ALSA: dice: purge generating channel cache

Dice interface design doesn't allow drivers to read supported combination
between sampling transfer frequencies and the number of Multi bit linear
audio data channels. Due to the design, ALSA dice driver changes current
sampling transfer frequency to generate cache of the combinations at
device probing processing.

Although, this idea is worse because ALSA dice driver changes the state of
clock. This is not what users want when they save favorite configuration
to the device in advance.

Furthermore, there's a possibility that the format of data block is decided
not only according to current sampling transfer frequency, but also the
other factors, i.e. data format for digital interface. It's not good to
generate channel cache according to the sampling transfer frequency only.

This commit purges processing cache data and related structure members. As
a result, users must set preferable sampling transfer frequency before
using ALSA PCM applications, as long as they want to start any PCM
substreams at the rate except for current one.

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

1 files changed, 2 insertions, 22 deletions

diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 4f74e3ef58fd..716db092d7c7 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -24,23 +24,6 @@ const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = {
 	[6] = 192000,
 };
 
-int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
-				  unsigned int *mode)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) {
-		if (!(dice->clock_caps & BIT(i)))
-			continue;
-		if (snd_dice_rates[i] != rate)
-			continue;
-
-		*mode = (i - 1) / 2;
-		return 0;
-	}
-	return -EINVAL;
-}
-
 static void release_resources(struct snd_dice *dice,
 			      struct fw_iso_resources *resources)
 {
@@ -100,13 +83,10 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
 {
 	struct fw_iso_resources *resources;
 	__be32 reg[2];
-	unsigned int i, mode, pcm_chs, midi_ports;
+	unsigned int i, pcm_chs, midi_ports;
 	bool double_pcm_frames;
 	int err;
 
-	err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
-	if (err < 0)
-		goto end;
 	if (stream == &dice->tx_stream) {
 		resources = &dice->tx_resources;
 		err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO,
@@ -133,7 +113,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
 	 * For this quirk, blocking mode is required and PCM buffer size should
 	 * be aligned to SYT_INTERVAL.
 	 */
-	double_pcm_frames = mode > 1;
+	double_pcm_frames = rate > 96000;
 	if (double_pcm_frames) {
 		rate /= 2;
 		pcm_chs *= 2;