Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel…

…/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: usb/endpoint, fix dangling pointer use
  ALSA: asihpi - Get rid of incorrect "long" types and casts.
  ASoC: DaVinci: Fix McASP hardware FIFO configuration
  ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
  ALSA: usb-audio: fix UAC2 control value queries
  ALSA: usb-audio: parse UAC2 sample rate ranges correctly
  ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
  ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
  ALSA: hda - Don't check capture source mixer if no ADC is available

sound/pci/asihpi/hpi6205.c

@@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
 
 }
 
-static long outstream_get_space_available(struct hpi_hostbuffer_status
+static u32 outstream_get_space_available(struct hpi_hostbuffer_status
 	*status)
 {
-	return status->size_in_bytes - ((long)(status->host_index) -
-		(long)(status->dSP_index));
+	return status->size_in_bytes - (status->host_index -
+		status->dSP_index);
 }
 
 static void outstream_write(struct hpi_adapter_obj *pao,
@@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	struct hpi_hw_obj *phw = pao->priv;
 	struct bus_master_interface *interface = phw->p_interface_buffer;
 	struct hpi_hostbuffer_status *status;
-	long space_available;
+	u32 space_available;
 
 	if (!phw->outstream_host_buffer_size[phm->obj_index]) {
 		/* there  is no BBM buffer, write via message */
@@ -1007,7 +1007,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	}
 
 	space_available = outstream_get_space_available(status);
-	if (space_available < (long)phm->u.d.u.data.data_size) {
+	if (space_available < phm->u.d.u.data.data_size) {
 		phr->error = HPI_ERROR_INVALID_DATASIZE;
 		return;
 	}
@@ -1018,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 		&& hpios_locked_mem_valid(&phw->outstream_host_buffers[phm->
 				obj_index])) {
 		u8 *p_bbm_data;
-		long l_first_write;
+		u32 l_first_write;
 		u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
 		if (hpios_locked_mem_get_virt_addr(&phw->
@@ -1248,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao,
 	hw_message(pao, phm, phr);
 }
 
-static long instream_get_bytes_available(struct hpi_hostbuffer_status *status)
+static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status)
 {
-	return (long)(status->dSP_index) - (long)(status->host_index);
+	return status->dSP_index - status->host_index;
 }
 
 static void instream_read(struct hpi_adapter_obj *pao,
@@ -1259,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao,
 	struct hpi_hw_obj *phw = pao->priv;
 	struct bus_master_interface *interface = phw->p_interface_buffer;
 	struct hpi_hostbuffer_status *status;
-	long data_available;
+	u32 data_available;
 	u8 *p_bbm_data;
-	long l_first_read;
+	u32 l_first_read;
 	u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
 	if (!phw->instream_host_buffer_size[phm->obj_index]) {
@@ -1272,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao,
 
 	status = &interface->instream_host_buffer_status[phm->obj_index];
 	data_available = instream_get_bytes_available(status);
-	if (data_available < (long)phm->u.d.u.data.data_size) {
+	if (data_available < phm->u.d.u.data.data_size) {
 		phr->error = HPI_ERROR_INVALID_DATASIZE;
 		return;
 	}

sound/pci/hda/patch_realtek.c

@@ -2619,16 +2619,18 @@ static int alc_build_controls(struct hda_codec *codec)
 	}
 
 	/* assign Capture Source enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-	if (!kctl)
-		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-	for (i = 0; kctl && i < kctl->count; i++) {
-		hda_nid_t *nids = spec->capsrc_nids;
-		if (!nids)
-			nids = spec->adc_nids;
-		err = snd_hda_add_nid(codec, kctl, i, nids[i]);
-		if (err < 0)
-			return err;
+	if (spec->capsrc_nids || spec->adc_nids) {
+		kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
+		if (!kctl)
+			kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
+		for (i = 0; kctl && i < kctl->count; i++) {
+			hda_nid_t *nids = spec->capsrc_nids;
+			if (!nids)
+				nids = spec->adc_nids;
+			err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+			if (err < 0)
+				return err;
+		}
 	}
 	if (spec->cap_mixer) {
 		const char *kname = kctl ? kctl->id.name : NULL;
@@ -6948,7 +6950,7 @@ static struct hda_input_mux mb5_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x1 },
-		{ "Line", 0x2 },
+		{ "Line", 0x7 },
 		{ "CD", 0x4 },
 	},
 };
@@ -7469,8 +7471,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
@@ -7853,10 +7855,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
 	{ }
 };
 

sound/soc/davinci/davinci-mcasp.c

@@ -612,7 +612,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 								NUMDMA_MASK);
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
 				((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
 	}
 
 	if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -623,7 +622,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 								NUMDMA_MASK);
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
 				((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
 	}
 }
 

sound/usb/clock.c

@@ -103,7 +103,8 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i
 	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
 			      UAC2_CS_CUR,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
-			      UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8,
+			      UAC2_CX_CLOCK_SELECTOR << 8,
+			      snd_usb_ctrl_intf(chip) | (selector_id << 8),
 			      &buf, sizeof(buf), 1000);
 
 	if (ret < 0)
@@ -120,7 +121,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
 
 	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8,
+			      UAC2_CS_CONTROL_CLOCK_VALID << 8,
+			      snd_usb_ctrl_intf(chip) | (source_id << 8),
 			      &data, sizeof(data), 1000);
 
 	if (err < 0) {
@@ -269,7 +271,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 	data[3] = rate >> 24;
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8,
+				   snd_usb_ctrl_intf(chip) | (clock << 8),
 				   data, sizeof(data), 1000)) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
 			   dev->devnum, iface, fmt->altsetting, rate);
@@ -278,7 +281,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 
 	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8,
+				   snd_usb_ctrl_intf(chip) | (clock << 8),
 				   data, sizeof(data), 1000)) < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);

sound/usb/endpoint.c

@@ -427,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
 			kfree(fp->rate_table);
 			kfree(fp);
+			fp = NULL;
 			continue;
 		}
 

sound/usb/format.c

@@ -205,6 +205,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 	return 0;
 }
 
+/*
+ * Helper function to walk the array of sample rate triplets reported by
+ * the device. The problem is that we need to parse whole array first to
+ * get to know how many sample rates we have to expect.
+ * Then fp->rate_table can be allocated and filled.
+ */
+static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
+					const unsigned char *data)
+{
+	int i, nr_rates = 0;
+
+	fp->rates = fp->rate_min = fp->rate_max = 0;
+
+	for (i = 0; i < nr_triplets; i++) {
+		int min = combine_quad(&data[2 + 12 * i]);
+		int max = combine_quad(&data[6 + 12 * i]);
+		int res = combine_quad(&data[10 + 12 * i]);
+		int rate;
+
+		if ((max < 0) || (min < 0) || (res < 0) || (max < min))
+			continue;
+
+		/*
+		 * for ranges with res == 1, we announce a continuous sample
+		 * rate range, and this function should return 0 for no further
+		 * parsing.
+		 */
+		if (res == 1) {
+			fp->rate_min = min;
+			fp->rate_max = max;
+			fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+			return 0;
+		}
+
+		for (rate = min; rate <= max; rate += res) {
+			if (fp->rate_table)
+				fp->rate_table[nr_rates] = rate;
+			if (!fp->rate_min || rate < fp->rate_min)
+				fp->rate_min = rate;
+			if (!fp->rate_max || rate > fp->rate_max)
+				fp->rate_max = rate;
+			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+
+			nr_rates++;
+
+			/* avoid endless loop */
+			if (res == 0)
+				break;
+		}
+	}
+
+	return nr_rates;
+}
+
 /*
  * parse the format descriptor and stores the possible sample rates
  * on the audioformat table (audio class v2).
@@ -215,13 +269,20 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 {
 	struct usb_device *dev = chip->dev;
 	unsigned char tmp[2], *data;
-	int i, nr_rates, data_size, ret = 0;
+	int nr_triplets, data_size, ret = 0;
 	int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock);
 
+	if (clock < 0) {
+		snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n",
+				__func__, clock);
+		goto err;
+	}
+
 	/* get the number of sample rates first by only fetching 2 bytes */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
 			      tmp, sizeof(tmp), 1000);
 
 	if (ret < 0) {
@@ -230,8 +291,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 		goto err;
 	}
 
-	nr_rates = (tmp[1] << 8) | tmp[0];
-	data_size = 2 + 12 * nr_rates;
+	nr_triplets = (tmp[1] << 8) | tmp[0];
+	data_size = 2 + 12 * nr_triplets;
 	data = kzalloc(data_size, GFP_KERNEL);
 	if (!data) {
 		ret = -ENOMEM;
@@ -241,7 +302,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 	/* now get the full information */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
 			      data, data_size, 1000);
 
 	if (ret < 0) {
@@ -251,26 +313,28 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 		goto err_free;
 	}
 
-	fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
+	/* Call the triplet parser, and make sure fp->rate_table is NULL.
+	 * We just use the return value to know how many sample rates we
+	 * will have to deal with. */
+	kfree(fp->rate_table);
+	fp->rate_table = NULL;
+	fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data);
+
+	if (fp->nr_rates == 0) {
+		/* SNDRV_PCM_RATE_CONTINUOUS */
+		ret = 0;
+		goto err_free;
+	}
+
+	fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
 	if (!fp->rate_table) {
 		ret = -ENOMEM;
 		goto err_free;
 	}
 
-	fp->nr_rates = 0;
-	fp->rate_min = fp->rate_max = 0;
-
-	for (i = 0; i < nr_rates; i++) {
-		int rate = combine_quad(&data[2 + 12 * i]);
-
-		fp->rate_table[fp->nr_rates] = rate;
-		if (!fp->rate_min || rate < fp->rate_min)
-			fp->rate_min = rate;
-		if (!fp->rate_max || rate > fp->rate_max)
-			fp->rate_max = rate;
-		fp->rates |= snd_pcm_rate_to_rate_bit(rate);
-		fp->nr_rates++;
-	}
+	/* Call the triplet parser again, but this time, fp->rate_table is
+	 * allocated, so the rates will be stored */
+	parse_uac2_sample_rate_range(fp, nr_triplets, data);
 
 err_free:
 	kfree(data);

sound/usb/helper.h

@@ -28,5 +28,9 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
 #define snd_usb_get_speed(dev) ((dev)->speed)
 #endif
 
+static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
+{
+	return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
+}
 
 #endif /* __USBAUDIO_HELPER_H */