[alsa-devel] [PATCH 0/8] ASoC: qdsp6: db820c: Add support for external and bluetooth audio
This patch set implements PCM audio support in qdsp6 and PCM and MI2S in apq8096/db820c to enable use of bluetooth audio codec and external MI2S port on db820c.
The db820c uses qca6174a for bluetooth, which by default is configured to use what qualcomm refers to as "PCM" format, which is a variation of TDM.
CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
Adam Serbinski (8): ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding ASoC: qdsp6: q6afe: add support to pcm ports ASoC: qdsp6: q6afe-dai: add support to pcm port dais ASoC: qdsp6: q6routing: add pcm port routing ASoC: qcom: apq8096: add support for primary and quaternary I2S/PCM ASoC: qcom/common: Use snd-soc-dummy-dai when codec is not specified dts: msm8996/db820c: enable primary pcm and quaternary i2s ASoC: qcom: apq8096: add kcontrols to set PCM rate
arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 113 +++++++++ arch/arm64/boot/dts/qcom/msm8996-pins.dtsi | 162 ++++++++++++ include/dt-bindings/sound/qcom,q6afe.h | 8 + sound/soc/qcom/apq8096.c | 172 +++++++++++-- sound/soc/qcom/common.c | 22 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 198 ++++++++++++++- sound/soc/qcom/qdsp6/q6afe.c | 246 +++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 9 +- sound/soc/qcom/qdsp6/q6routing.c | 44 ++++ 9 files changed, 953 insertions(+), 21 deletions(-)
This patch adds bindings required for PCM ports on AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- include/dt-bindings/sound/qcom,q6afe.h | 8 ++++++++ 1 file changed, 8 insertions(+)
diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index 1df06f8ad5c3..f3a435a112cb 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -107,6 +107,14 @@ #define QUINARY_TDM_RX_7 102 #define QUINARY_TDM_TX_7 103 #define DISPLAY_PORT_RX 104 +#define PRIMARY_PCM_RX 105 +#define PRIMARY_PCM_TX 106 +#define SECONDARY_PCM_RX 107 +#define SECONDARY_PCM_TX 108 +#define TERTIARY_PCM_RX 109 +#define TERTIARY_PCM_TX 110 +#define QUATERNARY_PCM_RX 111 +#define QUATERNARY_PCM_TX 112
#endif /* __DT_BINDINGS_Q6_AFE_H__ */
This patch adds support to pcm ports in AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6afe.c | 246 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 9 +- 2 files changed, 254 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..b53ad14a78fd 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -40,6 +40,7 @@
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 #define AFE_PARAM_ID_I2S_CONFIG 0x0001020D +#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297
@@ -117,6 +118,15 @@ #define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006 #define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A +#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B +#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C +#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D +#define AFE_PORT_ID_TERTIARY_PCM_RX 0x1012 +#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013 +#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014 +#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015 + /* Start of the range of port IDs for TDM devices. */ #define AFE_PORT_ID_TDM_PORT_RANGE_START 0x9000
@@ -421,6 +431,166 @@ struct afe_digital_clk_cfg { u16 reserved; } __packed;
+#define AFE_API_VERSION_PCM_CONFIG 0x1 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an external source. + */ + +#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an internal source. + */ +#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1 +/* Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use + * short synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_PCM 0x0 +/* + * Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use long + * synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_AUX 0x1 +/* + * Enumeration for setting the PCM configuration frame to 8. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0 +/* + * Enumeration for setting the PCM configuration frame to 16. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1 + +/* Enumeration for setting the PCM configuration frame to 32.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2 + +/* Enumeration for setting the PCM configuration frame to 64.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3 + +/* Enumeration for setting the PCM configuration frame to 128.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4 + +/* Enumeration for setting the PCM configuration frame to 256.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5 + +/* Enumeration for setting the PCM configuration + * quantype parameter to A-law with no padding. + */ +#define AFE_PORT_PCM_ALAW_NOPADDING 0x0 + +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with no padding. + */ +#define AFE_PORT_PCM_MULAW_NOPADDING 0x1 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with no padding. + */ +#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2 +/* Enumeration for setting the PCM configuration quantype + * parameter to A-law with padding. + */ +#define AFE_PORT_PCM_ALAW_PADDING 0x3 +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with padding. + */ +#define AFE_PORT_PCM_MULAW_PADDING 0x4 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with padding. + */ +#define AFE_PORT_PCM_LINEAR_PADDING 0x5 +/* Enumeration for disabling the PCM configuration + * ctrl_data_out_enable parameter. + * The PCM block is the only master. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0 +/* + * Enumeration for enabling the PCM configuration + * ctrl_data_out_enable parameter. The PCM block shares + * the signal with other masters. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1 + +/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's + * (PCM configuration parameter). + */ + +struct afe_param_id_pcm_cfg { + u32 pcm_cfg_minor_version; +/* Minor version used for tracking the version of the AUX PCM + * configuration interface. + * Supported values: #AFE_API_VERSION_PCM_CONFIG + */ + + u16 aux_mode; +/* PCM synchronization setting. + * Supported values: + * - #AFE_PORT_PCM_AUX_MODE_PCM + * - #AFE_PORT_PCM_AUX_MODE_AUX + */ + + u16 sync_src; +/* Synchronization source. + * Supported values: + * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL + * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL + */ + + u16 frame_setting; +/* Number of bits per frame. + * Supported values: + * - #AFE_PORT_PCM_BITS_PER_FRAME_8 + * - #AFE_PORT_PCM_BITS_PER_FRAME_16 + * - #AFE_PORT_PCM_BITS_PER_FRAME_32 + * - #AFE_PORT_PCM_BITS_PER_FRAME_64 + * - #AFE_PORT_PCM_BITS_PER_FRAME_128 + * - #AFE_PORT_PCM_BITS_PER_FRAME_256 + */ + + u16 quantype; +/* PCM quantization type. + * Supported values: + * - #AFE_PORT_PCM_ALAW_NOPADDING + * - #AFE_PORT_PCM_MULAW_NOPADDING + * - #AFE_PORT_PCM_LINEAR_NOPADDING + * - #AFE_PORT_PCM_ALAW_PADDING + * - #AFE_PORT_PCM_MULAW_PADDING + * - #AFE_PORT_PCM_LINEAR_PADDING + */ + + u16 ctrl_data_out_enable; +/* Specifies whether the PCM block shares the data-out + * signal to the drive with other masters. + * Supported values: + * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE + * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE + */ + u16 reserved; + /* This field must be set to zero. */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to 4 + */ + + u16 slot_number_mapping[4]; +/* Specifies the slot number for the each channel in + * multi channel scenario. + * Supported values: 1 to 32 + */ +} __packed; + struct afe_param_id_i2s_cfg { u32 i2s_cfg_minor_version; u16 bit_width; @@ -452,6 +622,7 @@ 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_pcm_cfg pcm_cfg; struct afe_param_id_tdm_cfg tdm_cfg; } __packed;
@@ -707,6 +878,22 @@ 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}, + [PRIMARY_PCM_RX] = { AFE_PORT_ID_PRIMARY_PCM_RX, + PRIMARY_PCM_RX, 1, 1}, + [PRIMARY_PCM_TX] = { AFE_PORT_ID_PRIMARY_PCM_TX, + PRIMARY_PCM_RX, 0, 1}, + [SECONDARY_PCM_RX] = { AFE_PORT_ID_SECONDARY_PCM_RX, + SECONDARY_PCM_RX, 1, 1}, + [SECONDARY_PCM_TX] = { AFE_PORT_ID_SECONDARY_PCM_TX, + SECONDARY_PCM_TX, 0, 1}, + [TERTIARY_PCM_RX] = { AFE_PORT_ID_TERTIARY_PCM_RX, + TERTIARY_PCM_RX, 1, 1}, + [TERTIARY_PCM_TX] = { AFE_PORT_ID_TERTIARY_PCM_TX, + TERTIARY_PCM_TX, 0, 1}, + [QUATERNARY_PCM_RX] = { AFE_PORT_ID_QUATERNARY_PCM_RX, + QUATERNARY_PCM_RX, 1, 1}, + [QUATERNARY_PCM_TX] = { AFE_PORT_ID_QUATERNARY_PCM_TX, + QUATERNARY_PCM_TX, 0, 1}, };
static void q6afe_port_free(struct kref *ref) @@ -993,6 +1180,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, break; 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: + /* TDM cases overlap with PCM */ case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET; cset.clk_id = clk_id; @@ -1145,6 +1333,54 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+/** + * q6afe_pcm_port_prepare() - Prepare pcm afe port. + * + * @port: Instance of afe port + * @cfg: PCM configuration for the afe port + * + */ +int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->pcm_cfg.pcm_cfg_minor_version = AFE_API_VERSION_PCM_CONFIG; + pcfg->pcm_cfg.aux_mode = AFE_PORT_PCM_AUX_MODE_PCM; + + switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CPU is slave */ + pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL; + break; + default: + break; + } + + switch (cfg->sample_rate) { + case 8000: + pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128; + break; + case 16000: + pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64; + break; + } + pcfg->pcm_cfg.quantype = AFE_PORT_PCM_LINEAR_NOPADDING; + pcfg->pcm_cfg.ctrl_data_out_enable = AFE_PORT_PCM_CTRL_DATA_OE_DISABLE; + pcfg->pcm_cfg.reserved = 0; + pcfg->pcm_cfg.sample_rate = cfg->sample_rate; + + /* 16 bit mono */ + pcfg->pcm_cfg.bit_width = 16; + pcfg->pcm_cfg.num_channels = 1; + pcfg->pcm_cfg.slot_number_mapping[0] = 1; + + return 0; +} +EXPORT_SYMBOL_GPL(q6afe_pcm_port_prepare); + /** * q6afe_i2s_port_prepare() - Prepare i2s afe port. * @@ -1417,6 +1653,16 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_QUATERNARY_MI2S_TX: cfg_type = AFE_PARAM_ID_I2S_CONFIG; break; + case AFE_PORT_ID_PRIMARY_PCM_RX: + case AFE_PORT_ID_PRIMARY_PCM_TX: + case AFE_PORT_ID_SECONDARY_PCM_RX: + case AFE_PORT_ID_SECONDARY_PCM_TX: + case AFE_PORT_ID_TERTIARY_PCM_RX: + case AFE_PORT_ID_TERTIARY_PCM_TX: + case AFE_PORT_ID_QUATERNARY_PCM_RX: + case AFE_PORT_ID_QUATERNARY_PCM_TX: + cfg_type = AFE_PARAM_ID_PCM_CONFIG; + break; case AFE_PORT_ID_PRIMARY_TDM_RX ... AFE_PORT_ID_QUINARY_TDM_TX_7: cfg_type = AFE_PARAM_ID_TDM_CONFIG; break; diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index c7ed5422baff..c832be6d0ff5 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 113
#define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -170,6 +170,11 @@ struct q6afe_i2s_cfg { int fmt; };
+struct q6afe_pcm_cfg { + u32 sample_rate; + int fmt; +}; + struct q6afe_tdm_cfg { u16 num_channels; u32 sample_rate; @@ -188,6 +193,7 @@ struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; struct q6afe_slim_cfg slim; struct q6afe_i2s_cfg i2s_cfg; + struct q6afe_pcm_cfg pcm_cfg; struct q6afe_tdm_cfg tdm; };
@@ -203,6 +209,7 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, 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); +int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg); void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
This patch adds support of AFE DAI for PCM port.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6afe-dai.c | 198 ++++++++++++++++++++++++++++++- 1 file changed, 197 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..23b29591ef47 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -151,6 +151,28 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; }
+static int q6pcm_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_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + + pcm->sample_rate = params_rate(params); + + return 0; +} + +static int q6pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + + pcm->fmt = fmt; + + return 0; +} + static int q6i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -358,6 +380,15 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, return rc; } break; + case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX: + rc = q6afe_pcm_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].pcm_cfg); + if (rc < 0) { + dev_err(dai->dev, "fail to prepare AFE port %x\n", + dai->id); + return rc; + } + break; case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7: q6afe_tdm_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].tdm); @@ -429,11 +460,32 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR: + case Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR: case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); + } + + return 0; +} + +static int q6afe_tdm_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_port *port = dai_data->port[dai->id]; + + switch (clk_id) { + case LPAIF_DIG_CLK: + return q6afe_port_set_sysclk(port, clk_id, 0, 5, freq, dir); + case LPAIF_BIT_CLK: + case LPAIF_OSR_CLK: + return q6afe_port_set_sysclk(port, clk_id, + Q6AFE_LPASS_CLK_SRC_INTERNAL, + Q6AFE_LPASS_CLK_ROOT_DEFAULT, + freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO, @@ -468,6 +520,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
+ {"Primary PCM Playback", NULL, "PRI_PCM_RX"}, + {"Secondary PCM Playback", NULL, "SEC_PCM_RX"}, + {"Tertiary PCM Playback", NULL, "TERT_PCM_RX"}, + {"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"}, + {"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"}, {"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"}, {"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"}, @@ -562,6 +619,11 @@ 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"}, + + {"PRI_PCM_TX", NULL, "Primary PCM Capture"}, + {"SEC_PCM_TX", NULL, "Secondary PCM Capture"}, + {"TERT_PCM_TX", NULL, "Tertiary PCM Capture"}, + {"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"}, };
static const struct snd_soc_dai_ops q6hdmi_ops = { @@ -578,6 +640,14 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .set_sysclk = q6afe_mi2s_set_sysclk, };
+static const struct snd_soc_dai_ops q6pcm_ops = { + .prepare = q6afe_dai_prepare, + .hw_params = q6pcm_hw_params, + .set_fmt = q6pcm_set_fmt, + .shutdown = q6afe_dai_shutdown, + .set_sysclk = q6afe_mi2s_set_sysclk, +}; + static const struct snd_soc_dai_ops q6slim_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6slim_hw_params, @@ -588,7 +658,7 @@ static const struct snd_soc_dai_ops q6slim_ops = { static const struct snd_soc_dai_ops q6tdm_ops = { .prepare = q6afe_dai_prepare, .shutdown = q6afe_dai_shutdown, - .set_sysclk = q6afe_mi2s_set_sysclk, + .set_sysclk = q6afe_tdm_set_sysclk, .set_tdm_slot = q6tdm_set_tdm_slot, .set_channel_map = q6tdm_set_channel_map, .hw_params = q6tdm_hw_params, @@ -1012,6 +1082,115 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .ops = &q6i2s_ops, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Primary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = PRIMARY_PCM_RX, + .name = "PRI_PCM_RX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Primary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = PRIMARY_PCM_TX, + .name = "PRI_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Secondary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "SEC_PCM_RX", + .id = SECONDARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Secondary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = SECONDARY_PCM_TX, + .name = "SEC_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Tertiary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "TERT_PCM_RX", + .id = TERTIARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Tertiary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = TERTIARY_PCM_TX, + .name = "TERT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Quaternary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "QUAT_PCM_RX", + .id = QUATERNARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Quaternary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = QUATERNARY_PCM_TX, + .name = "QUAT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, }, Q6AFE_TDM_PB_DAI("Primary", 0, PRIMARY_TDM_RX_0), Q6AFE_TDM_PB_DAI("Primary", 1, PRIMARY_TDM_RX_1), @@ -1169,6 +1348,23 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
This patch adds support to PCM_PORT mixers required to select path between ASM stream and AFE ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a..3a81d2161707 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -67,6 +67,10 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "PRI_PCM_TX", "PRI_PCM_TX" }, \ + { mix_name, "SEC_PCM_TX", "SEC_PCM_TX" }, \ + { mix_name, "TERT_PCM_TX", "TERT_PCM_TX" }, \ + { mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" }, \ { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ @@ -128,6 +132,18 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ @@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) };
+static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) }; + +static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) }; + +static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) }; + +static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) }; + static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) };
@@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0, tertiary_mi2s_rx_mixer_controls, ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + primary_pcm_rx_mixer_controls, + ARRAY_SIZE(primary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + secondary_pcm_rx_mixer_controls, + ARRAY_SIZE(secondary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + tertiary_pcm_rx_mixer_controls, + ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + quaternary_pcm_rx_mixer_controls, + ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)), SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, pri_tdm_rx_0_mixer_controls, ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)), @@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"), + Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer", "PRIMARY_TDM_RX_0"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer",
This adds support to primary and quarternary I2S and PCM ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/apq8096.c | 86 +++++++++++++++++++++++++++++++++------- 1 file changed, 71 insertions(+), 15 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 94363fd6846a..1edcaa15234f 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -8,24 +8,13 @@ #include <sound/soc-dapm.h> #include <sound/pcm.h> #include "common.h" +#include "qdsp6/q6afe.h"
#define SLIM_MAX_TX_PORTS 16 #define SLIM_MAX_RX_PORTS 16 #define WCD9335_DEFAULT_MCLK_RATE 9600000 - -static int apq8096_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; -} +#define MI2S_BCLK_RATE 1536000 +#define PCM_BCLK_RATE 1024000
static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -33,10 +22,32 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + 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); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0;
+ switch (cpu_dai->id) { + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + rate->min = 16000; + rate->max = 16000; + channels->min = 1; + channels->max = 1; + break; + default: + rate->min = 48000; + rate->max = 48000; + channels->min = 1; + channels->max = 2; + break; + } + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -60,8 +71,54 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; }
+static int msm_snd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + 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, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_MI2S_RX: + case QUATERNARY_MI2S_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + default: + return -1; + } + return 0; +} + static struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, + .startup = msm_snd_startup, };
static int apq8096_init(struct snd_soc_pcm_runtime *rtd) @@ -96,7 +153,6 @@ static void apq8096_add_be_ops(struct snd_soc_card *card)
for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) { - link->be_hw_params_fixup = apq8096_be_hw_params_fixup; link->init = apq8096_init; link->ops = &apq8096_ops; }
When not specifying a codec, use snd-soc-dummy-dai. This supports the case where a fixed configuration codec is attached, such as bluetooth hfp.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/common.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 6c20bdd850f3..aa2f2238aca0 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -84,7 +84,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; }
- if (codec && platform) { + if (platform) { link->platforms->of_node = of_parse_phandle(platform, "sound-dai", 0); @@ -94,10 +94,22 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; }
- ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "%s: codec dai not found\n", link->name); - goto err; + if (codec) { + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "%s: codec dai not found\n", link->name); + goto err; + } + } else { + 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->no_pcm = 1; link->ignore_pmdown_time = 1;
This patch adds support to primary pcm and quaternary i2s ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 113 +++++++++++++ arch/arm64/boot/dts/qcom/msm8996-pins.dtsi | 162 +++++++++++++++++++ 2 files changed, 275 insertions(+)
diff --git a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi index dba3488492f1..4149ac4147a0 100644 --- a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi +++ b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi @@ -683,8 +683,31 @@ }; };
+/* PRI I2S on QCA6174 and QUAT I2S on LS each uses 2 I2S SD Lines for audio */ +&q6afedai { + pi2s@16 { + reg = <16>; + qcom,sd-lines = <1>; + }; + pi2s@17 { + reg = <17>; + qcom,sd-lines = <0>; + }; + qi2s@22 { + reg = <22>; + qcom,sd-lines = <0>; + }; + qi2s@23 { + reg = <23>; + qcom,sd-lines = <1>; + }; +}; + &sound { compatible = "qcom,apq8096-sndcard"; + pinctrl-0 = <&quat_mi2s_active &quat_mi2s_sd0_active &quat_mi2s_sd1_active &pri_mi2s_active &pri_mi2s_sd0_active &pri_mi2s_sd1_active>; + pinctrl-names = "default"; + model = "DB820c"; audio-routing = "RX_BIAS", "MCLK";
@@ -709,6 +732,41 @@ }; };
+ mm4-dai-link { + link-name = "MultiMedia4"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA4>; + }; + }; + + mm5-dai-link { + link-name = "MultiMedia5"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA5>; + }; + }; + + mm6-dai-link { + link-name = "MultiMedia6"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA6>; + }; + }; + + mm7-dai-link { + link-name = "MultiMedia7"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA7>; + }; + }; + + mm8-dai-link { + link-name = "MultiMedia8"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA8>; + }; + }; + hdmi-dai-link { link-name = "HDMI"; cpu { @@ -753,4 +811,59 @@ sound-dai = <&wcd9335 1>; }; }; + + scoplay-dai-link { + link-name = "SCO-PCM-Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_PCM_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; + + scocap-dai-link { + link-name = "SCO-PCM-Capture"; + cpu { + sound-dai = <&q6afedai PRIMARY_PCM_TX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; + + mi2splay-dai-link { + link-name = "QUAT-MI2S-Playback"; + cpu { + sound-dai = <&q6afedai QUATERNARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + +// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm5142_4c>, <&pcm5142_4d>; +// }; + + }; + + mi2scap-dai-link { + link-name = "QUAT-MI2S-Capture"; + cpu { + sound-dai = <&q6afedai QUATERNARY_MI2S_TX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + +// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm1865>; +// }; + }; }; diff --git a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi index ac1ede579361..e8221c4d05f7 100644 --- a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi +++ b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi @@ -288,6 +288,168 @@ }; };
+ pri_mi2s_active: pri_mi2s_active { + mux { + pins = "gpio65", "gpio66"; + function = "pri_mi2s"; + }; + config { + pins = "gpio65", "gpio66"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + output-high; + }; + }; + + pri_mi2s_sleep: pri_mi2s_sleep { + mux { + pins = "gpio65", "gpio66"; + function = "gpio"; + }; + + config { + pins = "gpio65", "gpio66"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd0_sleep: pri_mi2s_sd0_sleep { + mux { + pins = "gpio67"; + function = "gpio"; + }; + + config { + pins = "gpio67"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd0_active: pri_mi2s_sd0_active { + mux { + pins = "gpio67"; + function = "pri_mi2s"; + }; + + config { + pins = "gpio67"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + pri_mi2s_sd1_sleep: pri_mi2s_sd1_sleep { + mux { + pins = "gpio68"; + function = "gpio"; + }; + + config { + pins = "gpio68"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd1_active: pri_mi2s_sd1_active { + mux { + pins = "gpio68"; + function = "pri_mi2s"; + }; + + config { + pins = "gpio68"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + quat_mi2s_active: quat_mi2s_active { + mux { + pins = "gpio58", "gpio59"; + function = "qua_mi2s"; + }; + config { + pins = "gpio58", "gpio59"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + output-high; + }; + }; + + quat_mi2s_sleep: quat_mi2s_sleep { + mux { + pins = "gpio58", "gpio59"; + function = "gpio"; + }; + + config { + pins = "gpio58", "gpio59"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd0_sleep: quat_mi2s_sd0_sleep { + mux { + pins = "gpio60"; + function = "gpio"; + }; + + config { + pins = "gpio60"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd0_active: quat_mi2s_sd0_active { + mux { + pins = "gpio60"; + function = "qua_mi2s"; + }; + + config { + pins = "gpio60"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + quat_mi2s_sd1_sleep: quat_mi2s_sd1_sleep { + mux { + pins = "gpio61"; + function = "gpio"; + }; + + config { + pins = "gpio61"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd1_active: quat_mi2s_sd1_active { + mux { + pins = "gpio61"; + function = "qua_mi2s"; + }; + + config { + pins = "gpio61"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + sdc2_clk_on: sdc2_clk_on { config { pins = "sdc2_clk";
On Fri 07 Feb 12:50 PST 2020, Adam Serbinski wrote:
Please make subject
"arm64: dts: qcom: db820c: Enable primary PCM and quaternary I2S"
Regards, Bjorn
This patch adds support to primary pcm and quaternary i2s ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 113 +++++++++++++ arch/arm64/boot/dts/qcom/msm8996-pins.dtsi | 162 +++++++++++++++++++ 2 files changed, 275 insertions(+)
diff --git a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi index dba3488492f1..4149ac4147a0 100644 --- a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi +++ b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi @@ -683,8 +683,31 @@ }; };
+/* PRI I2S on QCA6174 and QUAT I2S on LS each uses 2 I2S SD Lines for audio */ +&q6afedai {
- pi2s@16 {
reg = <16>;
qcom,sd-lines = <1>;
- };
- pi2s@17 {
reg = <17>;
qcom,sd-lines = <0>;
- };
- qi2s@22 {
reg = <22>;
qcom,sd-lines = <0>;
- };
- qi2s@23 {
reg = <23>;
qcom,sd-lines = <1>;
- };
+};
&sound { compatible = "qcom,apq8096-sndcard";
- pinctrl-0 = <&quat_mi2s_active &quat_mi2s_sd0_active &quat_mi2s_sd1_active &pri_mi2s_active &pri_mi2s_sd0_active &pri_mi2s_sd1_active>;
- pinctrl-names = "default";
- model = "DB820c"; audio-routing = "RX_BIAS", "MCLK";
@@ -709,6 +732,41 @@ }; };
- mm4-dai-link {
link-name = "MultiMedia4";
cpu {
sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA4>;
};
- };
- mm5-dai-link {
link-name = "MultiMedia5";
cpu {
sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA5>;
};
- };
- mm6-dai-link {
link-name = "MultiMedia6";
cpu {
sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA6>;
};
- };
- mm7-dai-link {
link-name = "MultiMedia7";
cpu {
sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA7>;
};
- };
- mm8-dai-link {
link-name = "MultiMedia8";
cpu {
sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA8>;
};
- };
- hdmi-dai-link { link-name = "HDMI"; cpu {
@@ -753,4 +811,59 @@ sound-dai = <&wcd9335 1>; }; };
- scoplay-dai-link {
link-name = "SCO-PCM-Playback";
cpu {
sound-dai = <&q6afedai PRIMARY_PCM_RX>;
};
platform {
sound-dai = <&q6routing>;
};
- };
- scocap-dai-link {
link-name = "SCO-PCM-Capture";
cpu {
sound-dai = <&q6afedai PRIMARY_PCM_TX>;
};
platform {
sound-dai = <&q6routing>;
};
- };
- mi2splay-dai-link {
link-name = "QUAT-MI2S-Playback";
cpu {
sound-dai = <&q6afedai QUATERNARY_MI2S_RX>;
};
platform {
sound-dai = <&q6routing>;
};
+// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm5142_4c>, <&pcm5142_4d>; +// };
- };
- mi2scap-dai-link {
link-name = "QUAT-MI2S-Capture";
cpu {
sound-dai = <&q6afedai QUATERNARY_MI2S_TX>;
};
platform {
sound-dai = <&q6routing>;
};
+// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm1865>; +// };
- };
}; diff --git a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi index ac1ede579361..e8221c4d05f7 100644 --- a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi +++ b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi @@ -288,6 +288,168 @@ }; };
- pri_mi2s_active: pri_mi2s_active {
mux {
pins = "gpio65", "gpio66";
function = "pri_mi2s";
};
config {
pins = "gpio65", "gpio66";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
output-high;
};
- };
- pri_mi2s_sleep: pri_mi2s_sleep {
mux {
pins = "gpio65", "gpio66";
function = "gpio";
};
config {
pins = "gpio65", "gpio66";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- pri_mi2s_sd0_sleep: pri_mi2s_sd0_sleep {
mux {
pins = "gpio67";
function = "gpio";
};
config {
pins = "gpio67";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- pri_mi2s_sd0_active: pri_mi2s_sd0_active {
mux {
pins = "gpio67";
function = "pri_mi2s";
};
config {
pins = "gpio67";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
};
- };
- pri_mi2s_sd1_sleep: pri_mi2s_sd1_sleep {
mux {
pins = "gpio68";
function = "gpio";
};
config {
pins = "gpio68";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- pri_mi2s_sd1_active: pri_mi2s_sd1_active {
mux {
pins = "gpio68";
function = "pri_mi2s";
};
config {
pins = "gpio68";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
};
- };
- quat_mi2s_active: quat_mi2s_active {
mux {
pins = "gpio58", "gpio59";
function = "qua_mi2s";
};
config {
pins = "gpio58", "gpio59";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
output-high;
};
- };
- quat_mi2s_sleep: quat_mi2s_sleep {
mux {
pins = "gpio58", "gpio59";
function = "gpio";
};
config {
pins = "gpio58", "gpio59";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- quat_mi2s_sd0_sleep: quat_mi2s_sd0_sleep {
mux {
pins = "gpio60";
function = "gpio";
};
config {
pins = "gpio60";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- quat_mi2s_sd0_active: quat_mi2s_sd0_active {
mux {
pins = "gpio60";
function = "qua_mi2s";
};
config {
pins = "gpio60";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
};
- };
- quat_mi2s_sd1_sleep: quat_mi2s_sd1_sleep {
mux {
pins = "gpio61";
function = "gpio";
};
config {
pins = "gpio61";
drive-strength = <2>; /* 2 mA */
bias-pull-down; /* PULL DOWN */
input-enable;
};
- };
- quat_mi2s_sd1_active: quat_mi2s_sd1_active {
mux {
pins = "gpio61";
function = "qua_mi2s";
};
config {
pins = "gpio61";
drive-strength = <8>; /* 8 mA */
bias-disable; /* NO PULL */
};
- };
- sdc2_clk_on: sdc2_clk_on { config { pins = "sdc2_clk";
-- 2.21.1
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/apq8096.c | 92 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 90 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1edcaa15234f..882f2c456321 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -16,6 +16,9 @@ #define MI2S_BCLK_RATE 1536000 #define PCM_BCLK_RATE 1024000
+static int pri_pcm_sample_rate = 16000; +static int quat_pcm_sample_rate = 16000; + static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -33,10 +36,15 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, switch (cpu_dai->id) { case PRIMARY_PCM_RX: case PRIMARY_PCM_TX: + rate->min = pri_pcm_sample_rate; + rate->max = pri_pcm_sample_rate; + channels->min = 1; + channels->max = 1; + break; case QUATERNARY_PCM_RX: case QUATERNARY_PCM_TX: - rate->min = 16000; - rate->max = 16000; + rate->min = quat_pcm_sample_rate; + rate->max = quat_pcm_sample_rate; channels->min = 1; channels->max = 1; break; @@ -121,6 +129,83 @@ static struct snd_soc_ops apq8096_ops = { .startup = msm_snd_startup, };
+static char const *pcm_sample_rate_text[] = {"8 kHz", "16 kHz"}; +static const struct soc_enum pcm_snd_enum = + SOC_ENUM_SINGLE_EXT(2, pcm_sample_rate_text); + +static int get_sample_rate_idx(int sample_rate) +{ + int sample_rate_idx = 0; + + switch (sample_rate) { + case 8000: + sample_rate_idx = 0; + break; + case 16000: + default: + sample_rate_idx = 1; + break; + } + + return sample_rate_idx; +} + +static int pri_pcm_sample_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + get_sample_rate_idx(pri_pcm_sample_rate); + return 0; +} + +static int quat_pcm_sample_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + get_sample_rate_idx(quat_pcm_sample_rate); + return 0; +} + +static int get_sample_rate(int idx) +{ + int sample_rate_val = 0; + + switch (idx) { + case 0: + sample_rate_val = 8000; + break; + case 1: + default: + sample_rate_val = 16000; + break; + } + + return sample_rate_val; +} + +static int pri_pcm_sample_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + pri_pcm_sample_rate = + get_sample_rate(ucontrol->value.integer.value[0]); + return 0; +} + +static int quat_pcm_sample_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + quat_pcm_sample_rate = + get_sample_rate(ucontrol->value.integer.value[0]); + return 0; +} + +static const struct snd_kcontrol_new card_controls[] = { + SOC_ENUM_EXT("PRI_PCM SampleRate", pcm_snd_enum, + pri_pcm_sample_rate_get, pri_pcm_sample_rate_put), + SOC_ENUM_EXT("QUAT_PCM SampleRate", pcm_snd_enum, + quat_pcm_sample_rate_get, quat_pcm_sample_rate_put), +}; + static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; @@ -182,6 +267,9 @@ static int apq8096_platform_probe(struct platform_device *pdev) if (ret) goto err_card_register;
+ snd_soc_add_card_controls(card, card_controls, + ARRAY_SIZE(card_controls)); + return 0;
err_card_register:
Changes from V1:
Rename patch: from: dts: msm8996/db820c: enable primary pcm and quaternary i2s to: dts: qcom: db820c: Enable primary PCM and quaternary I2S
CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
Adam Serbinski (8): ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding ASoC: qdsp6: q6afe: add support to pcm ports ASoC: qdsp6: q6afe-dai: add support to pcm port dais ASoC: qdsp6: q6routing: add pcm port routing ASoC: qcom: apq8096: add support for primary and quaternary I2S/PCM ASoC: qcom/common: Use snd-soc-dummy-dai when codec is not specified arm64: dts: qcom: db820c: Enable primary PCM and quaternary I2S ASoC: qcom: apq8096: add kcontrols to set PCM rate
arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 113 +++++++++ arch/arm64/boot/dts/qcom/msm8996-pins.dtsi | 162 ++++++++++++ include/dt-bindings/sound/qcom,q6afe.h | 8 + sound/soc/qcom/apq8096.c | 172 +++++++++++-- sound/soc/qcom/common.c | 22 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 198 ++++++++++++++- sound/soc/qcom/qdsp6/q6afe.c | 246 +++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 9 +- sound/soc/qcom/qdsp6/q6routing.c | 44 ++++ 9 files changed, 953 insertions(+), 21 deletions(-)
This patch adds bindings required for PCM ports on AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- include/dt-bindings/sound/qcom,q6afe.h | 8 ++++++++ 1 file changed, 8 insertions(+)
diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index 1df06f8ad5c3..f3a435a112cb 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -107,6 +107,14 @@ #define QUINARY_TDM_RX_7 102 #define QUINARY_TDM_TX_7 103 #define DISPLAY_PORT_RX 104 +#define PRIMARY_PCM_RX 105 +#define PRIMARY_PCM_TX 106 +#define SECONDARY_PCM_RX 107 +#define SECONDARY_PCM_TX 108 +#define TERTIARY_PCM_RX 109 +#define TERTIARY_PCM_TX 110 +#define QUATERNARY_PCM_RX 111 +#define QUATERNARY_PCM_TX 112
#endif /* __DT_BINDINGS_Q6_AFE_H__ */
On 09/02/2020 15:47, Adam Serbinski wrote:
This patch adds bindings required for PCM ports on AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
include/dt-bindings/sound/qcom,q6afe.h | 8 ++++++++ 1 file changed, 8 insertions(+)
Reviewed-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index 1df06f8ad5c3..f3a435a112cb 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -107,6 +107,14 @@ #define QUINARY_TDM_RX_7 102 #define QUINARY_TDM_TX_7 103 #define DISPLAY_PORT_RX 104 +#define PRIMARY_PCM_RX 105 +#define PRIMARY_PCM_TX 106 +#define SECONDARY_PCM_RX 107 +#define SECONDARY_PCM_TX 108 +#define TERTIARY_PCM_RX 109 +#define TERTIARY_PCM_TX 110 +#define QUATERNARY_PCM_RX 111 +#define QUATERNARY_PCM_TX 112
#endif /* __DT_BINDINGS_Q6_AFE_H__ */
This patch adds support to pcm ports in AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6afe.c | 246 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 9 +- 2 files changed, 254 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..b53ad14a78fd 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -40,6 +40,7 @@
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 #define AFE_PARAM_ID_I2S_CONFIG 0x0001020D +#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297
@@ -117,6 +118,15 @@ #define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006 #define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A +#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B +#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C +#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D +#define AFE_PORT_ID_TERTIARY_PCM_RX 0x1012 +#define AFE_PORT_ID_TERTIARY_PCM_TX 0x1013 +#define AFE_PORT_ID_QUATERNARY_PCM_RX 0x1014 +#define AFE_PORT_ID_QUATERNARY_PCM_TX 0x1015 + /* Start of the range of port IDs for TDM devices. */ #define AFE_PORT_ID_TDM_PORT_RANGE_START 0x9000
@@ -421,6 +431,166 @@ struct afe_digital_clk_cfg { u16 reserved; } __packed;
+#define AFE_API_VERSION_PCM_CONFIG 0x1 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an external source. + */ + +#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 +/* Enumeration for the auxiliary PCM synchronization signal + * provided by an internal source. + */ +#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1 +/* Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use + * short synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_PCM 0x0 +/* + * Enumeration for the PCM configuration aux_mode parameter, + * which configures the auxiliary PCM interface to use long + * synchronization. + */ +#define AFE_PORT_PCM_AUX_MODE_AUX 0x1 +/* + * Enumeration for setting the PCM configuration frame to 8. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0 +/* + * Enumeration for setting the PCM configuration frame to 16. + */ +#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1 + +/* Enumeration for setting the PCM configuration frame to 32.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2 + +/* Enumeration for setting the PCM configuration frame to 64.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3 + +/* Enumeration for setting the PCM configuration frame to 128.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4 + +/* Enumeration for setting the PCM configuration frame to 256.*/ +#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5 + +/* Enumeration for setting the PCM configuration + * quantype parameter to A-law with no padding. + */ +#define AFE_PORT_PCM_ALAW_NOPADDING 0x0 + +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with no padding. + */ +#define AFE_PORT_PCM_MULAW_NOPADDING 0x1 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with no padding. + */ +#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2 +/* Enumeration for setting the PCM configuration quantype + * parameter to A-law with padding. + */ +#define AFE_PORT_PCM_ALAW_PADDING 0x3 +/* Enumeration for setting the PCM configuration quantype + * parameter to mu-law with padding. + */ +#define AFE_PORT_PCM_MULAW_PADDING 0x4 +/* Enumeration for setting the PCM configuration quantype + * parameter to linear with padding. + */ +#define AFE_PORT_PCM_LINEAR_PADDING 0x5 +/* Enumeration for disabling the PCM configuration + * ctrl_data_out_enable parameter. + * The PCM block is the only master. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0 +/* + * Enumeration for enabling the PCM configuration + * ctrl_data_out_enable parameter. The PCM block shares + * the signal with other masters. + */ +#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1 + +/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's + * (PCM configuration parameter). + */ + +struct afe_param_id_pcm_cfg { + u32 pcm_cfg_minor_version; +/* Minor version used for tracking the version of the AUX PCM + * configuration interface. + * Supported values: #AFE_API_VERSION_PCM_CONFIG + */ + + u16 aux_mode; +/* PCM synchronization setting. + * Supported values: + * - #AFE_PORT_PCM_AUX_MODE_PCM + * - #AFE_PORT_PCM_AUX_MODE_AUX + */ + + u16 sync_src; +/* Synchronization source. + * Supported values: + * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL + * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL + */ + + u16 frame_setting; +/* Number of bits per frame. + * Supported values: + * - #AFE_PORT_PCM_BITS_PER_FRAME_8 + * - #AFE_PORT_PCM_BITS_PER_FRAME_16 + * - #AFE_PORT_PCM_BITS_PER_FRAME_32 + * - #AFE_PORT_PCM_BITS_PER_FRAME_64 + * - #AFE_PORT_PCM_BITS_PER_FRAME_128 + * - #AFE_PORT_PCM_BITS_PER_FRAME_256 + */ + + u16 quantype; +/* PCM quantization type. + * Supported values: + * - #AFE_PORT_PCM_ALAW_NOPADDING + * - #AFE_PORT_PCM_MULAW_NOPADDING + * - #AFE_PORT_PCM_LINEAR_NOPADDING + * - #AFE_PORT_PCM_ALAW_PADDING + * - #AFE_PORT_PCM_MULAW_PADDING + * - #AFE_PORT_PCM_LINEAR_PADDING + */ + + u16 ctrl_data_out_enable; +/* Specifies whether the PCM block shares the data-out + * signal to the drive with other masters. + * Supported values: + * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE + * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE + */ + u16 reserved; + /* This field must be set to zero. */ + + u32 sample_rate; +/* Sampling rate of the port. + * Supported values: + * - #AFE_PORT_SAMPLE_RATE_8K + * - #AFE_PORT_SAMPLE_RATE_16K + */ + + u16 bit_width; +/* Bit width of the sample. + * Supported values: 16 + */ + + u16 num_channels; +/* Number of channels. + * Supported values: 1 to 4 + */ + + u16 slot_number_mapping[4]; +/* Specifies the slot number for the each channel in + * multi channel scenario. + * Supported values: 1 to 32 + */ +} __packed; + struct afe_param_id_i2s_cfg { u32 i2s_cfg_minor_version; u16 bit_width; @@ -452,6 +622,7 @@ 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_pcm_cfg pcm_cfg; struct afe_param_id_tdm_cfg tdm_cfg; } __packed;
@@ -707,6 +878,22 @@ 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}, + [PRIMARY_PCM_RX] = { AFE_PORT_ID_PRIMARY_PCM_RX, + PRIMARY_PCM_RX, 1, 1}, + [PRIMARY_PCM_TX] = { AFE_PORT_ID_PRIMARY_PCM_TX, + PRIMARY_PCM_RX, 0, 1}, + [SECONDARY_PCM_RX] = { AFE_PORT_ID_SECONDARY_PCM_RX, + SECONDARY_PCM_RX, 1, 1}, + [SECONDARY_PCM_TX] = { AFE_PORT_ID_SECONDARY_PCM_TX, + SECONDARY_PCM_TX, 0, 1}, + [TERTIARY_PCM_RX] = { AFE_PORT_ID_TERTIARY_PCM_RX, + TERTIARY_PCM_RX, 1, 1}, + [TERTIARY_PCM_TX] = { AFE_PORT_ID_TERTIARY_PCM_TX, + TERTIARY_PCM_TX, 0, 1}, + [QUATERNARY_PCM_RX] = { AFE_PORT_ID_QUATERNARY_PCM_RX, + QUATERNARY_PCM_RX, 1, 1}, + [QUATERNARY_PCM_TX] = { AFE_PORT_ID_QUATERNARY_PCM_TX, + QUATERNARY_PCM_TX, 0, 1}, };
static void q6afe_port_free(struct kref *ref) @@ -993,6 +1180,7 @@ int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, break; 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: + /* TDM cases overlap with PCM */ case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET; cset.clk_id = clk_id; @@ -1145,6 +1333,54 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+/** + * q6afe_pcm_port_prepare() - Prepare pcm afe port. + * + * @port: Instance of afe port + * @cfg: PCM configuration for the afe port + * + */ +int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->pcm_cfg.pcm_cfg_minor_version = AFE_API_VERSION_PCM_CONFIG; + pcfg->pcm_cfg.aux_mode = AFE_PORT_PCM_AUX_MODE_PCM; + + switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CPU is slave */ + pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL; + break; + default: + break; + } + + switch (cfg->sample_rate) { + case 8000: + pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128; + break; + case 16000: + pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64; + break; + } + pcfg->pcm_cfg.quantype = AFE_PORT_PCM_LINEAR_NOPADDING; + pcfg->pcm_cfg.ctrl_data_out_enable = AFE_PORT_PCM_CTRL_DATA_OE_DISABLE; + pcfg->pcm_cfg.reserved = 0; + pcfg->pcm_cfg.sample_rate = cfg->sample_rate; + + /* 16 bit mono */ + pcfg->pcm_cfg.bit_width = 16; + pcfg->pcm_cfg.num_channels = 1; + pcfg->pcm_cfg.slot_number_mapping[0] = 1; + + return 0; +} +EXPORT_SYMBOL_GPL(q6afe_pcm_port_prepare); + /** * q6afe_i2s_port_prepare() - Prepare i2s afe port. * @@ -1417,6 +1653,16 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_QUATERNARY_MI2S_TX: cfg_type = AFE_PARAM_ID_I2S_CONFIG; break; + case AFE_PORT_ID_PRIMARY_PCM_RX: + case AFE_PORT_ID_PRIMARY_PCM_TX: + case AFE_PORT_ID_SECONDARY_PCM_RX: + case AFE_PORT_ID_SECONDARY_PCM_TX: + case AFE_PORT_ID_TERTIARY_PCM_RX: + case AFE_PORT_ID_TERTIARY_PCM_TX: + case AFE_PORT_ID_QUATERNARY_PCM_RX: + case AFE_PORT_ID_QUATERNARY_PCM_TX: + cfg_type = AFE_PARAM_ID_PCM_CONFIG; + break; case AFE_PORT_ID_PRIMARY_TDM_RX ... AFE_PORT_ID_QUINARY_TDM_TX_7: cfg_type = AFE_PARAM_ID_TDM_CONFIG; break; diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index c7ed5422baff..c832be6d0ff5 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 113
#define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -170,6 +170,11 @@ struct q6afe_i2s_cfg { int fmt; };
+struct q6afe_pcm_cfg { + u32 sample_rate; + int fmt; +}; + struct q6afe_tdm_cfg { u16 num_channels; u32 sample_rate; @@ -188,6 +193,7 @@ struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; struct q6afe_slim_cfg slim; struct q6afe_i2s_cfg i2s_cfg; + struct q6afe_pcm_cfg pcm_cfg; struct q6afe_tdm_cfg tdm; };
@@ -203,6 +209,7 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, 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); +int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg); void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
On Sun, Feb 09, 2020 at 10:47:42AM -0500, Adam Serbinski wrote:
+#define AFE_API_VERSION_PCM_CONFIG 0x1 +/* Enumeration for the auxiliary PCM synchronization signal
- provided by an external source.
- */
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 +/* Enumeration for the auxiliary PCM synchronization signal
- provided by an internal source.
- */
This is a *weird* commenting style for these #defines and it's not consistent within the block, I'm seeing at least 3 different styles.
+/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
- (PCM configuration parameter).
- */
+struct afe_param_id_pcm_cfg {
Similar weird commenting here, please follow coding-style.rst.
- switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL;
break;
- case SND_SOC_DAIFMT_CBM_CFM:
/* CPU is slave */
pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL;
break;
- default:
break;
- }
Why is this not returning an error on unsupported values?
- switch (cfg->sample_rate) {
- case 8000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128;
break;
- case 16000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64;
break;
- }
Same here.
On 2020-02-10 08:31, Mark Brown wrote:
On Sun, Feb 09, 2020 at 10:47:42AM -0500, Adam Serbinski wrote:
+#define AFE_API_VERSION_PCM_CONFIG 0x1 +/* Enumeration for the auxiliary PCM synchronization signal
- provided by an external source.
- */
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0 +/* Enumeration for the auxiliary PCM synchronization signal
- provided by an internal source.
- */
This is a *weird* commenting style for these #defines and it's not consistent within the block, I'm seeing at least 3 different styles.
I will clean up the commenting.
- default:
break;
- }
Why is this not returning an error on unsupported values?
Only to be consistent with the pre-existing implementation for i2s ports. I will add an error return.
- switch (cfg->sample_rate) {
- case 8000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128;
break;
- case 16000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64;
break;
- }
Same here.
I will also add the error return here.
On 09/02/2020 15:47, Adam Serbinski wrote:
This patch adds support to pcm ports in AFE.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
sound/soc/qcom/qdsp6/q6afe.c | 246 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 9 +- 2 files changed, 254 insertions(+), 1 deletion(-)
Few general comments.
1>documentation to "struct afe_param_id_pcm_cfg " Either we follow kerneldoc style or not add this as we did with other similar afe port config structures. Am okay either way!
2> some of the defines in this patch has no reals users, so we better remove all the unused constants.
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..b53ad14a78fd 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -40,6 +40,7 @@
...
+/**
- q6afe_pcm_port_prepare() - Prepare pcm afe port.
- @port: Instance of afe port
- @cfg: PCM configuration for the afe port
- */
+int q6afe_pcm_port_prepare(struct q6afe_port *port, struct q6afe_pcm_cfg *cfg) +{
- union afe_port_config *pcfg = &port->port_cfg;
- pcfg->pcm_cfg.pcm_cfg_minor_version = AFE_API_VERSION_PCM_CONFIG;
- pcfg->pcm_cfg.aux_mode = AFE_PORT_PCM_AUX_MODE_PCM;
- switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_INTERNAL;
break;
- case SND_SOC_DAIFMT_CBM_CFM:
/* CPU is slave */
pcfg->pcm_cfg.sync_src = AFE_PORT_PCM_SYNC_SRC_EXTERNAL;
break;
- default:
break;
- }
- switch (cfg->sample_rate) {
- case 8000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_128;
break;
- case 16000:
pcfg->pcm_cfg.frame_setting = AFE_PORT_PCM_BITS_PER_FRAME_64;
break;
- }
- pcfg->pcm_cfg.quantype = AFE_PORT_PCM_LINEAR_NOPADDING;
- pcfg->pcm_cfg.ctrl_data_out_enable = AFE_PORT_PCM_CTRL_DATA_OE_DISABLE;
- pcfg->pcm_cfg.reserved = 0;
- pcfg->pcm_cfg.sample_rate = cfg->sample_rate;
- /* 16 bit mono */
- pcfg->pcm_cfg.bit_width = 16;
- pcfg->pcm_cfg.num_channels = 1;
- pcfg->pcm_cfg.slot_number_mapping[0] = 1;
PCM quantization type and Slot Mapping should come from device tree.
- return 0;
+} +EXPORT_SYMBOL_GPL(q6afe_pcm_port_prepare);
This patch adds support of AFE DAI for PCM port.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6afe-dai.c | 198 ++++++++++++++++++++++++++++++- 1 file changed, 197 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..23b29591ef47 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -151,6 +151,28 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; }
+static int q6pcm_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_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + + pcm->sample_rate = params_rate(params); + + return 0; +} + +static int q6pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + + pcm->fmt = fmt; + + return 0; +} + static int q6i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -358,6 +380,15 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, return rc; } break; + case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX: + rc = q6afe_pcm_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].pcm_cfg); + if (rc < 0) { + dev_err(dai->dev, "fail to prepare AFE port %x\n", + dai->id); + return rc; + } + break; case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7: q6afe_tdm_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].tdm); @@ -429,11 +460,32 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR: + case Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR: case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); + } + + return 0; +} + +static int q6afe_tdm_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_port *port = dai_data->port[dai->id]; + + switch (clk_id) { + case LPAIF_DIG_CLK: + return q6afe_port_set_sysclk(port, clk_id, 0, 5, freq, dir); + case LPAIF_BIT_CLK: + case LPAIF_OSR_CLK: + return q6afe_port_set_sysclk(port, clk_id, + Q6AFE_LPASS_CLK_SRC_INTERNAL, + Q6AFE_LPASS_CLK_ROOT_DEFAULT, + freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO, @@ -468,6 +520,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
+ {"Primary PCM Playback", NULL, "PRI_PCM_RX"}, + {"Secondary PCM Playback", NULL, "SEC_PCM_RX"}, + {"Tertiary PCM Playback", NULL, "TERT_PCM_RX"}, + {"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"}, + {"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"}, {"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"}, {"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"}, @@ -562,6 +619,11 @@ 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"}, + + {"PRI_PCM_TX", NULL, "Primary PCM Capture"}, + {"SEC_PCM_TX", NULL, "Secondary PCM Capture"}, + {"TERT_PCM_TX", NULL, "Tertiary PCM Capture"}, + {"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"}, };
static const struct snd_soc_dai_ops q6hdmi_ops = { @@ -578,6 +640,14 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .set_sysclk = q6afe_mi2s_set_sysclk, };
+static const struct snd_soc_dai_ops q6pcm_ops = { + .prepare = q6afe_dai_prepare, + .hw_params = q6pcm_hw_params, + .set_fmt = q6pcm_set_fmt, + .shutdown = q6afe_dai_shutdown, + .set_sysclk = q6afe_mi2s_set_sysclk, +}; + static const struct snd_soc_dai_ops q6slim_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6slim_hw_params, @@ -588,7 +658,7 @@ static const struct snd_soc_dai_ops q6slim_ops = { static const struct snd_soc_dai_ops q6tdm_ops = { .prepare = q6afe_dai_prepare, .shutdown = q6afe_dai_shutdown, - .set_sysclk = q6afe_mi2s_set_sysclk, + .set_sysclk = q6afe_tdm_set_sysclk, .set_tdm_slot = q6tdm_set_tdm_slot, .set_channel_map = q6tdm_set_channel_map, .hw_params = q6tdm_hw_params, @@ -1012,6 +1082,115 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .ops = &q6i2s_ops, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Primary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = PRIMARY_PCM_RX, + .name = "PRI_PCM_RX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Primary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = PRIMARY_PCM_TX, + .name = "PRI_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Secondary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "SEC_PCM_RX", + .id = SECONDARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Secondary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = SECONDARY_PCM_TX, + .name = "SEC_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Tertiary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "TERT_PCM_RX", + .id = TERTIARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Tertiary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = TERTIARY_PCM_TX, + .name = "TERT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Quaternary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .name = "QUAT_PCM_RX", + .id = QUATERNARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Quaternary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 8000, + .rate_max = 16000, + }, + .id = QUATERNARY_PCM_TX, + .name = "QUAT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, }, Q6AFE_TDM_PB_DAI("Primary", 0, PRIMARY_TDM_RX_0), Q6AFE_TDM_PB_DAI("Primary", 1, PRIMARY_TDM_RX_1), @@ -1169,6 +1348,23 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL, + 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
On Sun, Feb 09, 2020 at 10:47:43AM -0500, Adam Serbinski wrote:
+static int q6pcm_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_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg;
- pcm->sample_rate = params_rate(params);
This and set_fmt() don't do any validation of the value being set.
static const struct snd_soc_dai_ops q6tdm_ops = { .prepare = q6afe_dai_prepare, .shutdown = q6afe_dai_shutdown,
- .set_sysclk = q6afe_mi2s_set_sysclk,
- .set_sysclk = q6afe_tdm_set_sysclk, .set_tdm_slot = q6tdm_set_tdm_slot, .set_channel_map = q6tdm_set_channel_map, .hw_params = q6tdm_hw_params,
This looks like a separate bug fix that should be split out?
- }, {
.playback = {
.stream_name = "Primary PCM Playback",
.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 8000,
.rate_max = 16000,
},
It is surprising to see rate_min and rate_max specified when we're not using _KNOT, and again there's weird formatting here with the tabs before the rate values.
Few minor comments
On 09/02/2020 15:47, Adam Serbinski wrote:
This patch adds support of AFE DAI for PCM port.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
sound/soc/qcom/qdsp6/q6afe-dai.c | 198 ++++++++++++++++++++++++++++++- 1 file changed, 197 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..23b29591ef47 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
...
+static int q6afe_tdm_set_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
+{
Why are we adding exactly duplicate function of q6afe_mi2s_set_sysclk here?
- struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
- struct q6afe_port *port = dai_data->port[dai->id];
- switch (clk_id) {
- case LPAIF_DIG_CLK:
return q6afe_port_set_sysclk(port, clk_id, 0, 5, freq, dir);
- case LPAIF_BIT_CLK:
- case LPAIF_OSR_CLK:
return q6afe_port_set_sysclk(port, clk_id,
Q6AFE_LPASS_CLK_SRC_INTERNAL,
Q6AFE_LPASS_CLK_ROOT_DEFAULT,
case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO,freq, dir);
@@ -468,6 +520,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
- {"Primary PCM Playback", NULL, "PRI_PCM_RX"},
- {"Secondary PCM Playback", NULL, "SEC_PCM_RX"},
- {"Tertiary PCM Playback", NULL, "TERT_PCM_RX"},
- {"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"},
- {"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"}, {"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"}, {"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"},
@@ -562,6 +619,11 @@ 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"},
- {"PRI_PCM_TX", NULL, "Primary PCM Capture"},
- {"SEC_PCM_TX", NULL, "Secondary PCM Capture"},
- {"TERT_PCM_TX", NULL, "Tertiary PCM Capture"},
- {"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"}, };
...
- SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL,
0, 0, 0, 0),
This can be in single line, same for below
- SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL,
0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL,
0, 0, 0, 0),
On 2020-02-10 12:13, Srinivas Kandagatla wrote:
Few minor comments
+static int q6afe_tdm_set_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
+{
Why are we adding exactly duplicate function of q6afe_mi2s_set_sysclk here?
It isn't an exact duplicate.
The reason I split off the new function is because the clock IDs for PCM overlap/duplicate the clock IDs for TDM, yet the parameters to q6afe_port_set_sysclk are not the same for PCM and TDM.
- SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL,
0, 0, 0, 0),
This can be in single line, same for below
I will adjust these.
On 10/02/2020 17:22, Adam Serbinski wrote:
Why are we adding exactly duplicate function of q6afe_mi2s_set_sysclk here?
It isn't an exact duplicate.
The reason I split off the new function is because the clock IDs for PCM overlap/duplicate the clock IDs for TDM, yet the parameters to q6afe_port_set_sysclk are not the same for PCM and TDM.
we should be able to use dai->id to make that decision.
--srini
+ SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, + 0, 0, 0, 0),
This patch adds support to PCM_PORT mixers required to select path between ASM stream and AFE ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a..3a81d2161707 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -67,6 +67,10 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "PRI_PCM_TX", "PRI_PCM_TX" }, \ + { mix_name, "SEC_PCM_TX", "SEC_PCM_TX" }, \ + { mix_name, "TERT_PCM_TX", "TERT_PCM_TX" }, \ + { mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" }, \ { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ @@ -128,6 +132,18 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ @@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) };
+static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) }; + +static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) }; + +static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) }; + +static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) }; + static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) };
@@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0, tertiary_mi2s_rx_mixer_controls, ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + primary_pcm_rx_mixer_controls, + ARRAY_SIZE(primary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + secondary_pcm_rx_mixer_controls, + ARRAY_SIZE(secondary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + tertiary_pcm_rx_mixer_controls, + ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + quaternary_pcm_rx_mixer_controls, + ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)), SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, pri_tdm_rx_0_mixer_controls, ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)), @@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"), + Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer", "PRIMARY_TDM_RX_0"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer",
On 09/02/2020 15:47, Adam Serbinski wrote:
This patch adds support to PCM_PORT mixers required to select path between ASM stream and AFE ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++
Reviewed-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
1 file changed, 44 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a..3a81d2161707 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -67,6 +67,10 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \
- { mix_name, "PRI_PCM_TX", "PRI_PCM_TX" }, \
- { mix_name, "SEC_PCM_TX", "SEC_PCM_TX" }, \
- { mix_name, "TERT_PCM_TX", "TERT_PCM_TX" }, \
- { mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" }, \ { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \
@@ -128,6 +132,18 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \
- SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX, \
id, 1, 0, msm_routing_get_audio_mixer, \
msm_routing_put_audio_mixer), \
- SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX, \
id, 1, 0, msm_routing_get_audio_mixer, \
msm_routing_put_audio_mixer), \
- SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX, \
id, 1, 0, msm_routing_get_audio_mixer, \
msm_routing_put_audio_mixer), \
- SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX, \
id, 1, 0, msm_routing_get_audio_mixer, \
SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \msm_routing_put_audio_mixer), \
@@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) };
+static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = {
- Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) };
+static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = {
- Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) };
+static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = {
- Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) };
+static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = {
- Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) };
- static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) };
@@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0, tertiary_mi2s_rx_mixer_controls, ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
- SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
primary_pcm_rx_mixer_controls,
ARRAY_SIZE(primary_pcm_rx_mixer_controls)),
- SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
secondary_pcm_rx_mixer_controls,
ARRAY_SIZE(secondary_pcm_rx_mixer_controls)),
- SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
tertiary_pcm_rx_mixer_controls,
ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)),
- SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
quaternary_pcm_rx_mixer_controls,
SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, pri_tdm_rx_0_mixer_controls, ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)),ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)),
@@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"),
- Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"),
- Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"),
- Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"),
- Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer", "PRIMARY_TDM_RX_0"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer",
This adds support to primary and quarternary I2S and PCM ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/apq8096.c | 86 +++++++++++++++++++++++++++++++++------- 1 file changed, 71 insertions(+), 15 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 94363fd6846a..1edcaa15234f 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -8,24 +8,13 @@ #include <sound/soc-dapm.h> #include <sound/pcm.h> #include "common.h" +#include "qdsp6/q6afe.h"
#define SLIM_MAX_TX_PORTS 16 #define SLIM_MAX_RX_PORTS 16 #define WCD9335_DEFAULT_MCLK_RATE 9600000 - -static int apq8096_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; -} +#define MI2S_BCLK_RATE 1536000 +#define PCM_BCLK_RATE 1024000
static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -33,10 +22,32 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + 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); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0;
+ switch (cpu_dai->id) { + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + rate->min = 16000; + rate->max = 16000; + channels->min = 1; + channels->max = 1; + break; + default: + rate->min = 48000; + rate->max = 48000; + channels->min = 1; + channels->max = 2; + break; + } + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -60,8 +71,54 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; }
+static int msm_snd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + 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, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_MI2S_RX: + case QUATERNARY_MI2S_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + default: + return -1; + } + return 0; +} + static struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, + .startup = msm_snd_startup, };
static int apq8096_init(struct snd_soc_pcm_runtime *rtd) @@ -96,7 +153,6 @@ static void apq8096_add_be_ops(struct snd_soc_card *card)
for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) { - link->be_hw_params_fixup = apq8096_be_hw_params_fixup; link->init = apq8096_init; link->ops = &apq8096_ops; }
When not specifying a codec, use snd-soc-dummy-dai. This supports the case where a fixed configuration codec is attached, such as bluetooth hfp.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/common.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 6c20bdd850f3..aa2f2238aca0 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -84,7 +84,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; }
- if (codec && platform) { + if (platform) { link->platforms->of_node = of_parse_phandle(platform, "sound-dai", 0); @@ -94,10 +94,22 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; }
- ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "%s: codec dai not found\n", link->name); - goto err; + if (codec) { + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "%s: codec dai not found\n", link->name); + goto err; + } + } else { + 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->no_pcm = 1; link->ignore_pmdown_time = 1;
On Sun, Feb 09, 2020 at 10:47:46AM -0500, Adam Serbinski wrote:
When not specifying a codec, use snd-soc-dummy-dai. This supports the case where a fixed configuration codec is attached, such as bluetooth hfp.
Fixed configuration devices should still have normal drivers that say what those fixed configurations are.
This patch adds support to primary pcm and quaternary i2s ports.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 113 +++++++++++++ arch/arm64/boot/dts/qcom/msm8996-pins.dtsi | 162 +++++++++++++++++++ 2 files changed, 275 insertions(+)
diff --git a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi index dba3488492f1..4149ac4147a0 100644 --- a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi +++ b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi @@ -683,8 +683,31 @@ }; };
+/* PRI I2S on QCA6174 and QUAT I2S on LS each uses 2 I2S SD Lines for audio */ +&q6afedai { + pi2s@16 { + reg = <16>; + qcom,sd-lines = <1>; + }; + pi2s@17 { + reg = <17>; + qcom,sd-lines = <0>; + }; + qi2s@22 { + reg = <22>; + qcom,sd-lines = <0>; + }; + qi2s@23 { + reg = <23>; + qcom,sd-lines = <1>; + }; +}; + &sound { compatible = "qcom,apq8096-sndcard"; + pinctrl-0 = <&quat_mi2s_active &quat_mi2s_sd0_active &quat_mi2s_sd1_active &pri_mi2s_active &pri_mi2s_sd0_active &pri_mi2s_sd1_active>; + pinctrl-names = "default"; + model = "DB820c"; audio-routing = "RX_BIAS", "MCLK";
@@ -709,6 +732,41 @@ }; };
+ mm4-dai-link { + link-name = "MultiMedia4"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA4>; + }; + }; + + mm5-dai-link { + link-name = "MultiMedia5"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA5>; + }; + }; + + mm6-dai-link { + link-name = "MultiMedia6"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA6>; + }; + }; + + mm7-dai-link { + link-name = "MultiMedia7"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA7>; + }; + }; + + mm8-dai-link { + link-name = "MultiMedia8"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA8>; + }; + }; + hdmi-dai-link { link-name = "HDMI"; cpu { @@ -753,4 +811,59 @@ sound-dai = <&wcd9335 1>; }; }; + + scoplay-dai-link { + link-name = "SCO-PCM-Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_PCM_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; + + scocap-dai-link { + link-name = "SCO-PCM-Capture"; + cpu { + sound-dai = <&q6afedai PRIMARY_PCM_TX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; + + mi2splay-dai-link { + link-name = "QUAT-MI2S-Playback"; + cpu { + sound-dai = <&q6afedai QUATERNARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + +// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm5142_4c>, <&pcm5142_4d>; +// }; + + }; + + mi2scap-dai-link { + link-name = "QUAT-MI2S-Capture"; + cpu { + sound-dai = <&q6afedai QUATERNARY_MI2S_TX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + +// EXAMPLE: For adding real codecs +// codec { +// sound-dai = <&pcm1865>; +// }; + }; }; diff --git a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi index ac1ede579361..e8221c4d05f7 100644 --- a/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi +++ b/arch/arm64/boot/dts/qcom/msm8996-pins.dtsi @@ -288,6 +288,168 @@ }; };
+ pri_mi2s_active: pri_mi2s_active { + mux { + pins = "gpio65", "gpio66"; + function = "pri_mi2s"; + }; + config { + pins = "gpio65", "gpio66"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + output-high; + }; + }; + + pri_mi2s_sleep: pri_mi2s_sleep { + mux { + pins = "gpio65", "gpio66"; + function = "gpio"; + }; + + config { + pins = "gpio65", "gpio66"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd0_sleep: pri_mi2s_sd0_sleep { + mux { + pins = "gpio67"; + function = "gpio"; + }; + + config { + pins = "gpio67"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd0_active: pri_mi2s_sd0_active { + mux { + pins = "gpio67"; + function = "pri_mi2s"; + }; + + config { + pins = "gpio67"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + pri_mi2s_sd1_sleep: pri_mi2s_sd1_sleep { + mux { + pins = "gpio68"; + function = "gpio"; + }; + + config { + pins = "gpio68"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + pri_mi2s_sd1_active: pri_mi2s_sd1_active { + mux { + pins = "gpio68"; + function = "pri_mi2s"; + }; + + config { + pins = "gpio68"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + quat_mi2s_active: quat_mi2s_active { + mux { + pins = "gpio58", "gpio59"; + function = "qua_mi2s"; + }; + config { + pins = "gpio58", "gpio59"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + output-high; + }; + }; + + quat_mi2s_sleep: quat_mi2s_sleep { + mux { + pins = "gpio58", "gpio59"; + function = "gpio"; + }; + + config { + pins = "gpio58", "gpio59"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd0_sleep: quat_mi2s_sd0_sleep { + mux { + pins = "gpio60"; + function = "gpio"; + }; + + config { + pins = "gpio60"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd0_active: quat_mi2s_sd0_active { + mux { + pins = "gpio60"; + function = "qua_mi2s"; + }; + + config { + pins = "gpio60"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + + quat_mi2s_sd1_sleep: quat_mi2s_sd1_sleep { + mux { + pins = "gpio61"; + function = "gpio"; + }; + + config { + pins = "gpio61"; + drive-strength = <2>; /* 2 mA */ + bias-pull-down; /* PULL DOWN */ + input-enable; + }; + }; + + quat_mi2s_sd1_active: quat_mi2s_sd1_active { + mux { + pins = "gpio61"; + function = "qua_mi2s"; + }; + + config { + pins = "gpio61"; + drive-strength = <8>; /* 8 mA */ + bias-disable; /* NO PULL */ + }; + }; + sdc2_clk_on: sdc2_clk_on { config { pins = "sdc2_clk";
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org --- sound/soc/qcom/apq8096.c | 92 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 90 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1edcaa15234f..882f2c456321 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -16,6 +16,9 @@ #define MI2S_BCLK_RATE 1536000 #define PCM_BCLK_RATE 1024000
+static int pri_pcm_sample_rate = 16000; +static int quat_pcm_sample_rate = 16000; + static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -33,10 +36,15 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, switch (cpu_dai->id) { case PRIMARY_PCM_RX: case PRIMARY_PCM_TX: + rate->min = pri_pcm_sample_rate; + rate->max = pri_pcm_sample_rate; + channels->min = 1; + channels->max = 1; + break; case QUATERNARY_PCM_RX: case QUATERNARY_PCM_TX: - rate->min = 16000; - rate->max = 16000; + rate->min = quat_pcm_sample_rate; + rate->max = quat_pcm_sample_rate; channels->min = 1; channels->max = 1; break; @@ -121,6 +129,83 @@ static struct snd_soc_ops apq8096_ops = { .startup = msm_snd_startup, };
+static char const *pcm_sample_rate_text[] = {"8 kHz", "16 kHz"}; +static const struct soc_enum pcm_snd_enum = + SOC_ENUM_SINGLE_EXT(2, pcm_sample_rate_text); + +static int get_sample_rate_idx(int sample_rate) +{ + int sample_rate_idx = 0; + + switch (sample_rate) { + case 8000: + sample_rate_idx = 0; + break; + case 16000: + default: + sample_rate_idx = 1; + break; + } + + return sample_rate_idx; +} + +static int pri_pcm_sample_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + get_sample_rate_idx(pri_pcm_sample_rate); + return 0; +} + +static int quat_pcm_sample_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + get_sample_rate_idx(quat_pcm_sample_rate); + return 0; +} + +static int get_sample_rate(int idx) +{ + int sample_rate_val = 0; + + switch (idx) { + case 0: + sample_rate_val = 8000; + break; + case 1: + default: + sample_rate_val = 16000; + break; + } + + return sample_rate_val; +} + +static int pri_pcm_sample_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + pri_pcm_sample_rate = + get_sample_rate(ucontrol->value.integer.value[0]); + return 0; +} + +static int quat_pcm_sample_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + quat_pcm_sample_rate = + get_sample_rate(ucontrol->value.integer.value[0]); + return 0; +} + +static const struct snd_kcontrol_new card_controls[] = { + SOC_ENUM_EXT("PRI_PCM SampleRate", pcm_snd_enum, + pri_pcm_sample_rate_get, pri_pcm_sample_rate_put), + SOC_ENUM_EXT("QUAT_PCM SampleRate", pcm_snd_enum, + quat_pcm_sample_rate_get, quat_pcm_sample_rate_put), +}; + static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; @@ -182,6 +267,9 @@ static int apq8096_platform_probe(struct platform_device *pdev) if (ret) goto err_card_register;
+ snd_soc_add_card_controls(card, card_controls, + ARRAY_SIZE(card_controls)); + return 0;
err_card_register:
On Sun, Feb 09, 2020 at 10:47:48AM -0500, Adam Serbinski wrote:
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
This would seem like an excellent thing to put in the driver for the baseband or bluetooth.
On 2020-02-10 08:36, Mark Brown wrote:
On Sun, Feb 09, 2020 at 10:47:48AM -0500, Adam Serbinski wrote:
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
This would seem like an excellent thing to put in the driver for the baseband or bluetooth.
The value that must be set to this control is not available to the bluetooth driver. It originates from the bluetooth stack in userspace, typically either blueZ or fluoride, as a result of a negotiation between the two devices participating in the HFP call.
On Mon, Feb 10, 2020 at 10:45:16AM -0500, Adam Serbinski wrote:
On 2020-02-10 08:36, Mark Brown wrote:
This would seem like an excellent thing to put in the driver for the baseband or bluetooth.
The value that must be set to this control is not available to the bluetooth driver. It originates from the bluetooth stack in userspace, typically either blueZ or fluoride, as a result of a negotiation between the two devices participating in the HFP call.
To repeat my comment on another patch in the series there should still be some representation of the DAI for this device in the kernel.
On 2020-02-10 13:26, Mark Brown wrote:
On Mon, Feb 10, 2020 at 10:45:16AM -0500, Adam Serbinski wrote:
On 2020-02-10 08:36, Mark Brown wrote:
This would seem like an excellent thing to put in the driver for the baseband or bluetooth.
The value that must be set to this control is not available to the bluetooth driver. It originates from the bluetooth stack in userspace, typically either blueZ or fluoride, as a result of a negotiation between the two devices participating in the HFP call.
To repeat my comment on another patch in the series there should still be some representation of the DAI for this device in the kernel.
Respectfully, I'm not sure I understand what it is that you are suggesting.
Is it your intention to suggest that instead of adding controls to the machine driver, I should instead write a codec driver to contain those controls?
Or is it your intention to suggest that something within the kernel is already aware of the rate to be set, and it is that which should set the rate rather than a control?
On Mon, Feb 10, 2020 at 03:00:55PM -0500, Adam Serbinski wrote:
On 2020-02-10 13:26, Mark Brown wrote:
To repeat my comment on another patch in the series there should still be some representation of the DAI for this device in the kernel.
Respectfully, I'm not sure I understand what it is that you are suggesting.
Is it your intention to suggest that instead of adding controls to the machine driver, I should instead write a codec driver to contain those controls?
I have already separately said that you should write a CODEC driver for this CODEC. I'm saying that this seems like the sort of thing that might fit in that CODEC driver.
Or is it your intention to suggest that something within the kernel is already aware of the rate to be set, and it is that which should set the rate rather than a control?
That would be one example of how such a CODEC driver could be configured, and is how other baseband/BT devices have ended up going (see cx20442.c for example).
On 2020-02-10 15:08, Mark Brown wrote:
On Mon, Feb 10, 2020 at 03:00:55PM -0500, Adam Serbinski wrote:
On 2020-02-10 13:26, Mark Brown wrote:
To repeat my comment on another patch in the series there should still be some representation of the DAI for this device in the kernel.
Respectfully, I'm not sure I understand what it is that you are suggesting.
Is it your intention to suggest that instead of adding controls to the machine driver, I should instead write a codec driver to contain those controls?
I have already separately said that you should write a CODEC driver for this CODEC. I'm saying that this seems like the sort of thing that might fit in that CODEC driver.
I see. My initial thought with respect to the codec driver would be just to use bt-sco.c, which is a dummy codec. I can certainly implement a new codec driver.
Or is it your intention to suggest that something within the kernel is already aware of the rate to be set, and it is that which should set the rate rather than a control?
That would be one example of how such a CODEC driver could be configured, and is how other baseband/BT devices have ended up going (see cx20442.c for example).
I am not aware of how this could be done for bluetooth, since the value still has to originate from userspace. The driver you referred to supports only a single sample rate, whereas for bluetooth, 2 sample rates are required, and nothing in the kernel is aware of the appropriate rate, at least in the case of the qca6174a I'm working with right now, or for that matter, TI Wilink 8, which I've also worked with.
My concern with implementing this in a new codec driver, is that this codec driver will be bound to qdsp6, since its purpose is to work around a characteristic of this DSP. Under simple-card, for instance, it would be redundant, since in that case, the parameters userspace uses to open the pcm will be propagated to the port. But under qdsp6, userspace could open the pcm at 44.1 kHz, yet the backend port is still set to 8 or 16 kHz, and the DSP resamples between them, so the sole purpose of this change is to allow userspace to deliver the required sample rate to the back end of qdsp6.
On Mon, Feb 10, 2020 at 04:13:52PM -0500, Adam Serbinski wrote:
I am not aware of how this could be done for bluetooth, since the value still has to originate from userspace. The driver you referred to supports only a single sample rate, whereas for bluetooth, 2 sample rates are required, and nothing in the kernel is aware of the appropriate rate, at least in the case of the qca6174a I'm working with right now, or for that matter, TI Wilink 8, which I've also worked with.
There's generic support in the CODEC<->CODEC link code for setting the DAI configuration from userspace.
On 2020-02-11 06:42, Mark Brown wrote:
On Mon, Feb 10, 2020 at 04:13:52PM -0500, Adam Serbinski wrote:
I am not aware of how this could be done for bluetooth, since the value still has to originate from userspace. The driver you referred to supports only a single sample rate, whereas for bluetooth, 2 sample rates are required, and nothing in the kernel is aware of the appropriate rate, at least in the case of the qca6174a I'm working with right now, or for that matter, TI Wilink 8, which I've also worked with.
There's generic support in the CODEC<->CODEC link code for setting the DAI configuration from userspace.
Ok. Its going to take some time to get my head around that, so for the time being I'm going to drop this feature and get the rest fixed for inclusion.
Thanks.
Dne 09. 02. 20 v 16:47 Adam Serbinski napsal(a):
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
Two points:
Why enum? It adds just more code than the integer value handlers.
Also, this belongs to the PCM interface, so it should be handled with SNDRV_CTL_ELEM_IFACE_PCM not mixer.
The name should be probably "Rate" and assigned to the corresponding PCM device.
Add this to Documentation/sound/designs/control-names.rst .
Jaroslav
Signed-off-by: Adam Serbinski adam@serbinski.com CC: Andy Gross agross@kernel.org CC: Mark Rutland mark.rutland@arm.com CC: Liam Girdwood lgirdwood@gmail.com CC: Patrick Lai plai@codeaurora.org CC: Banajit Goswami bgoswami@codeaurora.org CC: Jaroslav Kysela perex@perex.cz CC: Takashi Iwai tiwai@suse.com CC: alsa-devel@alsa-project.org CC: linux-arm-msm@vger.kernel.org CC: devicetree@vger.kernel.org CC: linux-kernel@vger.kernel.org
sound/soc/qcom/apq8096.c | 92 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 90 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1edcaa15234f..882f2c456321 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -16,6 +16,9 @@ #define MI2S_BCLK_RATE 1536000 #define PCM_BCLK_RATE 1024000
+static int pri_pcm_sample_rate = 16000; +static int quat_pcm_sample_rate = 16000;
- static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) {
@@ -33,10 +36,15 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, switch (cpu_dai->id) { case PRIMARY_PCM_RX: case PRIMARY_PCM_TX:
rate->min = pri_pcm_sample_rate;
rate->max = pri_pcm_sample_rate;
channels->min = 1;
channels->max = 1;
case QUATERNARY_PCM_RX: case QUATERNARY_PCM_TX:break;
rate->min = 16000;
rate->max = 16000;
rate->min = quat_pcm_sample_rate;
channels->min = 1; channels->max = 1; break;rate->max = quat_pcm_sample_rate;
@@ -121,6 +129,83 @@ static struct snd_soc_ops apq8096_ops = { .startup = msm_snd_startup, };
+static char const *pcm_sample_rate_text[] = {"8 kHz", "16 kHz"}; +static const struct soc_enum pcm_snd_enum =
SOC_ENUM_SINGLE_EXT(2, pcm_sample_rate_text);
+static int get_sample_rate_idx(int sample_rate) +{
- int sample_rate_idx = 0;
- switch (sample_rate) {
- case 8000:
sample_rate_idx = 0;
break;
- case 16000:
- default:
sample_rate_idx = 1;
break;
- }
- return sample_rate_idx;
+}
+static int pri_pcm_sample_rate_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
+{
- ucontrol->value.integer.value[0] =
get_sample_rate_idx(pri_pcm_sample_rate);
- return 0;
+}
+static int quat_pcm_sample_rate_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
+{
- ucontrol->value.integer.value[0] =
get_sample_rate_idx(quat_pcm_sample_rate);
- return 0;
+}
+static int get_sample_rate(int idx) +{
- int sample_rate_val = 0;
- switch (idx) {
- case 0:
sample_rate_val = 8000;
break;
- case 1:
- default:
sample_rate_val = 16000;
break;
- }
- return sample_rate_val;
+}
+static int pri_pcm_sample_rate_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
+{
- pri_pcm_sample_rate =
get_sample_rate(ucontrol->value.integer.value[0]);
- return 0;
+}
+static int quat_pcm_sample_rate_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
+{
- quat_pcm_sample_rate =
get_sample_rate(ucontrol->value.integer.value[0]);
- return 0;
+}
+static const struct snd_kcontrol_new card_controls[] = {
- SOC_ENUM_EXT("PRI_PCM SampleRate", pcm_snd_enum,
pri_pcm_sample_rate_get, pri_pcm_sample_rate_put),
- SOC_ENUM_EXT("QUAT_PCM SampleRate", pcm_snd_enum,
quat_pcm_sample_rate_get, quat_pcm_sample_rate_put),
+};
- static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai;
@@ -182,6 +267,9 @@ static int apq8096_platform_probe(struct platform_device *pdev) if (ret) goto err_card_register;
snd_soc_add_card_controls(card, card_controls,
ARRAY_SIZE(card_controls));
return 0;
err_card_register:
On 2020-02-10 11:18, Jaroslav Kysela wrote:
Dne 09. 02. 20 v 16:47 Adam Serbinski napsal(a):
This makes it possible for the backend sample rate to be set to 8000 or 16000 Hz, depending on the needs of the HFP call being set up.
Two points:
Why enum? It adds just more code than the integer value handlers.
Because enum allows the potential values to be restricted to a set of distinct values rather than a range. And while yes, I understand that the value can be validated, or the step can in this case be set to correspond to the difference between the current 2 values, this approach would neither make it clear to the user what the permitted values are, nor would it scale well once additional values are required.
Also, this belongs to the PCM interface, so it should be handled with SNDRV_CTL_ELEM_IFACE_PCM not mixer.
The name should be probably "Rate" and assigned to the corresponding PCM device.
Add this to Documentation/sound/designs/control-names.rst .
Above 3 lines are noted, I will make these changed.
On Sun, Feb 09, 2020 at 10:47:40AM -0500, Adam Serbinski wrote:
Changes from V1:
Rename patch: from: dts: msm8996/db820c: enable primary pcm and quaternary i2s
Please don't send new serieses in reply to old ones, it can make it confusing what's going on and what the current version is.
On 2020-02-10 07:17, Mark Brown wrote:
On Sun, Feb 09, 2020 at 10:47:40AM -0500, Adam Serbinski wrote:
Changes from V1:
Rename patch: from: dts: msm8996/db820c: enable primary pcm and quaternary i2s
Please don't send new serieses in reply to old ones, it can make it confusing what's going on and what the current version is.
My apologies. Its my first time doing this. Thank you for the advice.
-Adam
participants (5)
-
Adam Serbinski
-
Bjorn Andersson
-
Jaroslav Kysela
-
Mark Brown
-
Srinivas Kandagatla