_id + 1); if (ac) { dev_err(dev, "Audio Client already active\n"); return ac; @@ -853,17 +859,15 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, return ERR_PTR(-ENOMEM); spin_lock_irqsave(&a->slock, flags); - a->session[stream_id + 1] = ac; + a->session[session_id + 1] = ac; spin_unlock_irqrestore(&a->slock, flags); - ac->session = stream_id + 1; + ac->session = session_id + 1; ac->cb = cb; ac->dev = dev; ac->q6asm = a; ac->priv = priv; ac->io_mode = ASM_SYNC_IO_MODE; ac->perf_mode = perf_mode; - /* DSP expects stream id from 1 */ - ac->stream_id = 1; ac->adev = a->adev; kref_init(&ac->refcount); @@ -891,7 +895,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) rc = wait_event_timeout(ac->cmd_wait, (ac->result.opcode == hdr->opcode), 5 * HZ); if (!rc) { - dev_err(ac->dev, "CMD timeout\n"); + dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode); rc = -ETIMEDOUT; goto err; } @@ -919,8 +923,9 @@ err: * * Return: Will be an negative value on error or zero on success */ -int q6asm_open_write(struct audio_client *ac, uint32_t format, - u32 codec_profile, uint16_t bits_per_sample) +int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, + uint32_t format, u32 codec_profile, + uint16_t bits_per_sample, bool is_gapless) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; @@ -935,11 +940,13 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, pkt = p; open = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3; open->mode_flags = 0x00; open->mode_flags |= ASM_LEGACY_STREAM_SESSION; + if (is_gapless) + open->mode_flags |= ASM_SHIFT_GAPLESS_MODE_FLAG; /* source endpoint : matrix */ open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX; @@ -998,8 +1005,9 @@ err: } EXPORT_SYMBOL_GPL(q6asm_open_write); -static int __q6asm_run(struct audio_client *ac, uint32_t flags, - uint32_t msw_ts, uint32_t lsw_ts, bool wait) +static int __q6asm_run(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts, + bool wait) { struct asm_session_cmd_run_v2 *run; struct apr_pkt *pkt; @@ -1014,7 +1022,7 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags, pkt = p; run = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2; run->flags = flags; @@ -1042,10 +1050,10 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags, * * Return: Will be an negative value on error or zero on success */ -int q6asm_run(struct audio_client *ac, uint32_t flags, +int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts) { - return __q6asm_run(ac, flags, msw_ts, lsw_ts, true); + return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true); } EXPORT_SYMBOL_GPL(q6asm_run); @@ -1053,16 +1061,17 @@ EXPORT_SYMBOL_GPL(q6asm_run); * q6asm_run_nowait() - start the audio client withou blocking * * @ac: audio client pointer + * @stream_id: stream id * @flags: flags associated with write * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw * * Return: Will be an negative value on error or zero on success */ -int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, - uint32_t msw_ts, uint32_t lsw_ts) +int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts) { - return __q6asm_run(ac, flags, msw_ts, lsw_ts, false); + return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false); } EXPORT_SYMBOL_GPL(q6asm_run_nowait); @@ -1070,6 +1079,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait); * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration * * @ac: audio client pointer + * @stream_id: stream id * @rate: audio sample rate * @channels: number of audio channels. * @channel_map: channel map pointer @@ -1078,6 +1088,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait); * Return: Will be an negative value on error or zero on success */ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t stream_id, uint32_t rate, uint32_t channels, u8 channel_map[PCM_MAX_NUM_CHANNEL], uint16_t bits_per_sample) @@ -1096,7 +1107,7 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1125,8 +1136,8 @@ err: } EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm); - int q6asm_stream_media_format_block_flac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_flac_cfg *cfg) { struct asm_flac_fmt_blk_v2 *fmt; @@ -1142,7 +1153,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1163,6 +1174,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac); int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg) { struct asm_wmastdv9_fmt_blk_v2 *fmt; @@ -1178,7 +1190,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1200,6 +1212,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9); int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg) { struct asm_wmaprov10_fmt_blk_v2 *fmt; @@ -1215,7 +1228,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1238,6 +1251,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); int q6asm_stream_media_format_block_alac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_alac_cfg *cfg) { struct asm_alac_fmt_blk_v2 *fmt; @@ -1253,7 +1267,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1279,6 +1293,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac); int q6asm_stream_media_format_block_ape(struct audio_client *ac, + uint32_t stream_id, struct q6asm_ape_cfg *cfg) { struct asm_ape_fmt_blk_v2 *fmt; @@ -1294,7 +1309,7 @@ int q6asm_stream_media_format_block_ape(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1317,10 +1332,60 @@ int q6asm_stream_media_format_block_ape(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); +static int q6asm_stream_remove_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t cmd, + uint32_t num_samples) +{ + uint32_t *samples; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(uint32_t); + p = kzalloc(pkt_size, GFP_ATOMIC); + if (!p) + return -ENOMEM; + + pkt = p; + samples = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); + + pkt->hdr.opcode = cmd; + *samples = num_samples; + rc = apr_send_pkt(ac->adev, pkt); + if (rc == pkt_size) + rc = 0; + + kfree(pkt); + + return rc; +} + +int q6asm_stream_remove_initial_silence(struct audio_client *ac, + uint32_t stream_id, + uint32_t initial_samples) +{ + return q6asm_stream_remove_silence(ac, stream_id, + ASM_DATA_CMD_REMOVE_INITIAL_SILENCE, + initial_samples); +} +EXPORT_SYMBOL_GPL(q6asm_stream_remove_initial_silence); + +int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t trailing_samples) +{ + return q6asm_stream_remove_silence(ac, stream_id, + ASM_DATA_CMD_REMOVE_TRAILING_SILENCE, + trailing_samples); +} +EXPORT_SYMBOL_GPL(q6asm_stream_remove_trailing_silence); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * * @ac: audio client pointer + * @stream_id: stream id * @rate: audio sample rate * @channels: number of audio channels. * @bits_per_sample: bits per sample @@ -1328,7 +1393,9 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); * Return: Will be an negative value on error or zero on success */ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, - uint32_t rate, uint32_t channels, uint16_t bits_per_sample) + uint32_t stream_id, uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample) { struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg; struct apr_pkt *pkt; @@ -1344,7 +1411,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, pkt = p; enc_cfg = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM; enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2; @@ -1376,10 +1443,11 @@ EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support); * q6asm_read() - read data of period size from audio client * * @ac: audio client pointer + * @stream_id: stream id * * Return: Will be an negative value on error or zero on success */ -int q6asm_read(struct audio_client *ac) +int q6asm_read(struct audio_client *ac, uint32_t stream_id) { struct asm_data_cmd_read_v2 *read; struct audio_port_data *port; @@ -1400,7 +1468,7 @@ int q6asm_read(struct audio_client *ac) spin_lock_irqsave(&ac->lock, flags); port = &ac->port[SNDRV_PCM_STREAM_CAPTURE]; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id); ab = &port->buf[port->dsp_buf]; pkt->hdr.opcode = ASM_DATA_CMD_READ_V2; read->buf_addr_lsw = lower_32_bits(ab->phys); @@ -1428,7 +1496,7 @@ int q6asm_read(struct audio_client *ac) } EXPORT_SYMBOL_GPL(q6asm_read); -static int __q6asm_open_read(struct audio_client *ac, +static int __q6asm_open_read(struct audio_client *ac, uint32_t stream_id, uint32_t format, uint16_t bits_per_sample) { struct asm_stream_cmd_open_read_v3 *open; @@ -1444,7 +1512,7 @@ static int __q6asm_open_read(struct audio_client *ac, pkt = p; open = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3; /* Stream prio : High, provide meta info with encoded frames */ open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX; @@ -1475,15 +1543,16 @@ static int __q6asm_open_read(struct audio_client *ac, * q6asm_open_read() - Open audio client for reading * * @ac: audio client pointer + * @stream_id: stream id * @format: audio sample format * @bits_per_sample: bits per sample * * Return: Will be an negative value on error or zero on success */ -int q6asm_open_read(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample) +int q6asm_open_read(struct audio_client *ac, uint32_t stream_id, + uint32_t format, uint16_t bits_per_sample) { - return __q6asm_open_read(ac, format, bits_per_sample); + return __q6asm_open_read(ac, stream_id, format, bits_per_sample); } EXPORT_SYMBOL_GPL(q6asm_open_read); @@ -1491,6 +1560,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * q6asm_write_async() - non blocking write * * @ac: audio client pointer + * @stream_id: stream id * @len: length in bytes * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw @@ -1498,8 +1568,8 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * * Return: Will be an negative value on error or zero on success */ -int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, - uint32_t lsw_ts, uint32_t wflags) +int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, + uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags) { struct asm_data_cmd_write_v2 *write; struct audio_port_data *port; @@ -1520,10 +1590,10 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, spin_lock_irqsave(&ac->lock, flags); port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id); ab = &port->buf[port->dsp_buf]; - pkt->hdr.token = port->dsp_buf; + pkt->hdr.token = port->dsp_buf | (len << ASM_WRITE_TOKEN_LEN_SHIFT); pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2; write->buf_addr_lsw = lower_32_bits(ab->phys); write->buf_addr_msw = upper_32_bits(ab->phys); @@ -1534,12 +1604,10 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, write->mem_map_handle = ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle; - if (wflags == NO_TIMESTAMP) - write->flags = (wflags & 0x800000FF); - else - write->flags = (0x80000000 | wflags); + write->flags = wflags; - port->dsp_buf++; + if (len) + port->dsp_buf++; if (port->dsp_buf >= port->num_periods) port->dsp_buf = 0; @@ -1567,9 +1635,9 @@ static void q6asm_reset_buf_state(struct audio_client *ac) spin_unlock_irqrestore(&ac->lock, flags); } -static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait) +static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd, + bool wait) { - int stream_id = ac->stream_id; struct apr_pkt pkt; int rc; @@ -1616,13 +1684,14 @@ static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait) * q6asm_cmd() - run cmd on audio client * * @ac: audio client pointer + * @stream_id: stream id * @cmd: command to run on audio client. * * Return: Will be an negative value on error or zero on success */ -int q6asm_cmd(struct audio_client *ac, int cmd) +int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd) { - return __q6asm_cmd(ac, cmd, true); + return __q6asm_cmd(ac, stream_id, cmd, true); } EXPORT_SYMBOL_GPL(q6asm_cmd); @@ -1630,13 +1699,14 @@ EXPORT_SYMBOL_GPL(q6asm_cmd); * q6asm_cmd_nowait() - non blocking, run cmd on audio client * * @ac: audio client pointer + * @stream_id: stream id * @cmd: command to run on audio client. * * Return: Will be an negative value on error or zero on success */ -int q6asm_cmd_nowait(struct audio_client *ac, int cmd) +int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd) { - return __q6asm_cmd(ac, cmd, false); + return __q6asm_cmd(ac, stream_id, cmd, false); } EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 38a207d6cd95..34e01826e549 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -20,6 +20,9 @@ #define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008 #define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009 #define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a +#define ASM_WRITE_TOKEN_MASK GENMASK(15, 0) +#define ASM_WRITE_TOKEN_LEN_MASK GENMASK(31, 16) +#define ASM_WRITE_TOKEN_LEN_SHIFT 16 enum { LEGACY_PCM_MODE = 0, @@ -29,7 +32,7 @@ enum { }; #define MAX_SESSIONS 8 -#define NO_TIMESTAMP 0xFF00 +#define ASM_LAST_BUFFER_FLAG BIT(30) #define FORMAT_LINEAR_PCM 0x0000 struct q6asm_flac_cfg { @@ -93,37 +96,51 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, void *priv, int session_id, int perf_mode); void q6asm_audio_client_free(struct audio_client *ac); -int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, - uint32_t lsw_ts, uint32_t flags); -int q6asm_open_write(struct audio_client *ac, uint32_t format, - u32 codec_profile, uint16_t bits_per_sample); +int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, + uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); +int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, + uint32_t format, u32 codec_profile, + uint16_t bits_per_sample, bool is_gapless); -int q6asm_open_read(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample); +int q6asm_open_read(struct audio_client *ac, uint32_t stream_id, + uint32_t format, uint16_t bits_per_sample); int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, - uint32_t rate, uint32_t channels, uint16_t bits_per_sample); -int q6asm_read(struct audio_client *ac); + uint32_t stream_id, uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample); + +int q6asm_read(struct audio_client *ac, uint32_t stream_id); int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t stream_id, uint32_t rate, uint32_t channels, u8 channel_map[PCM_MAX_NUM_CHANNEL], uint16_t bits_per_sample); int q6asm_stream_media_format_block_flac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_flac_cfg *cfg); int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_alac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_alac_cfg *cfg); int q6asm_stream_media_format_block_ape(struct audio_client *ac, + uint32_t stream_id, struct q6asm_ape_cfg *cfg); -int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, - uint32_t lsw_ts); -int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, - uint32_t lsw_ts); -int q6asm_cmd(struct audio_client *ac, int cmd); -int q6asm_cmd_nowait(struct audio_client *ac, int cmd); +int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags, + uint32_t msw_ts, uint32_t lsw_ts); +int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); +int q6asm_stream_remove_initial_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t initial_samples); +int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t trailing_samples); +int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd); +int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd); int q6asm_get_session_id(struct audio_client *ac); int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 46e50612b92c..07f12f733d8c 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -113,7 +113,19 @@ { mix_name, "QUIN_TDM_TX_4", "QUIN_TDM_TX_4"}, \ { mix_name, "QUIN_TDM_TX_5", "QUIN_TDM_TX_5"}, \ { mix_name, "QUIN_TDM_TX_6", "QUIN_TDM_TX_6"}, \ - { mix_name, "QUIN_TDM_TX_7", "QUIN_TDM_TX_7"} + { mix_name, "QUIN_TDM_TX_7", "QUIN_TDM_TX_7"}, \ + { mix_name, "WSA_CODEC_DMA_TX_0", "WSA_CODEC_DMA_TX_0"}, \ + { mix_name, "WSA_CODEC_DMA_TX_1", "WSA_CODEC_DMA_TX_1"}, \ + { mix_name, "WSA_CODEC_DMA_TX_2", "WSA_CODEC_DMA_TX_2"}, \ + { mix_name, "VA_CODEC_DMA_TX_0", "VA_CODEC_DMA_TX_0"}, \ + { mix_name, "VA_CODEC_DMA_TX_1", "VA_CODEC_DMA_TX_1"}, \ + { mix_name, "VA_CODEC_DMA_TX_2", "VA_CODEC_DMA_TX_2"}, \ + { mix_name, "TX_CODEC_DMA_TX_0", "TX_CODEC_DMA_TX_0"}, \ + { mix_name, "TX_CODEC_DMA_TX_1", "TX_CODEC_DMA_TX_1"}, \ + { mix_name, "TX_CODEC_DMA_TX_2", "TX_CODEC_DMA_TX_2"}, \ + { mix_name, "TX_CODEC_DMA_TX_3", "TX_CODEC_DMA_TX_3"}, \ + { mix_name, "TX_CODEC_DMA_TX_4", "TX_CODEC_DMA_TX_4"}, \ + { mix_name, "TX_CODEC_DMA_TX_5", "TX_CODEC_DMA_TX_5"} #define Q6ROUTING_TX_MIXERS(id) \ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX, \ @@ -268,6 +280,42 @@ msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("QUIN_TDM_TX_7", QUINARY_TDM_TX_7, \ id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("WSA_CODEC_DMA_TX_0", WSA_CODEC_DMA_TX_0, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("WSA_CODEC_DMA_TX_1", WSA_CODEC_DMA_TX_1, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("WSA_CODEC_DMA_TX_2", WSA_CODEC_DMA_TX_2, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("VA_CODEC_DMA_TX_0", VA_CODEC_DMA_TX_0, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("VA_CODEC_DMA_TX_1", VA_CODEC_DMA_TX_1, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("VA_CODEC_DMA_TX_2", VA_CODEC_DMA_TX_2, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_0", TX_CODEC_DMA_TX_0, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_1", TX_CODEC_DMA_TX_1, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_2", TX_CODEC_DMA_TX_2, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_3", TX_CODEC_DMA_TX_3, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_4", TX_CODEC_DMA_TX_4, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TX_CODEC_DMA_TX_5", TX_CODEC_DMA_TX_5, \ + id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), struct session_data { @@ -609,6 +657,36 @@ static const struct snd_kcontrol_new quin_tdm_rx_6_mixer_controls[] = { static const struct snd_kcontrol_new quin_tdm_rx_7_mixer_controls[] = { Q6ROUTING_RX_MIXERS(QUINARY_TDM_RX_7) }; +static const struct snd_kcontrol_new wsa_codec_dma_rx_0_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(WSA_CODEC_DMA_RX_0) }; + +static const struct snd_kcontrol_new wsa_codec_dma_rx_1_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(WSA_CODEC_DMA_RX_1) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_0_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_0) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_1_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_1) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_2_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_2) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_3_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_3) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_4_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_4) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_5_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_5) }; + +static const struct snd_kcontrol_new rxcodec_dma_rx_6_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_6) }; + +static const struct snd_kcontrol_new rx_codec_dma_rx_7_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(RX_CODEC_DMA_RX_7) }; + static const struct snd_kcontrol_new mmul1_mixer_controls[] = { Q6ROUTING_TX_MIXERS(MSM_FRONTEND_DAI_MULTIMEDIA1) }; @@ -819,6 +897,37 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("QUIN_TDM_RX_7 Audio Mixer", SND_SOC_NOPM, 0, 0, quin_tdm_rx_7_mixer_controls, ARRAY_SIZE(quin_tdm_rx_7_mixer_controls)), + + SND_SOC_DAPM_MIXER("WSA_CODEC_DMA_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, + wsa_codec_dma_rx_0_mixer_controls, + ARRAY_SIZE(wsa_codec_dma_rx_0_mixer_controls)), + SND_SOC_DAPM_MIXER("WSA_CODEC_DMA_RX_1 Audio Mixer", SND_SOC_NOPM, 0, 0, + wsa_codec_dma_rx_1_mixer_controls, + ARRAY_SIZE(wsa_codec_dma_rx_1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_0_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_0_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_1 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_1_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_2 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_2_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_2_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_3 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_3_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_3_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_4 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_4_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_4_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_5 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_5_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_5_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_6 Audio Mixer", SND_SOC_NOPM, 0, 0, + rxcodec_dma_rx_6_mixer_controls, + ARRAY_SIZE(rxcodec_dma_rx_6_mixer_controls)), + SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_7 Audio Mixer", SND_SOC_NOPM, 0, 0, + rx_codec_dma_rx_7_mixer_controls, + ARRAY_SIZE(rx_codec_dma_rx_7_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0, mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0, @@ -901,6 +1010,16 @@ static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("QUIN_TDM_RX_5 Audio Mixer", "QUIN_TDM_RX_5"), Q6ROUTING_RX_DAPM_ROUTE("QUIN_TDM_RX_6 Audio Mixer", "QUIN_TDM_RX_6"), Q6ROUTING_RX_DAPM_ROUTE("QUIN_TDM_RX_7 Audio Mixer", "QUIN_TDM_RX_7"), + Q6ROUTING_RX_DAPM_ROUTE("WSA_CODEC_DMA_RX_0 Audio Mixer", "WSA_CODEC_DMA_RX_0"), + Q6ROUTING_RX_DAPM_ROUTE("WSA_CODEC_DMA_RX_1 Audio Mixer", "WSA_CODEC_DMA_RX_1"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_0 Audio Mixer", "RX_CODEC_DMA_RX_0"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_1 Audio Mixer", "RX_CODEC_DMA_RX_1"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_2 Audio Mixer", "RX_CODEC_DMA_RX_2"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_3 Audio Mixer", "RX_CODEC_DMA_RX_3"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_4 Audio Mixer", "RX_CODEC_DMA_RX_4"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_5 Audio Mixer", "RX_CODEC_DMA_RX_5"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_6 Audio Mixer", "RX_CODEC_DMA_RX_6"), + Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_7 Audio Mixer", "RX_CODEC_DMA_RX_7"), Q6ROUTING_TX_DAPM_ROUTE("MultiMedia1 Mixer"), Q6ROUTING_TX_DAPM_ROUTE("MultiMedia2 Mixer"), Q6ROUTING_TX_DAPM_ROUTE("MultiMedia3 Mixer"), diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c new file mode 100644 index 000000000000..8d6c3f84730b --- /dev/null +++ b/sound/soc/qcom/sm8250.c @@ -0,0 +1,277 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020, Linaro Limited + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include "qdsp6/q6afe.h" + +#define MI2S_BCLK_RATE 1536000 +#define DEFAULT_MCLK_RATE 24576000 + +static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + return 0; +} + +static int sm8250_snd_startup(struct snd_pcm_substream *substream) +{ + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF; + + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; + default: + break; + } + return 0; +} + +static const struct snd_soc_ops sm8250_be_ops = { + .startup = sm8250_snd_startup, +}; + +static void sm8250_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int i; + + for_each_card_prelinks(card, i, link) { + if (link->no_pcm == 1) { + link->be_hw_params_fixup = sm8250_be_hw_params_fixup; + link->ops = &sm8250_be_ops; + } + } +} + +int sm8250_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + struct of_phandle_args args; + struct snd_soc_dai_link_component *dlc; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + + for_each_child_of_node(dev->of_node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; + + link->num_cpus = 1; + link->num_platforms = 1; + + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + cpu = of_get_child_by_name(np, "cpu"); + platform = of_get_child_by_name(np, "platform"); + + if (!cpu) { + dev_err(dev, "%s: Can't find cpu DT node\n", link->name); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(card->dev, "%s: error getting cpu phandle\n", link->name); + goto err; + } + link->cpus->of_node = args.np; + link->id = args.args[0]; + + ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); + if (ret) { + dev_err(card->dev, "%s: error getting cpu dai name\n", link->name); + goto err; + } + + if (platform) { + link->platforms->of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platforms->of_node) { + dev_err(card->dev, "%s: platform dai not found\n", link->name); + ret = -EINVAL; + goto err; + } + + dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + link->codecs = dlc; + link->num_codecs = 1; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; + + link->dynamic = 1; + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + + if (q6afe_is_rx_port(link->id)) { + link->dpcm_playback = 1; + link->dpcm_capture = 0; + } else { + link->dpcm_playback = 0; + link->dpcm_capture = 1; + } + + } else { + dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + link->codecs = dlc; + link->num_codecs = 1; + + link->platforms->of_node = link->cpus->of_node; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; + link->dynamic = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + } + + link->ignore_suspend = 1; + link->nonatomic = 1; + link->stream_name = link->name; + link++; + + of_node_put(cpu); + of_node_put(platform); + } + + return 0; +err: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} + +static int sm8250_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + int ret; + + card = kzalloc(sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->dev = dev; + dev_set_drvdata(dev, card); + ret = sm8250_snd_parse_of(card); + if (ret) + goto err; + + sm8250_add_be_ops(card); + ret = snd_soc_register_card(card); + if (ret) + goto err_card_register; + + return 0; + +err_card_register: + kfree(card->dai_link); +err: + kfree(card); + return ret; +} + +static int sm8250_platform_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(card); + + return 0; +} + +static const struct of_device_id snd_sm8250_dt_match[] = { + {.compatible = "qcom,sm8250-sndcard"}, + {} +}; + +MODULE_DEVICE_TABLE(of, snd_sm8250_dt_match); + +static struct platform_driver snd_sm8250_driver = { + .probe = sm8250_platform_probe, + .remove = sm8250_platform_remove, + .driver = { + .name = "snd-sm8250", + .of_match_table = snd_sm8250_dt_match, + }, +}; +module_platform_driver(snd_sm8250_driver); +MODULE_AUTHOR("Srinivas Kandagatla <srinivas.kandagatla@linaro.org"); +MODULE_DESCRIPTION("SM8250 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 4984b6a2c370..e549e0197aca 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -518,6 +518,34 @@ out: return ret; } +static int soc_compr_set_codec_params(struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int i, ret; + + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = snd_soc_dai_compr_set_codec_params(cpu_dai, cstream, codec); + if (ret < 0) + goto err; + + for_each_rtd_components(rtd, i, component) { + if (!component->driver->compress_ops || + !component->driver->compress_ops->set_codec_params) + continue; + + ret = component->driver->compress_ops->set_codec_params(component, cstream, + codec); + break; + } +err: + mutex_unlock(&rtd->card->pcm_mutex); + return ret; +} + static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { @@ -728,6 +756,7 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, @@ -744,6 +773,7 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .free = soc_compr_free_fe, .set_params = soc_compr_set_params_fe, .get_params = soc_compr_get_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .trigger = soc_compr_trigger_fe, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index b05e18b63a1c..06481d0278b8 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -594,6 +594,20 @@ int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_compr_get_params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ int ret = 0; + + if (dai->driver->cops && + dai->driver->cops->set_codec_params) + ret = dai->driver->cops->set_codec_params(cstream, codec, dai); + + return soc_dai_ret(dai, ret); + +} +EXPORT_SYMBOL_GPL(snd_soc_dai_compr_set_codec_params); + int snd_soc_dai_compr_ack(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, size_t bytes) |