From: Wesley Cheng <quic_wcheng@quicinc.com>
To: <srinivas.kandagatla@linaro.org>, <mathias.nyman@intel.com>,
<perex@perex.cz>, <conor+dt@kernel.org>, <corbet@lwn.net>,
<lgirdwood@gmail.com>, <andersson@kernel.org>,
<krzysztof.kozlowski+dt@linaro.org>, <gregkh@linuxfoundation.org>,
<Thinh.Nguyen@synopsys.com>, <broonie@kernel.org>,
<bgoswami@quicinc.com>, <tiwai@suse.com>, <robh@kernel.org>,
<konrad.dybcio@linaro.org>
Cc: <linux-kernel@vger.kernel.org>, <devicetree@vger.kernel.org>,
<linux-sound@vger.kernel.org>, <linux-usb@vger.kernel.org>,
<linux-arm-msm@vger.kernel.org>, <linux-doc@vger.kernel.org>,
<alsa-devel@alsa-project.org>,
Wesley Cheng <quic_wcheng@quicinc.com>
Subject: [PATCH v20 11/41] ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp
Date: Thu, 25 Apr 2024 14:50:55 -0700 [thread overview]
Message-ID: <20240425215125.29761-12-quic_wcheng@quicinc.com> (raw)
In-Reply-To: <20240425215125.29761-1-quic_wcheng@quicinc.com>
The QC ADSP is able to support USB playback endpoints, so that the main
application processor can be placed into lower CPU power modes. This adds
the required AFE port configurations and port start command to start an
audio session.
Specifically, the QC ADSP can support all potential endpoints that are
exposed by the audio data interface. This includes, feedback endpoints
(both implicit and explicit) as well as the isochronous (data) endpoints.
The size of audio samples sent per USB frame (microframe) will be adjusted
based on information received on the feedback endpoint.
Some pre-requisites are needed before issuing the AFE port start command,
such as setting the USB AFE dev_token. This carries information about the
available USB SND cards and PCM devices that have been discovered on the
USB bus. The dev_token field is used by the audio DSP to notify the USB
offload driver of which card and PCM index to enable playback on.
Signed-off-by: Wesley Cheng <quic_wcheng@quicinc.com>
---
sound/soc/qcom/qdsp6/q6afe-dai.c | 60 +++++++
sound/soc/qcom/qdsp6/q6afe.c | 191 ++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6afe.h | 36 ++++-
sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++
sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 +
sound/soc/qcom/qdsp6/q6routing.c | 9 ++
6 files changed, 318 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index a9c4f896a7df..80b22d1a24d0 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -92,6 +92,39 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int q6afe_usb_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);
+ int channels = params_channels(params);
+ int rate = params_rate(params);
+ struct q6afe_usb_cfg *usb = &dai_data->port_config[dai->id].usb_audio;
+
+ usb->sample_rate = rate;
+ usb->num_channels = channels;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_U16_LE:
+ case SNDRV_PCM_FORMAT_S16_LE:
+ usb->bit_width = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ usb->bit_width = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ usb->bit_width = 32;
+ break;
+ default:
+ dev_err(dai->dev, "%s: invalid format %d\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
static int q6i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -392,6 +425,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
q6afe_cdc_dma_port_prepare(dai_data->port[dai->id],
&dai_data->port_config[dai->id].dma_cfg);
break;
+ case USB_RX:
+ q6afe_usb_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].usb_audio);
+ break;
default:
return -EINVAL;
}
@@ -618,6 +655,9 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
{"TX_CODEC_DMA_TX_5", NULL, "TX_CODEC_DMA_TX_5 Capture"},
{"RX_CODEC_DMA_RX_6 Playback", NULL, "RX_CODEC_DMA_RX_6"},
{"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"},
+
+ /* USB playback AFE port receives data for playback, hence use the RX port */
+ {"USB Playback", NULL, "USB_RX"},
};
static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
@@ -645,6 +685,23 @@ static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
return 0;
}
+static const struct snd_soc_dai_ops q6afe_usb_ops = {
+ .probe = msm_dai_q6_dai_probe,
+ .prepare = q6afe_dai_prepare,
+ .hw_params = q6afe_usb_hw_params,
+ /*
+ * Shutdown callback required to stop the USB AFE port, which is enabled
+ * by the prepare() stage. This stops the audio traffic on the USB AFE
+ * port on the Q6DSP.
+ */
+ .shutdown = q6afe_dai_shutdown,
+ /*
+ * Startup callback not needed, as AFE port start command passes the PCM
+ * parameters within the AFE command, which is provided by the PCM core
+ * during the prepare() stage.
+ */
+};
+
static const struct snd_soc_dai_ops q6hdmi_ops = {
.probe = msm_dai_q6_dai_probe,
.remove = msm_dai_q6_dai_remove,
@@ -943,6 +1000,8 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_7", "NULL",
0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_AIF_IN("USB_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_component_driver q6afe_dai_component = {
@@ -1057,6 +1116,7 @@ static int q6afe_dai_dev_probe(struct platform_device *pdev)
cfg.q6i2s_ops = &q6i2s_ops;
cfg.q6tdm_ops = &q6tdm_ops;
cfg.q6dma_ops = &q6dma_ops;
+ cfg.q6usb_ops = &q6afe_usb_ops;
dais = q6dsp_audio_ports_set_config(dev, &cfg, &num_dais);
return devm_snd_soc_register_component(dev, &q6afe_dai_component, dais, num_dais);
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index ef7557be5d66..948007955934 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -35,6 +35,8 @@
#define AFE_MODULE_TDM 0x0001028A
#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5
+#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA
#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
#define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG 0x00010239
@@ -44,6 +46,7 @@
#define AFE_PARAM_ID_TDM_CONFIG 0x0001029D
#define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297
#define AFE_PARAM_ID_CODEC_DMA_CONFIG 0x000102B8
+#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4
#define AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f4
#define AFE_CMD_RSP_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f5
#define AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST 0x000100f6
@@ -72,12 +75,16 @@
#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
#define AFE_LINEAR_PCM_DATA 0x0
+#define AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG 0x1
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
#define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020
+/* USB AFE port */
+#define AFE_PORT_ID_USB_RX 0x7000
+
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
/* Clock set API version */
#define AFE_API_VERSION_CLOCK_SET 1
@@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg {
u16 active_channels_mask;
} __packed;
+struct afe_param_id_usb_cfg {
+/* Minor version used for tracking USB audio device configuration.
+ * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ */
+ u32 cfg_minor_version;
+/* Sampling rate of the port.
+ * Supported values:
+ * - AFE_PORT_SAMPLE_RATE_8K
+ * - AFE_PORT_SAMPLE_RATE_11025
+ * - AFE_PORT_SAMPLE_RATE_12K
+ * - AFE_PORT_SAMPLE_RATE_16K
+ * - AFE_PORT_SAMPLE_RATE_22050
+ * - AFE_PORT_SAMPLE_RATE_24K
+ * - AFE_PORT_SAMPLE_RATE_32K
+ * - AFE_PORT_SAMPLE_RATE_44P1K
+ * - AFE_PORT_SAMPLE_RATE_48K
+ * - AFE_PORT_SAMPLE_RATE_96K
+ * - AFE_PORT_SAMPLE_RATE_192K
+ */
+ u32 sample_rate;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 bit_width;
+/* Number of channels.
+ * Supported values: 1 and 2
+ */
+ u16 num_channels;
+/* Data format supported by the USB. The supported value is
+ * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM).
+ */
+ u16 data_format;
+/* this field must be 0 */
+ u16 reserved;
+/* device token of actual end USB audio device */
+ u32 dev_token;
+/* endianness of this interface */
+ u32 endian;
+/* service interval */
+ u32 service_interval;
+} __packed;
+
+/**
+ * struct afe_param_id_usb_audio_dev_params
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @dev_token: device token of actual end USB audio device
+ **/
+struct afe_param_id_usb_audio_dev_params {
+ u32 cfg_minor_version;
+ u32 dev_token;
+} __packed;
+
+/**
+ * struct afe_param_id_usb_audio_dev_lpcm_fmt
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @endian: endianness of this interface
+ **/
+struct afe_param_id_usb_audio_dev_lpcm_fmt {
+ u32 cfg_minor_version;
+ u32 endian;
+} __packed;
+
+#define AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL 0x000102B7
+
+/**
+ * struct afe_param_id_usb_audio_svc_interval
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @svc_interval: service interval
+ **/
+struct afe_param_id_usb_audio_svc_interval {
+ u32 cfg_minor_version;
+ u32 svc_interval;
+} __packed;
+
union afe_port_config {
struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
struct afe_param_id_slimbus_cfg slim_cfg;
struct afe_param_id_i2s_cfg i2s_cfg;
struct afe_param_id_tdm_cfg tdm_cfg;
struct afe_param_id_cdc_dma_cfg dma_cfg;
+ struct afe_param_id_usb_cfg usb_cfg;
} __packed;
@@ -833,6 +924,7 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
RX_CODEC_DMA_RX_6, 1, 1},
[RX_CODEC_DMA_RX_7] = { AFE_PORT_ID_RX_CODEC_DMA_RX_7,
RX_CODEC_DMA_RX_7, 1, 1},
+ [USB_RX] = { AFE_PORT_ID_USB_RX, USB_RX, 1, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -1290,6 +1382,100 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port,
}
EXPORT_SYMBOL_GPL(q6afe_tdm_port_prepare);
+/**
+ * afe_port_send_usb_dev_param() - Send USB dev token
+ *
+ * @port: Instance of afe port
+ * @cardidx: USB SND card index to reference
+ * @pcmidx: USB SND PCM device index to reference
+ *
+ * The USB dev token carries information about which USB SND card instance and
+ * PCM device to execute the offload on. This information is carried through
+ * to the stream enable QMI request, which is handled by the offload class
+ * driver. The information is parsed to determine which USB device to query
+ * the required resources for.
+ */
+int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx)
+{
+ struct afe_param_id_usb_audio_dev_params usb_dev;
+ int ret;
+
+ memset(&usb_dev, 0, sizeof(usb_dev));
+
+ usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+ usb_dev.dev_token = (cardidx << 16) | (pcmidx << 8);
+ ret = q6afe_port_set_param_v2(port, &usb_dev,
+ AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS,
+ AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(usb_dev));
+ if (ret)
+ dev_err(port->afe->dev, "%s: AFE device param cmd failed %d\n",
+ __func__, ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(afe_port_send_usb_dev_param);
+
+static int afe_port_send_usb_params(struct q6afe_port *port, struct q6afe_usb_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+ struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt;
+ struct afe_param_id_usb_audio_svc_interval svc_int;
+ int ret;
+
+ if (!pcfg) {
+ dev_err(port->afe->dev, "%s: Error, no configuration data\n", __func__);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ memset(&lpcm_fmt, 0, sizeof(lpcm_fmt));
+ memset(&svc_int, 0, sizeof(svc_int));
+
+ lpcm_fmt.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+ lpcm_fmt.endian = pcfg->usb_cfg.endian;
+ ret = q6afe_port_set_param_v2(port, &lpcm_fmt,
+ AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT,
+ AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(lpcm_fmt));
+ if (ret) {
+ dev_err(port->afe->dev, "%s: AFE device param cmd LPCM_FMT failed %d\n",
+ __func__, ret);
+ goto exit;
+ }
+
+ svc_int.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+ svc_int.svc_interval = pcfg->usb_cfg.service_interval;
+ ret = q6afe_port_set_param_v2(port, &svc_int,
+ AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL,
+ AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(svc_int));
+ if (ret)
+ dev_err(port->afe->dev, "%s: AFE device param cmd svc_interval failed %d\n",
+ __func__, ret);
+
+exit:
+ return ret;
+}
+
+/**
+ * q6afe_usb_port_prepare() - Prepare usb afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: USB configuration for the afe port
+ *
+ */
+void q6afe_usb_port_prepare(struct q6afe_port *port,
+ struct q6afe_usb_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->usb_cfg.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+ pcfg->usb_cfg.sample_rate = cfg->sample_rate;
+ pcfg->usb_cfg.num_channels = cfg->num_channels;
+ pcfg->usb_cfg.bit_width = cfg->bit_width;
+
+ afe_port_send_usb_params(port, cfg);
+}
+EXPORT_SYMBOL_GPL(q6afe_usb_port_prepare);
+
/**
* q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
*
@@ -1612,7 +1798,10 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
break;
case AFE_PORT_ID_WSA_CODEC_DMA_RX_0 ... AFE_PORT_ID_RX_CODEC_DMA_RX_7:
cfg_type = AFE_PARAM_ID_CODEC_DMA_CONFIG;
- break;
+ break;
+ case AFE_PORT_ID_USB_RX:
+ cfg_type = AFE_PARAM_ID_USB_AUDIO_CONFIG;
+ break;
default:
dev_err(dev, "Invalid port id 0x%x\n", port_id);
return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 65d0676075e1..04c70f904b86 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -3,7 +3,7 @@
#ifndef __Q6AFE_H__
#define __Q6AFE_H__
-#define AFE_PORT_MAX 129
+#define AFE_PORT_MAX 137
#define MSM_AFE_PORT_TYPE_RX 0
#define MSM_AFE_PORT_TYPE_TX 1
@@ -203,6 +203,36 @@ struct q6afe_cdc_dma_cfg {
u16 active_channels_mask;
};
+/**
+ * struct q6afe_usb_cfg
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @sample_rate: Sampling rate of the port
+ * Supported values:
+ * AFE_PORT_SAMPLE_RATE_8K
+ * AFE_PORT_SAMPLE_RATE_11025
+ * AFE_PORT_SAMPLE_RATE_12K
+ * AFE_PORT_SAMPLE_RATE_16K
+ * AFE_PORT_SAMPLE_RATE_22050
+ * AFE_PORT_SAMPLE_RATE_24K
+ * AFE_PORT_SAMPLE_RATE_32K
+ * AFE_PORT_SAMPLE_RATE_44P1K
+ * AFE_PORT_SAMPLE_RATE_48K
+ * AFE_PORT_SAMPLE_RATE_96K
+ * AFE_PORT_SAMPLE_RATE_192K
+ * @bit_width: Bit width of the sample.
+ * Supported values: 16, 24
+ * @num_channels: Number of channels
+ * Supported values: 1, 2
+ **/
+struct q6afe_usb_cfg {
+ u32 cfg_minor_version;
+ u32 sample_rate;
+ u16 bit_width;
+ u16 num_channels;
+};
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
@@ -210,6 +240,7 @@ struct q6afe_port_config {
struct q6afe_i2s_cfg i2s_cfg;
struct q6afe_tdm_cfg tdm;
struct q6afe_cdc_dma_cfg dma_cfg;
+ struct q6afe_usb_cfg usb_audio;
};
struct q6afe_port;
@@ -219,6 +250,8 @@ int q6afe_port_start(struct q6afe_port *port);
int q6afe_port_stop(struct q6afe_port *port);
void q6afe_port_put(struct q6afe_port *port);
int q6afe_get_port_id(int index);
+void q6afe_usb_port_prepare(struct q6afe_port *port,
+ struct q6afe_usb_cfg *cfg);
void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
void q6afe_slim_port_prepare(struct q6afe_port *port,
@@ -228,6 +261,7 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
void q6afe_cdc_dma_port_prepare(struct q6afe_port *port,
struct q6afe_cdc_dma_cfg *cfg);
+int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx);
int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
int clk_src, int clk_root,
unsigned int freq, int dir);
diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
index 4919001de08b..4a96b11f7fd1 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
+++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
@@ -97,6 +97,26 @@
}
static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = {
+ {
+ .playback = {
+ .stream_name = "USB Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .id = USB_RX,
+ .name = "USB_RX",
+ },
{
.playback = {
.stream_name = "HDMI Playback",
@@ -624,6 +644,9 @@ struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev,
case WSA_CODEC_DMA_RX_0 ... RX_CODEC_DMA_RX_7:
q6dsp_audio_fe_dais[i].ops = cfg->q6dma_ops;
break;
+ case USB_RX:
+ q6dsp_audio_fe_dais[i].ops = cfg->q6usb_ops;
+ break;
default:
break;
}
diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
index 7f052c8a1257..d8dde6dd0aca 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
+++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
@@ -11,6 +11,7 @@ struct q6dsp_audio_port_dai_driver_config {
const struct snd_soc_dai_ops *q6i2s_ops;
const struct snd_soc_dai_ops *q6tdm_ops;
const struct snd_soc_dai_ops *q6dma_ops;
+ const struct snd_soc_dai_ops *q6usb_ops;
};
struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 81fde0681f95..45811f0f0d44 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -515,6 +515,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
return 1;
}
+static const struct snd_kcontrol_new usb_mixer_controls[] = {
+ Q6ROUTING_RX_MIXERS(USB_RX) };
+
static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
Q6ROUTING_RX_MIXERS(HDMI_RX) };
@@ -950,6 +953,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
+ SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0,
+ usb_mixer_controls,
+ ARRAY_SIZE(usb_mixer_controls)),
+
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -1043,6 +1050,8 @@ static const struct snd_soc_dapm_route intercon[] = {
{"MM_UL6", NULL, "MultiMedia6 Mixer"},
{"MM_UL7", NULL, "MultiMedia7 Mixer"},
{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+
+ Q6ROUTING_RX_DAPM_ROUTE("USB Mixer", "USB_RX"),
};
static int routing_hw_params(struct snd_soc_component *component,
next prev parent reply other threads:[~2024-04-25 21:52 UTC|newest]
Thread overview: 55+ messages / expand[flat|nested] mbox.gz Atom feed top
2024-04-25 21:50 [PATCH v20 00/41] Introduce QC USB SND audio offloading support Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 01/41] xhci: add helper to stop endpoint and wait for completion Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 02/41] usb: host: xhci: Export enable and disable interrupter APIs Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 03/41] usb: host: xhci: Repurpose event handler for skipping interrupter events Wesley Cheng
2024-04-30 11:02 ` Mathias Nyman
2024-04-30 21:57 ` Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 04/41] xhci: export XHCI IMOD setting helper for interrupters Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 05/41] xhci: sideband: add initial api to register a sideband entity Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 06/41] usb: host: xhci-sideband: Expose a sideband interrupter enable API Wesley Cheng
2024-05-02 11:07 ` Mathias Nyman
2024-05-02 23:37 ` Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 07/41] usb: host: xhci-mem: Cleanup pending secondary event ring events Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 08/41] usb: host: xhci-mem: Allow for interrupter clients to choose specific index Wesley Cheng
2024-04-26 13:24 ` Amadeusz Sławiński
2024-04-26 20:03 ` Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 09/41] ASoC: Add SOC USB APIs for adding an USB backend Wesley Cheng
2024-04-26 13:25 ` Amadeusz Sławiński
2024-04-26 21:55 ` Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 10/41] ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port Wesley Cheng
2024-04-25 21:50 ` Wesley Cheng [this message]
2024-04-25 21:50 ` [PATCH v20 12/41] ASoC: qdsp6: q6afe: Increase APR timeout Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 13/41] ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 14/41] ALSA: usb-audio: Introduce USB SND platform op callbacks Wesley Cheng
2024-04-25 21:50 ` [PATCH v20 15/41] ALSA: usb-audio: Export USB SND APIs for modules Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 16/41] ALSA: usb-audio: Save UAC sample size information Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 17/41] usb: dwc3: Specify maximum number of XHCI interrupters Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 18/41] usb: host: xhci-plat: Set XHCI max interrupters if property is present Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 19/41] ALSA: usb-audio: qcom: Add USB QMI definitions Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 20/41] ALSA: usb-audio: qcom: Introduce QC USB SND offloading support Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 21/41] ALSA: usb-audio: Check for support for requested audio format Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 22/41] ASoC: usb: Add PCM format check API for USB backend Wesley Cheng
2024-04-26 13:25 ` Amadeusz Sławiński
2024-04-30 8:01 ` Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 23/41] ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 24/41] ALSA: usb-audio: Prevent starting of audio stream if in use Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 25/41] ALSA: usb-audio: Do not allow USB offload path if PCM device is " Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 26/41] ASoC: dt-bindings: Update example for enabling USB offload on SM8250 Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 27/41] ALSA: usb-audio: qcom: Populate PCM and USB chip information Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 28/41] ASoC: qcom: qdsp6: Add support to track available USB PCM devices Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 29/41] ASoC: Introduce SND kcontrols to select sound card and PCM device Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 30/41] ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 31/41] ASoC: Introduce SND kcontrols to track USB offloading state Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 32/41] ASoC: qcom: qdsp6: Add PCM ops to track current state Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 33/41] ASoC: usb: Create SOC USB SND jack kcontrol Wesley Cheng
2024-04-26 13:26 ` Amadeusz Sławiński
2024-05-01 0:32 ` Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 34/41] ASoC: qcom: qdsp6: Add headphone jack for offload connection status Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 35/41] ASoC: usb: Fetch ASoC sound card information Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 36/41] ALSA: usb-audio: Add USB offloading capable kcontrol Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 37/41] ALSA: usb-audio: Allow for rediscovery of connected USB SND devices Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 38/41] ALSA: usb-audio: qcom: Use card and PCM index from QMI request Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 39/41] ASoC: usb: Rediscover USB SND devices on USB port add Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 40/41] ASoC: qcom: Populate SoC components string Wesley Cheng
2024-04-25 21:51 ` [PATCH v20 41/41] ASoC: doc: Add documentation for SOC USB Wesley Cheng
2024-04-26 3:36 ` Bagas Sanjaya
Reply instructions:
You may reply publicly to this message via plain-text email
using any one of the following methods:
* Save the following mbox file, import it into your mail client,
and reply-to-all from there: mbox
Avoid top-posting and favor interleaved quoting:
https://en.wikipedia.org/wiki/Posting_style#Interleaved_style
* Reply using the --to, --cc, and --in-reply-to
switches of git-send-email(1):
git send-email \
--in-reply-to=20240425215125.29761-12-quic_wcheng@quicinc.com \
--to=quic_wcheng@quicinc.com \
--cc=Thinh.Nguyen@synopsys.com \
--cc=alsa-devel@alsa-project.org \
--cc=andersson@kernel.org \
--cc=bgoswami@quicinc.com \
--cc=broonie@kernel.org \
--cc=conor+dt@kernel.org \
--cc=corbet@lwn.net \
--cc=devicetree@vger.kernel.org \
--cc=gregkh@linuxfoundation.org \
--cc=konrad.dybcio@linaro.org \
--cc=krzysztof.kozlowski+dt@linaro.org \
--cc=lgirdwood@gmail.com \
--cc=linux-arm-msm@vger.kernel.org \
--cc=linux-doc@vger.kernel.org \
--cc=linux-kernel@vger.kernel.org \
--cc=linux-sound@vger.kernel.org \
--cc=linux-usb@vger.kernel.org \
--cc=mathias.nyman@intel.com \
--cc=perex@perex.cz \
--cc=robh@kernel.org \
--cc=srinivas.kandagatla@linaro.org \
--cc=tiwai@suse.com \
/path/to/YOUR_REPLY
https://kernel.org/pub/software/scm/git/docs/git-send-email.html
* If your mail client supports setting the In-Reply-To header
via mailto: links, try the mailto: link
Be sure your reply has a Subject: header at the top and a blank line
before the message body.
This is a public inbox, see mirroring instructions
for how to clone and mirror all data and code used for this inbox;
as well as URLs for read-only IMAP folder(s) and NNTP newsgroup(s).