usb: gadget: uac: add support of different sample rate

The sample rate is force to 44100 in current code. This
code add numbers of sample rate up to 15 and support
set or get by PC.

Change-Id: I543d84f3081f82a4c96071c2b2442d37a9a835fb
Signed-off-by: Meng Dongyang <daniel.meng@rock-chips.com>

Author: Meng Dongyang

drivers/usb/gadget/function/f_audio_source.c

@@ -29,8 +29,10 @@
 #include <linux/usb/composite.h>
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#define SAMPLE_RATE 44100
-#define FRAMES_PER_MSEC (SAMPLE_RATE / 1000)
+#define DEFAULT_SAMPLE_RATE 44100
+#define MAX_SAMPLE_RATE		96000
+#define MIN_SAMPLE_RATE		8000
+#define DEFAULT_FRAMES_PER_MSEC (DEFAULT_SAMPLE_RATE / 1000)
 
 #define IN_EP_MAX_PACKET_SIZE 256
 
@@ -42,6 +44,12 @@
 #define AUDIO_NUM_INTERFACES	2
 #define MAX_INST_NAME_LEN     40
 
+static u32 sample_rate_table[] = {
+8000, 11025, 16000, 22050, 24000,
+32000, 40000, 44100, 48000, 56000,
+64000, 72000, 80000, 88200, 96000,
+};
+
 /* B.3.1  Standard AC Interface Descriptor */
 static struct usb_interface_descriptor ac_interface_desc = {
 	.bLength =		USB_DT_INTERFACE_SIZE,
@@ -135,16 +143,16 @@ static struct uac1_as_header_descriptor as_header_desc = {
 	.wFormatTag =		UAC_FORMAT_TYPE_I_PCM,
 };
 
-DECLARE_UAC_FORMAT_TYPE_I_DISCRETE_DESC(1);
+DECLARE_UAC_FORMAT_TYPE_I_DISCRETE_DESC(15);
 
-static struct uac_format_type_i_discrete_descriptor_1 as_type_i_desc = {
-	.bLength =		UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(1),
+static struct uac_format_type_i_discrete_descriptor_15 as_type_i_desc = {
+	.bLength =		UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(15),
 	.bDescriptorType =	USB_DT_CS_INTERFACE,
 	.bDescriptorSubtype =	UAC_FORMAT_TYPE,
 	.bFormatType =		UAC_FORMAT_TYPE_I,
 	.bSubframeSize =	2,
 	.bBitResolution =	16,
-	.bSamFreqType =		1,
+	.bSamFreqType =		ARRAY_SIZE(sample_rate_table),
 };
 
 /* Standard ISO IN Endpoint Descriptor for highspeed */
@@ -227,8 +235,8 @@ static struct snd_pcm_hardware audio_hw_info = {
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.channels_min		= 2,
 	.channels_max		= 2,
-	.rate_min		= SAMPLE_RATE,
-	.rate_max		= SAMPLE_RATE,
+	.rate_min		= MIN_SAMPLE_RATE,
+	.rate_max		= MAX_SAMPLE_RATE,
 
 	.buffer_bytes_max =	1024 * 1024,
 	.period_bytes_min =	64,
@@ -239,6 +247,9 @@ static struct snd_pcm_hardware audio_hw_info = {
 
 /*-------------------------------------------------------------------------*/
 
+static u16 g_w_value;
+static u8 g_bRequest;
+
 struct audio_source_config {
 	int	card;
 	int	device;
@@ -275,6 +286,13 @@ struct audio_dev {
 	ktime_t				start_time;
 	/* number of frames sent since start_time */
 	s64				frames_sent;
+
+	/* number of frames sent per millisecond */
+	s64				frames_per_msec;
+
+	/* current sample rate */
+	s64				sample_rate;
+
 	struct audio_source_config	*config;
 	/* for creating and issuing QoS requests */
 	struct pm_qos_request pm_qos;
@@ -468,20 +486,20 @@ static void audio_send(struct audio_dev *audio)
 	now = ktime_get();
 	msecs = div_s64((ktime_to_ns(now) - ktime_to_ns(audio->start_time)),
 			1000000);
-	frames = div_s64((msecs * SAMPLE_RATE), 1000);
+	frames = div_s64((msecs * audio->sample_rate), 1000);
 
 	/* Readjust our frames_sent if we fall too far behind.
 	 * If we get too far behind it is better to drop some frames than
 	 * to keep sending data too fast in an attempt to catch up.
 	 */
-	if (frames - audio->frames_sent > 10 * FRAMES_PER_MSEC)
-		audio->frames_sent = frames - FRAMES_PER_MSEC;
+	if (frames - audio->frames_sent > 10 * audio->frames_per_msec)
+		audio->frames_sent = frames - audio->frames_per_msec;
 
 	frames -= audio->frames_sent;
 
 	/* We need to send something to keep the pipeline going */
 	if (frames <= 0)
-		frames = FRAMES_PER_MSEC;
+		frames = audio->frames_per_msec;
 
 	while (frames > 0) {
 		req = audio_req_get(audio);
@@ -526,7 +544,40 @@ static void audio_send(struct audio_dev *audio)
 
 static void audio_control_complete(struct usb_ep *ep, struct usb_request *req)
 {
-	/* nothing to do here */
+	struct audio_dev *audio = ep->driver_data;
+	u8 *buf = req->buf;
+
+	pr_debug("audio_control_complete req->status %d req->actual %d\n",
+		 req->status, req->actual);
+
+	if (req->status)
+		return;
+
+	if (g_w_value == UAC_EP_CS_ATTR_SAMPLE_RATE << 8) {
+		switch (g_bRequest) {
+		case UAC_SET_CUR:
+			g_w_value = 0;
+			g_bRequest = 0;
+
+			audio->sample_rate = (s64)buf[0];
+			audio->sample_rate += ((s64)buf[1]) << 8;
+			audio->sample_rate += ((s64)buf[2]) << 16;
+			audio->frames_per_msec = audio->sample_rate;
+			do_div(audio->frames_per_msec, 1000);
+			pr_info("audio source set sample rate to %lld\n",
+				audio->sample_rate);
+			break;
+		case UAC_SET_MIN:
+			/* fallthrough */
+		case UAC_SET_MAX:
+			/* fallthrough */
+		case UAC_SET_RES:
+			/* fallthrough */
+			break;
+		default:
+			break;
+		}
+	}
 }
 
 static void audio_data_complete(struct usb_ep *ep, struct usb_request *req)
@@ -562,8 +613,15 @@ static int audio_set_endpoint_req(struct usb_function *f,
 
 	switch (ctrl->bRequest) {
 	case UAC_SET_CUR:
+		if (w_value == UAC_EP_CS_ATTR_SAMPLE_RATE << 8) {
+			g_w_value = w_value;
+			g_bRequest = ctrl->bRequest;
+		}
+		/* fallthrough */
 	case UAC_SET_MIN:
+		/* fallthrough */
 	case UAC_SET_MAX:
+		/* fallthrough */
 	case UAC_SET_RES:
 		value = len;
 		break;
@@ -577,6 +635,7 @@ static int audio_set_endpoint_req(struct usb_function *f,
 static int audio_get_endpoint_req(struct usb_function *f,
 		const struct usb_ctrlrequest *ctrl)
 {
+	struct audio_dev *audio = func_to_audio(f);
 	struct usb_composite_dev *cdev = f->config->cdev;
 	int value = -EOPNOTSUPP;
 	u8 ep = ((le16_to_cpu(ctrl->wIndex) >> 8) & 0xFF);
@@ -590,13 +649,31 @@ static int audio_get_endpoint_req(struct usb_function *f,
 	if (w_value == UAC_EP_CS_ATTR_SAMPLE_RATE << 8) {
 		switch (ctrl->bRequest) {
 		case UAC_GET_CUR:
+			buf[0] = (u8)audio->sample_rate;
+			buf[1] = (u8)(audio->sample_rate >> 8);
+			buf[2] = (u8)(audio->sample_rate >> 16);
+			value = 3;
+			break;
+
 		case UAC_GET_MIN:
+			buf[0] = (u8)MIN_SAMPLE_RATE;
+			buf[1] = (u8)(MIN_SAMPLE_RATE >> 8);
+			buf[2] = (u8)(MIN_SAMPLE_RATE >> 16);
+			value = 3;
+			break;
+
 		case UAC_GET_MAX:
+			buf[0] = (u8)MAX_SAMPLE_RATE;
+			buf[1] = (u8)(MAX_SAMPLE_RATE >> 8);
+			buf[2] = (u8)(MAX_SAMPLE_RATE >> 16);
+			value = 3;
+			break;
+
 		case UAC_GET_RES:
 			/* return our sample rate */
-			buf[0] = (u8)SAMPLE_RATE;
-			buf[1] = (u8)(SAMPLE_RATE >> 8);
-			buf[2] = (u8)(SAMPLE_RATE >> 16);
+			buf[0] = (u8)MAX_SAMPLE_RATE;
+			buf[1] = (u8)(MAX_SAMPLE_RATE >> 8);
+			buf[2] = (u8)(MAX_SAMPLE_RATE >> 16);
 			value = 3;
 			break;
 		default:
@@ -686,16 +763,18 @@ static void audio_free_func(struct usb_function *f)
 static void audio_build_desc(struct audio_dev *audio)
 {
 	u8 *sam_freq;
-	int rate;
+	u32 rate, i;
 
 	/* Set channel numbers */
 	input_terminal_desc.bNrChannels = 2;
 	as_type_i_desc.bNrChannels = 2;
 
 	/* Set sample rates */
-	rate = SAMPLE_RATE;
-	sam_freq = as_type_i_desc.tSamFreq[0];
-	memcpy(sam_freq, &rate, 3);
+	for (i = 0; i < ARRAY_SIZE(sample_rate_table); i++) {
+		rate = sample_rate_table[i];
+		sam_freq = as_type_i_desc.tSamFreq[i];
+		memcpy(sam_freq, &rate, 3);
+	}
 }
 
 
@@ -723,6 +802,10 @@ audio_bind(struct usb_configuration *c, struct usb_function *f)
 		struct audio_source_config *config =
 				fi_audio->config;
 
+		audio->alt = 0;
+		audio->sample_rate = DEFAULT_SAMPLE_RATE;
+		audio->frames_per_msec = DEFAULT_FRAMES_PER_MSEC;
+
 		err = snd_card_setup(c, config);
 		if (err)
 			return err;
@@ -874,7 +957,7 @@ static int audio_pcm_hw_params(struct snd_pcm_substream *substream,
 	unsigned int channels = params_channels(params);
 	unsigned int rate = params_rate(params);
 
-	if (rate != SAMPLE_RATE)
+	if (rate > MAX_SAMPLE_RATE || rate < MIN_SAMPLE_RATE)
 		return -EINVAL;
 	if (channels != 2)
 		return -EINVAL;
@@ -973,6 +1056,10 @@ int audio_source_bind_config(struct usb_configuration *c,
 
 	audio = &_audio_dev;
 
+	audio->alt = 0;
+	audio->sample_rate = DEFAULT_SAMPLE_RATE;
+	audio->frames_per_msec = DEFAULT_FRAMES_PER_MSEC;
+
 	err = snd_card_setup(c, config);
 	if (err)
 		return err;