ALSA: bebob: Prepare for device specific operations

This commit is for some devices which have its own operations or quirks.

Many functionality should be implemented in user land. Then this commit adds
functionality related to stream such as sampling frequency or clock source. For
help to debug, this commit adds the functionality to get metering information
if it's available.

To help these functionalities, this commit adds some AV/C commands defined in
'AV/C Audio Subunit Specification (1394TA).

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

1 files changed, 22 insertions, 5 deletions

diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 85b4fd661787..4a21dcf95d95 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -119,13 +119,27 @@ end:
 int
 snd_bebob_stream_check_internal_clock(struct snd_bebob *bebob, bool *internal)
 {
+	struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
 	u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
+	unsigned int id;
 	int err = 0;
 
 	*internal = false;
 
+	/* 1.The device has its own operation to switch source of clock */
+	if (clk_spec) {
+		err = clk_spec->get(bebob, &id);
+		if (err < 0)
+			dev_err(&bebob->unit->device,
+				"fail to get clock source: %d\n", err);
+		else if (strncmp(clk_spec->labels[id], SND_BEBOB_CLOCK_INTERNAL,
+				 strlen(SND_BEBOB_CLOCK_INTERNAL)) == 0)
+			*internal = true;
+		goto end;
+	}
+
 	/*
-	 * 1.The device don't support to switch source of clock then assumed
+	 * 2.The device don't support to switch source of clock then assumed
 	 *   to use internal clock always
 	 */
 	if (bebob->sync_input_plug < 0) {
@@ -134,7 +148,7 @@ snd_bebob_stream_check_internal_clock(struct snd_bebob *bebob, bool *internal)
 	}
 
 	/*
-	 * 2.The device supports to switch source of clock by an usual way.
+	 * 3.The device supports to switch source of clock by an usual way.
 	 *   Let's check input for 'Music Sub Unit Sync Input' plug.
 	 */
 	avc_bridgeco_fill_msu_addr(addr, AVC_BRIDGECO_PLUG_DIR_IN,
@@ -442,6 +456,7 @@ end:
 
 int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, int rate)
 {
+	struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
 	struct amdtp_stream *master, *slave;
 	atomic_t *slave_substreams;
 	enum cip_flags sync_mode;
@@ -508,7 +523,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, int rate)
 		break_both_connections(bebob);
 
 	/* stop streams if rate is different */
-	err = snd_bebob_stream_get_rate(bebob, &curr_rate);
+	err = rate_spec->get(bebob, &curr_rate);
 	if (err < 0) {
 		dev_err(&bebob->unit->device,
 			"fail to get sampling rate: %d\n", err);
@@ -532,7 +547,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, int rate)
 		 * If establishing connections at first, Yamaha GO46
 		 * (and maybe Terratec X24) don't generate sound.
 		 */
-		err = snd_bebob_stream_set_rate(bebob, rate);
+		err = rate_spec->set(bebob, rate);
 		if (err < 0) {
 			dev_err(&bebob->unit->device,
 				"fail to set sampling rate: %d\n",
@@ -822,6 +837,7 @@ end:
 
 int snd_bebob_stream_discover(struct snd_bebob *bebob)
 {
+	struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
 	u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
 	enum avc_bridgeco_plug_type type;
 	unsigned int i;
@@ -908,7 +924,8 @@ int snd_bebob_stream_discover(struct snd_bebob *bebob)
 	}
 
 	/* for check source of clock later */
-	err = seek_msu_sync_input_plug(bebob);
+	if (!clk_spec)
+		err = seek_msu_sync_input_plug(bebob);
 end:
 	return err;
 }