From a02cb8f8def634159b60c3a4d6ba16e471b46e0c Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Mon, 22 May 2017 22:22:21 +0900
Subject: ALSA: firewire: remove support for 16 bit PCM samples in playback
 substream

In IEC 61883-6, AM824 is described as format of data block. In this
format, one data block consists of several data channels, which is aligned
to 32 bit. One data channel has 8 bit label field and 24 bit data field.
PCM frames are transferred in Multi Bit Linear Audio (MBLA) data channel.
This channel can include 16/20/24 bit PCM sample.

As long as I know, models which support IEC 61883-1/6 doesn't allow to
switch bit length of PCM sample in MBLA data channel. They always
transmit/receive PCM frames of 24 bit length. This can be seen for the
other models which support protocols similar to IEC 61883-1/6.

On the other hand, current drivers for these protocols supports 16 bit
length PCM sample in playback substream. In this case, PCM sample is put
into the MBLA data channel with 8 bit padding in LSB side. Although 16
bit PCM sample is major because it's in CD format, this doesn't represent
device capability as is.

This commit removes support for 16 bit PCM samples in playback substream.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-am824.c             | 71 +-------------------------------
 sound/firewire/amdtp-am824.h             |  6 +--
 sound/firewire/bebob/bebob_pcm.c         |  4 --
 sound/firewire/dice/dice-pcm.c           |  6 ---
 sound/firewire/digi00x/amdtp-dot.c       | 63 +---------------------------
 sound/firewire/digi00x/digi00x-pcm.c     |  7 +---
 sound/firewire/digi00x/digi00x.h         |  1 -
 sound/firewire/fireworks/fireworks_pcm.c |  4 --
 sound/firewire/oxfw/oxfw-pcm.c           |  4 --
 sound/firewire/tascam/amdtp-tascam.c     | 62 +---------------------------
 sound/firewire/tascam/tascam-pcm.c       |  7 +---
 sound/firewire/tascam/tascam.h           |  1 -
 12 files changed, 9 insertions(+), 227 deletions(-)

(limited to 'sound/firewire')

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index bebddc60fde8..23ccddb20de1 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -38,10 +38,6 @@ struct amdtp_am824 {
 	u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM];
 	u8 midi_position;
 
-	void (*transfer_samples)(struct amdtp_stream *s,
-				 struct snd_pcm_substream *pcm,
-				 __be32 *buffer, unsigned int frames);
-
 	unsigned int frame_multiplier;
 };
 
@@ -177,32 +173,6 @@ static void write_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void write_pcm_s16(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
-{
-	struct amdtp_am824 *p = s->protocol;
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	const u16 *src;
-
-	channels = p->pcm_channels;
-	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			buffer[p->pcm_positions[c]] =
-					cpu_to_be32((*src << 8) | 0x42000000);
-			src++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			src = (void *)runtime->dma_area;
-	}
-}
-
 static void read_pcm_s32(struct amdtp_stream *s,
 			 struct snd_pcm_substream *pcm,
 			 __be32 *buffer, unsigned int frames)
@@ -241,43 +211,6 @@ static void write_pcm_silence(struct amdtp_stream *s,
 	}
 }
 
-/**
- * amdtp_am824_set_pcm_format - set the PCM format
- * @s: the AMDTP stream to configure
- * @format: the format of the ALSA PCM device
- *
- * The sample format must be set after the other parameters (rate/PCM channels/
- * MIDI) and before the stream is started, and must not be changed while the
- * stream is running.
- */
-void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
-{
-	struct amdtp_am824 *p = s->protocol;
-
-	if (WARN_ON(amdtp_stream_pcm_running(s)))
-		return;
-
-	switch (format) {
-	default:
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S16:
-		if (s->direction == AMDTP_OUT_STREAM) {
-			p->transfer_samples = write_pcm_s16;
-			break;
-		}
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S32:
-		if (s->direction == AMDTP_OUT_STREAM)
-			p->transfer_samples = write_pcm_s32;
-		else
-			p->transfer_samples = read_pcm_s32;
-		break;
-	}
-}
-EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format);
-
 /**
  * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream
  * @s:		the AMDTP stream for AM824 data block, must be initialized.
@@ -407,7 +340,7 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffe
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		write_pcm_s32(s, pcm, buffer, data_blocks);
 		pcm_frames = data_blocks * p->frame_multiplier;
 	} else {
 		write_pcm_silence(s, buffer, data_blocks);
@@ -428,7 +361,7 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffe
 	unsigned int pcm_frames;
 
 	if (pcm) {
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		read_pcm_s32(s, pcm, buffer, data_blocks);
 		pcm_frames = data_blocks * p->frame_multiplier;
 	} else {
 		pcm_frames = 0;
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h
index 73b07b3109db..b56e61fc997d 100644
--- a/sound/firewire/amdtp-am824.h
+++ b/sound/firewire/amdtp-am824.h
@@ -8,8 +8,7 @@
 
 #define AM824_IN_PCM_FORMAT_BITS	SNDRV_PCM_FMTBIT_S32
 
-#define AM824_OUT_PCM_FORMAT_BITS	(SNDRV_PCM_FMTBIT_S16 | \
-					 SNDRV_PCM_FMTBIT_S32)
+#define AM824_OUT_PCM_FORMAT_BITS	SNDRV_PCM_FMTBIT_S32
 
 /*
  * This module supports maximum 64 PCM channels for one PCM stream
@@ -41,9 +40,6 @@ void amdtp_am824_set_midi_position(struct amdtp_stream *s,
 int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
 				       struct snd_pcm_runtime *runtime);
 
-void amdtp_am824_set_pcm_format(struct amdtp_stream *s,
-				snd_pcm_format_t format);
-
 void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
 			      struct snd_rawmidi_substream *midi);
 
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index 9e27eb8e1dd4..e2f023f3cd9f 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -224,8 +224,6 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&bebob->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params));
-
 	return 0;
 }
 static int
@@ -246,8 +244,6 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&bebob->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 6074fe1f00f7..b633805a885f 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -146,7 +146,6 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_dice *dice = substream->private_data;
-	struct amdtp_stream *stream = &dice->tx_stream[substream->pcm->device];
 	int err;
 
 	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
@@ -160,15 +159,12 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dice->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(stream, params_format(hw_params));
-
 	return 0;
 }
 static int playback_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_dice *dice = substream->private_data;
-	struct amdtp_stream *stream = &dice->rx_stream[substream->pcm->device];
 	int err;
 
 	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
@@ -182,8 +178,6 @@ static int playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dice->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index a4688545339c..1453c34ce99f 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -48,10 +48,6 @@ struct amdtp_dot {
 	struct snd_rawmidi_substream *midi[MAX_MIDI_PORTS];
 	int midi_fifo_used[MAX_MIDI_PORTS];
 	int midi_fifo_limit;
-
-	void (*transfer_samples)(struct amdtp_stream *s,
-				 struct snd_pcm_substream *pcm,
-				 __be32 *buffer, unsigned int frames);
 };
 
 /*
@@ -173,32 +169,6 @@ static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
 	}
 }
 
-static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
-{
-	struct amdtp_dot *p = s->protocol;
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	const u16 *src;
-
-	channels = p->pcm_channels;
-	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	buffer++;
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
-			dot_encode_step(&p->state, &buffer[c]);
-			src++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			src = (void *)runtime->dma_area;
-	}
-}
-
 static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
 			 __be32 *buffer, unsigned int frames)
 {
@@ -351,33 +321,6 @@ int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
 	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
 }
 
-void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
-{
-	struct amdtp_dot *p = s->protocol;
-
-	if (WARN_ON(amdtp_stream_pcm_running(s)))
-		return;
-
-	switch (format) {
-	default:
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S16:
-		if (s->direction == AMDTP_OUT_STREAM) {
-			p->transfer_samples = write_pcm_s16;
-			break;
-		}
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S32:
-		if (s->direction == AMDTP_OUT_STREAM)
-			p->transfer_samples = write_pcm_s32;
-		else
-			p->transfer_samples = read_pcm_s32;
-		break;
-	}
-}
-
 void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 			  struct snd_rawmidi_substream *midi)
 {
@@ -392,13 +335,12 @@ static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   unsigned int data_blocks,
 					   unsigned int *syt)
 {
-	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
 
 	pcm = ACCESS_ONCE(s->pcm);
 	if (pcm) {
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		read_pcm_s32(s, pcm, buffer, data_blocks);
 		pcm_frames = data_blocks;
 	} else {
 		pcm_frames = 0;
@@ -414,13 +356,12 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   unsigned int data_blocks,
 					   unsigned int *syt)
 {
-	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
 
 	pcm = ACCESS_ONCE(s->pcm);
 	if (pcm) {
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		write_pcm_s32(s, pcm, buffer, data_blocks);
 		pcm_frames = data_blocks;
 	} else {
 		write_pcm_silence(s, buffer, data_blocks);
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
index 68d1c52db051..6379f67416d7 100644
--- a/sound/firewire/digi00x/digi00x-pcm.c
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -88,8 +88,7 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x,
 		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
 		s = &dg00x->tx_stream;
 	} else {
-		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 |
-						 SNDRV_PCM_FMTBIT_S32;
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
 		s = &dg00x->rx_stream;
 	}
 
@@ -184,8 +183,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dg00x->mutex);
 	}
 
-	amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params));
-
 	return 0;
 }
 
@@ -206,8 +203,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dg00x->mutex);
 	}
 
-	amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
index 1275a50956c0..4dd1bbf2ed3c 100644
--- a/sound/firewire/digi00x/digi00x.h
+++ b/sound/firewire/digi00x/digi00x.h
@@ -121,7 +121,6 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
 void amdtp_dot_reset(struct amdtp_stream *s);
 int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
 				     struct snd_pcm_runtime *runtime);
-void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
 void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
 			  struct snd_rawmidi_substream *midi);
 
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index 9171702f7d0b..f10aec117998 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -257,8 +257,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&efw->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params));
-
 	return 0;
 }
 static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
@@ -278,8 +276,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&efw->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index f3530f89a025..d4594f7115ae 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -244,8 +244,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&oxfw->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
-
 	return 0;
 }
 static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
@@ -265,8 +263,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&oxfw->mutex);
 	}
 
-	amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index 9dd0fccd5ccc..6aff1fc1c72d 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -14,10 +14,6 @@
 
 struct amdtp_tscm {
 	unsigned int pcm_channels;
-
-	void (*transfer_samples)(struct amdtp_stream *s,
-				 struct snd_pcm_substream *pcm,
-				 __be32 *buffer, unsigned int frames);
 };
 
 int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
@@ -62,31 +58,6 @@ static void write_pcm_s32(struct amdtp_stream *s,
 	}
 }
 
-static void write_pcm_s16(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
-{
-	struct amdtp_tscm *p = s->protocol;
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	const u16 *src;
-
-	channels = p->pcm_channels;
-	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			buffer[c] = cpu_to_be32(*src << 16);
-			src++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			src = (void *)runtime->dma_area;
-	}
-}
-
 static void read_pcm_s32(struct amdtp_stream *s,
 			 struct snd_pcm_substream *pcm,
 			 __be32 *buffer, unsigned int frames)
@@ -146,44 +117,16 @@ int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
 	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
 }
 
-void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
-{
-	struct amdtp_tscm *p = s->protocol;
-
-	if (WARN_ON(amdtp_stream_pcm_running(s)))
-		return;
-
-	switch (format) {
-	default:
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S16:
-		if (s->direction == AMDTP_OUT_STREAM) {
-			p->transfer_samples = write_pcm_s16;
-			break;
-		}
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S32:
-		if (s->direction == AMDTP_OUT_STREAM)
-			p->transfer_samples = write_pcm_s32;
-		else
-			p->transfer_samples = read_pcm_s32;
-		break;
-	}
-}
-
 static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
 					   __be32 *buffer,
 					   unsigned int data_blocks,
 					   unsigned int *syt)
 {
-	struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
 	struct snd_pcm_substream *pcm;
 
 	pcm = ACCESS_ONCE(s->pcm);
 	if (data_blocks > 0 && pcm)
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		read_pcm_s32(s, pcm, buffer, data_blocks);
 
 	/* A place holder for control messages. */
 
@@ -195,7 +138,6 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 					   unsigned int data_blocks,
 					   unsigned int *syt)
 {
-	struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
 	struct snd_pcm_substream *pcm;
 
 	/* This field is not used. */
@@ -203,7 +145,7 @@ static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
 
 	pcm = ACCESS_ONCE(s->pcm);
 	if (pcm)
-		p->transfer_samples(s, pcm, buffer, data_blocks);
+		write_pcm_s32(s, pcm, buffer, data_blocks);
 	else
 		write_pcm_silence(s, buffer, data_blocks);
 
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index f5dd6ce6b6f1..6207588d7c73 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -48,8 +48,7 @@ static int pcm_init_hw_params(struct snd_tscm *tscm,
 		stream = &tscm->tx_stream;
 		pcm_channels = tscm->spec->pcm_capture_analog_channels;
 	} else {
-		runtime->hw.formats =
-				SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32;
+		runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
 		stream = &tscm->rx_stream;
 		pcm_channels = tscm->spec->pcm_playback_analog_channels;
 	}
@@ -125,8 +124,6 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&tscm->mutex);
 	}
 
-	amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params));
-
 	return 0;
 }
 
@@ -147,8 +144,6 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&tscm->mutex);
 	}
 
-	amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params));
-
 	return 0;
 }
 
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
index 08ecfae5c584..a5bd167eb5d9 100644
--- a/sound/firewire/tascam/tascam.h
+++ b/sound/firewire/tascam/tascam.h
@@ -131,7 +131,6 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
 int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate);
 int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
 				      struct snd_pcm_runtime *runtime);
-void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
 
 int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate);
 int snd_tscm_stream_get_clock(struct snd_tscm *tscm,
-- 
cgit v1.2.3-58-ga151


From 905e46acd3272d04566fec49afbd7ad9e2ed9ae3 Mon Sep 17 00:00:00 2001
From: Bhumika Goyal <bhumirks@gmail.com>
Date: Sat, 27 May 2017 20:16:15 +0530
Subject: ALSA: declare snd_kcontrol_new structures as const

Declare snd_kcontrol_new structures as const as they are only passed an
argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const.
Done using Coccinelle:

@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};

@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)

@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p

@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;

Cross compiled these files:
sound/aoa/codecs/tas.c - powerpc
sound/mips/{hal2.c/sgio2audio.c} - mips
sound/ppc/{awacs.c/beep.c/tumbler.c} - powerpc
sound/soc/sh/siu_dai.c - sh
Could not find an architecture to compile sound/sh/aica.c.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/aoa/codecs/tas.c              | 14 +++++++-------
 sound/aoa/fabrics/layout.c          |  8 ++++----
 sound/core/ctljack.c                |  2 +-
 sound/drivers/vx/vx_mixer.c         | 24 ++++++++++++------------
 sound/firewire/bebob/bebob_maudio.c |  8 ++++----
 sound/mips/hal2.c                   |  4 ++--
 sound/mips/sgio2audio.c             | 14 +++++++-------
 sound/pcmcia/vx/vxp_mixer.c         |  4 ++--
 sound/ppc/awacs.c                   |  4 ++--
 sound/ppc/beep.c                    |  2 +-
 sound/ppc/tumbler.c                 |  8 ++++----
 sound/sh/aica.c                     |  4 ++--
 sound/soc/sh/siu_dai.c              |  4 ++--
 sound/usb/mixer_quirks.c            |  2 +-
 14 files changed, 51 insertions(+), 51 deletions(-)

(limited to 'sound/firewire')

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index 78ed1ffbf786..733b6365dad6 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -271,7 +271,7 @@ static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new volume_control = {
+static const struct snd_kcontrol_new volume_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Master Playback Volume",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -314,7 +314,7 @@ static int tas_snd_mute_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new mute_control = {
+static const struct snd_kcontrol_new mute_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Master Playback Switch",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -426,7 +426,7 @@ static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new drc_range_control = {
+static const struct snd_kcontrol_new drc_range_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "DRC Range",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -466,7 +466,7 @@ static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new drc_switch_control = {
+static const struct snd_kcontrol_new drc_switch_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "DRC Range Switch",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -524,7 +524,7 @@ static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new capture_source_control = {
+static const struct snd_kcontrol_new capture_source_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	/* If we name this 'Input Source', it properly shows up in
 	 * alsamixer as a selection, * but it's shown under the
@@ -586,7 +586,7 @@ static int tas_snd_treble_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new treble_control = {
+static const struct snd_kcontrol_new treble_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Treble",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -637,7 +637,7 @@ static int tas_snd_bass_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new bass_control = {
+static const struct snd_kcontrol_new bass_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Bass",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index a0c4a5de809c..1eddf8fa188f 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -707,7 +707,7 @@ static int detect_choice_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new headphone_detect_choice = {
+static const struct snd_kcontrol_new headphone_detect_choice = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Headphone Detect Autoswitch",
 	.info = control_info,
@@ -717,7 +717,7 @@ static struct snd_kcontrol_new headphone_detect_choice = {
 	.private_value = 0,
 };
 
-static struct snd_kcontrol_new lineout_detect_choice = {
+static const struct snd_kcontrol_new lineout_detect_choice = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Line-Out Detect Autoswitch",
 	.info = control_info,
@@ -749,7 +749,7 @@ static int detected_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new headphone_detected = {
+static const struct snd_kcontrol_new headphone_detected = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Headphone Detected",
 	.info = control_info,
@@ -758,7 +758,7 @@ static struct snd_kcontrol_new headphone_detected = {
 	.private_value = 0,
 };
 
-static struct snd_kcontrol_new lineout_detected = {
+static const struct snd_kcontrol_new lineout_detected = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Line-Out Detected",
 	.info = control_info,
diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c
index 84a3cd683068..0249d5e6ac23 100644
--- a/sound/core/ctljack.c
+++ b/sound/core/ctljack.c
@@ -23,7 +23,7 @@ static int jack_detect_kctl_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new jack_detect_kctl = {
+static const struct snd_kcontrol_new jack_detect_kctl = {
 	/* name is filled later */
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.access = SNDRV_CTL_ELEM_ACCESS_READ,
diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c
index be9477e30739..98a41ac40b60 100644
--- a/sound/drivers/vx/vx_mixer.c
+++ b/sound/drivers/vx/vx_mixer.c
@@ -455,7 +455,7 @@ static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_output_level = {
+static const struct snd_kcontrol_new vx_control_output_level = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -514,7 +514,7 @@ static int vx_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_audio_src = {
+static const struct snd_kcontrol_new vx_control_audio_src = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Capture Source",
 	.info =		vx_audio_src_info,
@@ -558,7 +558,7 @@ static int vx_clock_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_clock_mode = {
+static const struct snd_kcontrol_new vx_control_clock_mode = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Clock Mode",
 	.info =		vx_clock_mode_info,
@@ -717,7 +717,7 @@ static int vx_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 
 static const DECLARE_TLV_DB_SCALE(db_scale_audio_gain, -10975, 25, 0);
 
-static struct snd_kcontrol_new vx_control_audio_gain = {
+static const struct snd_kcontrol_new vx_control_audio_gain = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -727,14 +727,14 @@ static struct snd_kcontrol_new vx_control_audio_gain = {
 	.put =          vx_audio_gain_put,
 	.tlv = { .p = db_scale_audio_gain },
 };
-static struct snd_kcontrol_new vx_control_output_switch = {
+static const struct snd_kcontrol_new vx_control_output_switch = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "PCM Playback Switch",
 	.info =         vx_audio_sw_info,
 	.get =          vx_audio_sw_get,
 	.put =          vx_audio_sw_put
 };
-static struct snd_kcontrol_new vx_control_monitor_gain = {
+static const struct snd_kcontrol_new vx_control_monitor_gain = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Monitoring Volume",
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -744,7 +744,7 @@ static struct snd_kcontrol_new vx_control_monitor_gain = {
 	.put =          vx_audio_monitor_put,
 	.tlv = { .p = db_scale_audio_gain },
 };
-static struct snd_kcontrol_new vx_control_monitor_switch = {
+static const struct snd_kcontrol_new vx_control_monitor_switch = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Monitoring Switch",
 	.info =         vx_audio_sw_info,	/* shared */
@@ -805,7 +805,7 @@ static int vx_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_iec958_mask = {
+static const struct snd_kcontrol_new vx_control_iec958_mask = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =		SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
@@ -813,7 +813,7 @@ static struct snd_kcontrol_new vx_control_iec958_mask = {
 	.get =		vx_iec958_mask_get,
 };
 
-static struct snd_kcontrol_new vx_control_iec958 = {
+static const struct snd_kcontrol_new vx_control_iec958 = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
 	.info =         vx_iec958_info,
@@ -878,7 +878,7 @@ static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_vu_meter = {
+static const struct snd_kcontrol_new vx_control_vu_meter = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	/* name will be filled later */
@@ -886,7 +886,7 @@ static struct snd_kcontrol_new vx_control_vu_meter = {
 	.get =		vx_vu_meter_get,
 };
 
-static struct snd_kcontrol_new vx_control_peak_meter = {
+static const struct snd_kcontrol_new vx_control_peak_meter = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	/* name will be filled later */
@@ -894,7 +894,7 @@ static struct snd_kcontrol_new vx_control_peak_meter = {
 	.get =		vx_peak_meter_get,
 };
 
-static struct snd_kcontrol_new vx_control_saturation = {
+static const struct snd_kcontrol_new vx_control_saturation = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Input Saturation",
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 07e5abdbceb5..d10208f92edf 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -396,7 +396,7 @@ static int special_clk_ctl_put(struct snd_kcontrol *kctl,
 
 	return err;
 }
-static struct snd_kcontrol_new special_clk_ctl = {
+static const struct snd_kcontrol_new special_clk_ctl = {
 	.name	= "Clock Source",
 	.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -429,7 +429,7 @@ static int special_sync_ctl_get(struct snd_kcontrol *kctl,
 
 	return 0;
 }
-static struct snd_kcontrol_new special_sync_ctl = {
+static const struct snd_kcontrol_new special_sync_ctl = {
 	.name	= "Sync Status",
 	.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access	= SNDRV_CTL_ELEM_ACCESS_READ,
@@ -521,7 +521,7 @@ end:
 	mutex_unlock(&bebob->mutex);
 	return err;
 }
-static struct snd_kcontrol_new special_dig_in_iface_ctl = {
+static const struct snd_kcontrol_new special_dig_in_iface_ctl = {
 	.name	= "Digital Input Interface",
 	.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -577,7 +577,7 @@ static int special_dig_out_iface_ctl_set(struct snd_kcontrol *kctl,
 	mutex_unlock(&bebob->mutex);
 	return err;
 }
-static struct snd_kcontrol_new special_dig_out_iface_ctl = {
+static const struct snd_kcontrol_new special_dig_out_iface_ctl = {
 	.name	= "Digital Output Interface",
 	.iface	= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access	= SNDRV_CTL_ELEM_ACCESS_READWRITE,
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index 684dc4ddef41..6505deb8c06b 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -264,7 +264,7 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol,
 	return old != new;
 }
 
-static struct snd_kcontrol_new hal2_ctrl_headphone = {
+static const struct snd_kcontrol_new hal2_ctrl_headphone = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Headphone Playback Volume",
 	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -274,7 +274,7 @@ static struct snd_kcontrol_new hal2_ctrl_headphone = {
 	.put            = hal2_gain_put,
 };
 
-static struct snd_kcontrol_new hal2_ctrl_mic = {
+static const struct snd_kcontrol_new hal2_ctrl_mic = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Mic Capture Volume",
 	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index f07aa3993f83..0ebc1c3727df 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -230,7 +230,7 @@ static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
 }
 
 /* dac1/pcm0 mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "PCM Playback Volume",
 	.index          = 0,
@@ -242,7 +242,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = {
 };
 
 /* dac2/pcm1 mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "PCM Playback Volume",
 	.index          = 1,
@@ -254,7 +254,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = {
 };
 
 /* record level mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Capture Volume",
 	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -265,7 +265,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_reclevel = {
 };
 
 /* record level source control */
-static struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Capture Source",
 	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -275,7 +275,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_recsource = {
 };
 
 /* line mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_line = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_line = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Line Playback Volume",
 	.index          = 0,
@@ -287,7 +287,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_line = {
 };
 
 /* cd mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_cd = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_cd = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Line Playback Volume",
 	.index          = 1,
@@ -299,7 +299,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_cd = {
 };
 
 /* mic mixer control */
-static struct snd_kcontrol_new sgio2audio_ctrl_mic = {
+static const struct snd_kcontrol_new sgio2audio_ctrl_mic = {
 	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name           = "Mic Playback Volume",
 	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c
index 1ca3eef9e305..304b153005a5 100644
--- a/sound/pcmcia/vx/vxp_mixer.c
+++ b/sound/pcmcia/vx/vxp_mixer.c
@@ -69,7 +69,7 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 
 static const DECLARE_TLV_DB_SCALE(db_scale_mic, -21, 3, 0);
 
-static struct snd_kcontrol_new vx_control_mic_level = {
+static const struct snd_kcontrol_new vx_control_mic_level = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -109,7 +109,7 @@ static int vx_mic_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_mic_boost = {
+static const struct snd_kcontrol_new vx_control_mic_boost = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Mic Boost",
 	.info =		vx_mic_boost_info,
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 1468e4b7bf93..d1e4ef1c5c30 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -514,7 +514,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] = {
 	},
 };
 
-static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = {
+static const struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Headphone Playback Switch",
 	.info = snd_pmac_boolean_stereo_info,
@@ -523,7 +523,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = {
 	.private_value = AMP_CH_HD,
 };
 
-static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw = {
+static const struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Speaker Playback Switch",
 	.info = snd_pmac_boolean_stereo_info,
diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c
index d3524f9fa05d..f19eb3e39937 100644
--- a/sound/ppc/beep.c
+++ b/sound/ppc/beep.c
@@ -206,7 +206,7 @@ static int snd_pmac_put_beep(struct snd_kcontrol *kcontrol,
 	return oval != chip->beep->volume;
 }
 
-static struct snd_kcontrol_new snd_pmac_beep_mixer = {
+static const struct snd_kcontrol_new snd_pmac_beep_mixer = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Beep Playback Volume",
 	.info = snd_pmac_info_beep,
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index 58ee8089bbf9..0779a2912237 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -897,7 +897,7 @@ static struct snd_kcontrol_new snapper_mixers[] = {
 	},
 };
 
-static struct snd_kcontrol_new tumbler_hp_sw = {
+static const struct snd_kcontrol_new tumbler_hp_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Headphone Playback Switch",
 	.info = snd_pmac_boolean_mono_info,
@@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw = {
 	.put = tumbler_put_mute_switch,
 	.private_value = TUMBLER_MUTE_HP,
 };
-static struct snd_kcontrol_new tumbler_speaker_sw = {
+static const struct snd_kcontrol_new tumbler_speaker_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Speaker Playback Switch",
 	.info = snd_pmac_boolean_mono_info,
@@ -913,7 +913,7 @@ static struct snd_kcontrol_new tumbler_speaker_sw = {
 	.put = tumbler_put_mute_switch,
 	.private_value = TUMBLER_MUTE_AMP,
 };
-static struct snd_kcontrol_new tumbler_lineout_sw = {
+static const struct snd_kcontrol_new tumbler_lineout_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Line Out Playback Switch",
 	.info = snd_pmac_boolean_mono_info,
@@ -921,7 +921,7 @@ static struct snd_kcontrol_new tumbler_lineout_sw = {
 	.put = tumbler_put_mute_switch,
 	.private_value = TUMBLER_MUTE_LINE,
 };
-static struct snd_kcontrol_new tumbler_drc_sw = {
+static const struct snd_kcontrol_new tumbler_drc_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "DRC Switch",
 	.info = snd_pmac_boolean_mono_info,
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index fbbc25279559..ab4802df62e1 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -535,7 +535,7 @@ static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new snd_aica_pcmswitch_control = {
+static const struct snd_kcontrol_new snd_aica_pcmswitch_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "PCM Playback Switch",
 	.index = 0,
@@ -544,7 +544,7 @@ static struct snd_kcontrol_new snd_aica_pcmswitch_control = {
 	.put = aica_pcmswitch_put
 };
 
-static struct snd_kcontrol_new snd_aica_pcmvolume_control = {
+static const struct snd_kcontrol_new snd_aica_pcmvolume_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "PCM Playback Volume",
 	.index = 0,
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index 76b2ab8c2b4a..4a22aadac294 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -441,7 +441,7 @@ static int siu_dai_put_volume(struct snd_kcontrol *kctrl,
 	return 0;
 }
 
-static struct snd_kcontrol_new playback_controls = {
+static const struct snd_kcontrol_new playback_controls = {
 	.iface		= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name		= "PCM Playback Volume",
 	.index		= 0,
@@ -451,7 +451,7 @@ static struct snd_kcontrol_new playback_controls = {
 	.private_value	= VOLUME_PLAYBACK,
 };
 
-static struct snd_kcontrol_new capture_controls = {
+static const struct snd_kcontrol_new capture_controls = {
 	.iface		= SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name		= "PCM Capture Volume",
 	.index		= 0,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 4fa0053a40af..e3d1dec48ee4 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -362,7 +362,7 @@ static int snd_audigy2nx_led_resume(struct usb_mixer_elem_list *list)
 }
 
 /* name and private_value are set dynamically */
-static struct snd_kcontrol_new snd_audigy2nx_control = {
+static const struct snd_kcontrol_new snd_audigy2nx_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = snd_audigy2nx_led_info,
 	.get = snd_audigy2nx_led_get,
-- 
cgit v1.2.3-58-ga151


From 875becf8412c60ffae93c5f69e95a4d023f0e8ee Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Wed, 7 Jun 2017 09:38:05 +0900
Subject: ALSA: firewire: process packets in 'struct snd_pcm_ops.ack' callback

In recent commit for ALSA PCM core, some arrangement is done for
'struct snd_pcm_ops.ack' callback. This is called when appl_ptr is
explicitly moved in intermediate buffer for PCM frames, except for
some cases described later.

For drivers in ALSA firewire stack, usage of this callback has a merit to
reduce latency between time of PCM frame queueing and handling actual
packets in recent isochronous cycle, because no need to wait for software
IRQ context from isochronous context of OHCI 1394.

If this works well in a case that mapped page frame is used for the
intermediate buffer, user process should execute some commands for ioctl(2)
to tell the number of handled PCM frames in the intermediate buffer just
after handling them. Therefore, at present, with a combination of below
conditions, this doesn't work as expected and user process should wait for
the software IRQ context as usual:
 - when ALSA PCM core judges page frame mapping is available for status
   data (struct snd_pcm_mmap_status) and control data
   (struct snd_pcm_mmap_control).
 - user process handles PCM frames by loop just with 'snd_pcm_mmap_begin()'
   and 'snd_pcm_mmap_commit()'.
 - user process uses PCM hw plugin in alsa-lib to operate I/O without
   'sync_ptr_ioctl' option.

Unfortunately, major use case include these three conditions.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c            | 19 +++++++++++++++++++
 sound/firewire/amdtp-stream.h            |  1 +
 sound/firewire/bebob/bebob_pcm.c         | 16 ++++++++++++++++
 sound/firewire/dice/dice-pcm.c           | 18 ++++++++++++++++++
 sound/firewire/digi00x/digi00x-pcm.c     | 16 ++++++++++++++++
 sound/firewire/fireface/ff-pcm.c         | 16 ++++++++++++++++
 sound/firewire/fireworks/fireworks_pcm.c | 16 ++++++++++++++++
 sound/firewire/motu/motu-pcm.c           | 16 ++++++++++++++++
 sound/firewire/oxfw/oxfw-pcm.c           | 16 ++++++++++++++++
 sound/firewire/tascam/tascam-pcm.c       | 16 ++++++++++++++++
 10 files changed, 150 insertions(+)

(limited to 'sound/firewire')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 9e6f54f8c45d..4316d9db404d 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -928,6 +928,25 @@ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s)
 }
 EXPORT_SYMBOL(amdtp_stream_pcm_pointer);
 
+/**
+ * amdtp_stream_pcm_ack - acknowledge queued PCM frames
+ * @s: the AMDTP stream that transfers the PCM frames
+ *
+ * Returns zero always.
+ */
+int amdtp_stream_pcm_ack(struct amdtp_stream *s)
+{
+	/*
+	 * Process isochronous packets for recent isochronous cycle to handle
+	 * queued PCM frames.
+	 */
+	if (amdtp_stream_running(s))
+		fw_iso_context_flush_completions(s->context);
+
+	return 0;
+}
+EXPORT_SYMBOL(amdtp_stream_pcm_ack);
+
 /**
  * amdtp_stream_update - update the stream after a bus reset
  * @s: the AMDTP stream
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 7e8831722821..6d613f2eb612 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -168,6 +168,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 
 void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
 unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
+int amdtp_stream_pcm_ack(struct amdtp_stream *s);
 void amdtp_stream_pcm_abort(struct amdtp_stream *s);
 
 extern const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT];
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index e2f023f3cd9f..657e15a27e5c 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -355,6 +355,20 @@ pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
 	return amdtp_stream_pcm_pointer(&bebob->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_bebob *bebob = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&bebob->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_bebob *bebob = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&bebob->rx_stream);
+}
+
 int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -366,6 +380,7 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
+		.ack		= pcm_capture_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 	};
 	static const struct snd_pcm_ops playback_ops = {
@@ -377,6 +392,7 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,
+		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index b633805a885f..2dda74695069 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -294,6 +294,22 @@ static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream)
 	return amdtp_stream_pcm_pointer(stream);
 }
 
+static int capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_dice *dice = substream->private_data;
+	struct amdtp_stream *stream = &dice->tx_stream[substream->pcm->device];
+
+	return amdtp_stream_pcm_ack(stream);
+}
+
+static int playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_dice *dice = substream->private_data;
+	struct amdtp_stream *stream = &dice->rx_stream[substream->pcm->device];
+
+	return amdtp_stream_pcm_ack(stream);
+}
+
 int snd_dice_create_pcm(struct snd_dice *dice)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -305,6 +321,7 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.prepare   = capture_prepare,
 		.trigger   = capture_trigger,
 		.pointer   = capture_pointer,
+		.ack       = capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
@@ -317,6 +334,7 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.prepare   = playback_prepare,
 		.trigger   = playback_trigger,
 		.pointer   = playback_pointer,
+		.ack       = playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
index 6379f67416d7..f76cf5e383db 100644
--- a/sound/firewire/digi00x/digi00x-pcm.c
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -324,6 +324,20 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
 	return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&dg00x->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&dg00x->rx_stream);
+}
+
 int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -335,6 +349,7 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
+		.ack		= pcm_capture_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 	};
 	static const struct snd_pcm_ops playback_ops = {
@@ -346,6 +361,7 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,
+		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index 93cee1978e8e..adb5c87f492f 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -365,6 +365,20 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
 	return amdtp_stream_pcm_pointer(&ff->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_ff *ff = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&ff->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_ff *ff = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&ff->rx_stream);
+}
+
 static struct snd_pcm_ops pcm_capture_ops = {
 	.open		= pcm_open,
 	.close		= pcm_close,
@@ -374,6 +388,7 @@ static struct snd_pcm_ops pcm_capture_ops = {
 	.prepare	= pcm_capture_prepare,
 	.trigger	= pcm_capture_trigger,
 	.pointer	= pcm_capture_pointer,
+	.ack		= pcm_capture_ack,
 	.page		= snd_pcm_lib_get_vmalloc_page,
 };
 
@@ -386,6 +401,7 @@ static struct snd_pcm_ops pcm_playback_ops = {
 	.prepare	= pcm_playback_prepare,
 	.trigger	= pcm_playback_trigger,
 	.pointer	= pcm_playback_pointer,
+	.ack		= pcm_playback_ack,
 	.page		= snd_pcm_lib_get_vmalloc_page,
 	.mmap		= snd_pcm_lib_mmap_vmalloc,
 };
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index f10aec117998..346e2647ed1f 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -379,6 +379,20 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
 	return amdtp_stream_pcm_pointer(&efw->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_efw *efw = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&efw->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_efw *efw = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&efw->rx_stream);
+}
+
 int snd_efw_create_pcm_devices(struct snd_efw *efw)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -390,6 +404,7 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw)
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
+		.ack		= pcm_capture_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 	};
 	static const struct snd_pcm_ops playback_ops = {
@@ -401,6 +416,7 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw)
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,
+		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c
index 94558f3d218b..e3ef89cee565 100644
--- a/sound/firewire/motu/motu-pcm.c
+++ b/sound/firewire/motu/motu-pcm.c
@@ -356,6 +356,20 @@ static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream)
 	return amdtp_stream_pcm_pointer(&motu->rx_stream);
 }
 
+static int capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_motu *motu = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&motu->tx_stream);
+}
+
+static int playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_motu *motu = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&motu->rx_stream);
+}
+
 int snd_motu_create_pcm_devices(struct snd_motu *motu)
 {
 	static struct snd_pcm_ops capture_ops = {
@@ -367,6 +381,7 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu)
 		.prepare   = capture_prepare,
 		.trigger   = capture_trigger,
 		.pointer   = capture_pointer,
+		.ack       = capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
@@ -379,6 +394,7 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu)
 		.prepare   = playback_prepare,
 		.trigger   = playback_trigger,
 		.pointer   = playback_pointer,
+		.ack       = playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index d4594f7115ae..bc1a3a36ab06 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -382,6 +382,20 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstm)
 	return amdtp_stream_pcm_pointer(&oxfw->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_oxfw *oxfw = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&oxfw->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_oxfw *oxfw = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&oxfw->rx_stream);
+}
+
 int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -393,6 +407,7 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 		.prepare   = pcm_capture_prepare,
 		.trigger   = pcm_capture_trigger,
 		.pointer   = pcm_capture_pointer,
+		.ack       = pcm_capture_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
@@ -405,6 +420,7 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 		.prepare   = pcm_playback_prepare,
 		.trigger   = pcm_playback_trigger,
 		.pointer   = pcm_playback_pointer,
+		.ack       = pcm_playback_ack,
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index 6207588d7c73..3c4482aa7231 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -263,6 +263,20 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
 	return amdtp_stream_pcm_pointer(&tscm->rx_stream);
 }
 
+static int pcm_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&tscm->tx_stream);
+}
+
+static int pcm_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	return amdtp_stream_pcm_ack(&tscm->rx_stream);
+}
+
 int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
 {
 	static const struct snd_pcm_ops capture_ops = {
@@ -274,6 +288,7 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
+		.ack		= pcm_capture_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 	};
 	static const struct snd_pcm_ops playback_ops = {
@@ -285,6 +300,7 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,
+		.ack		= pcm_playback_ack,
 		.page		= snd_pcm_lib_get_vmalloc_page,
 		.mmap		= snd_pcm_lib_mmap_vmalloc,
 	};
-- 
cgit v1.2.3-58-ga151


From d2dc2a96de9d58038ac1f9ee432e88a70958d59b Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Wed, 7 Jun 2017 09:38:06 +0900
Subject: ALSA: fireface: constify ALSA specific operations

ALSA fireface driver has ALSA specific operations for MIDI/PCM data.
Structured data for the operations can be constified. Additionally,
The structured data can be function local.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/fireface/ff-midi.c | 22 ++++++++---------
 sound/firewire/fireface/ff-pcm.c  | 52 +++++++++++++++++++--------------------
 2 files changed, 35 insertions(+), 39 deletions(-)

(limited to 'sound/firewire')

diff --git a/sound/firewire/fireface/ff-midi.c b/sound/firewire/fireface/ff-midi.c
index 29ee0a7365c3..949ee56b4e0e 100644
--- a/sound/firewire/fireface/ff-midi.c
+++ b/sound/firewire/fireface/ff-midi.c
@@ -74,18 +74,6 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substream,
 	spin_unlock_irqrestore(&ff->lock, flags);
 }
 
-static struct snd_rawmidi_ops midi_capture_ops = {
-	.open		= midi_capture_open,
-	.close		= midi_capture_close,
-	.trigger	= midi_capture_trigger,
-};
-
-static struct snd_rawmidi_ops midi_playback_ops = {
-	.open		= midi_playback_open,
-	.close		= midi_playback_close,
-	.trigger	= midi_playback_trigger,
-};
-
 static void set_midi_substream_names(struct snd_rawmidi_str *stream,
 				     const char *const name)
 {
@@ -99,6 +87,16 @@ static void set_midi_substream_names(struct snd_rawmidi_str *stream,
 
 int snd_ff_create_midi_devices(struct snd_ff *ff)
 {
+	static const struct snd_rawmidi_ops midi_capture_ops = {
+		.open		= midi_capture_open,
+		.close		= midi_capture_close,
+		.trigger	= midi_capture_trigger,
+	};
+	static const struct snd_rawmidi_ops midi_playback_ops = {
+		.open		= midi_playback_open,
+		.close		= midi_playback_close,
+		.trigger	= midi_playback_trigger,
+	};
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_str *stream;
 	int err;
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index adb5c87f492f..ad974b5a2561 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -379,35 +379,33 @@ static int pcm_playback_ack(struct snd_pcm_substream *substream)
 	return amdtp_stream_pcm_ack(&ff->rx_stream);
 }
 
-static struct snd_pcm_ops pcm_capture_ops = {
-	.open		= pcm_open,
-	.close		= pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= pcm_capture_hw_params,
-	.hw_free	= pcm_capture_hw_free,
-	.prepare	= pcm_capture_prepare,
-	.trigger	= pcm_capture_trigger,
-	.pointer	= pcm_capture_pointer,
-	.ack		= pcm_capture_ack,
-	.page		= snd_pcm_lib_get_vmalloc_page,
-};
-
-static struct snd_pcm_ops pcm_playback_ops = {
-	.open		= pcm_open,
-	.close		= pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= pcm_playback_hw_params,
-	.hw_free	= pcm_playback_hw_free,
-	.prepare	= pcm_playback_prepare,
-	.trigger	= pcm_playback_trigger,
-	.pointer	= pcm_playback_pointer,
-	.ack		= pcm_playback_ack,
-	.page		= snd_pcm_lib_get_vmalloc_page,
-	.mmap		= snd_pcm_lib_mmap_vmalloc,
-};
-
 int snd_ff_create_pcm_devices(struct snd_ff *ff)
 {
+	static const struct snd_pcm_ops pcm_capture_ops = {
+		.open		= pcm_open,
+		.close		= pcm_close,
+		.ioctl		= snd_pcm_lib_ioctl,
+		.hw_params	= pcm_capture_hw_params,
+		.hw_free	= pcm_capture_hw_free,
+		.prepare	= pcm_capture_prepare,
+		.trigger	= pcm_capture_trigger,
+		.pointer	= pcm_capture_pointer,
+		.ack		= pcm_capture_ack,
+		.page		= snd_pcm_lib_get_vmalloc_page,
+	};
+	static const struct snd_pcm_ops pcm_playback_ops = {
+		.open		= pcm_open,
+		.close		= pcm_close,
+		.ioctl		= snd_pcm_lib_ioctl,
+		.hw_params	= pcm_playback_hw_params,
+		.hw_free	= pcm_playback_hw_free,
+		.prepare	= pcm_playback_prepare,
+		.trigger	= pcm_playback_trigger,
+		.pointer	= pcm_playback_pointer,
+		.ack		= pcm_playback_ack,
+		.page		= snd_pcm_lib_get_vmalloc_page,
+		.mmap		= snd_pcm_lib_mmap_vmalloc,
+	};
 	struct snd_pcm *pcm;
 	int err;
 
-- 
cgit v1.2.3-58-ga151


From 55799c5ab99e8762e58c387e39f9fa36e8f14158 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 8 Jun 2017 09:11:03 +0900
Subject: ALSA: firewire: arrange common PCM info/constraints for AMDTP engine
 applications

In ALSA firewire stack, 8 drivers uses IEC 61883-1/6 engine for data
transmission. They have common PCM info/constraints and duplicated codes.

This commit unifies the codes into fireiwre-lib.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/amdtp-stream.c            | 19 +++++++++++++++++
 sound/firewire/bebob/bebob_pcm.c         | 21 -------------------
 sound/firewire/dice/dice-pcm.c           | 20 ------------------
 sound/firewire/digi00x/digi00x-pcm.c     | 33 ++++++++++--------------------
 sound/firewire/fireface/ff-pcm.c         | 21 -------------------
 sound/firewire/fireworks/fireworks_pcm.c | 21 -------------------
 sound/firewire/motu/motu-pcm.c           | 20 ------------------
 sound/firewire/oxfw/oxfw-pcm.c           | 20 ------------------
 sound/firewire/tascam/tascam-pcm.c       | 35 ++++++--------------------------
 9 files changed, 36 insertions(+), 174 deletions(-)

(limited to 'sound/firewire')

diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 4316d9db404d..9678bc75dc5b 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -148,8 +148,27 @@ EXPORT_SYMBOL(amdtp_rate_table);
 int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 					struct snd_pcm_runtime *runtime)
 {
+	struct snd_pcm_hardware *hw = &runtime->hw;
 	int err;
 
+	hw->info = SNDRV_PCM_INFO_BATCH |
+		   SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		   SNDRV_PCM_INFO_INTERLEAVED |
+		   SNDRV_PCM_INFO_JOINT_DUPLEX |
+		   SNDRV_PCM_INFO_MMAP |
+		   SNDRV_PCM_INFO_MMAP_VALID;
+
+	/* SNDRV_PCM_INFO_BATCH */
+	hw->periods_min = 2;
+	hw->periods_max = UINT_MAX;
+
+	/* bytes for a frame */
+	hw->period_bytes_min = 4 * hw->channels_max;
+
+	/* Just to prevent from allocating much pages. */
+	hw->period_bytes_max = hw->period_bytes_min * 2048;
+	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
+
 	/*
 	 * Currently firewire-lib processes 16 packets in one software
 	 * interrupt callback. This equals to 2msec but actually the
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index 657e15a27e5c..e6adab3ef42e 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -92,19 +92,6 @@ limit_channels_and_rates(struct snd_pcm_hardware *hw,
 	}
 }
 
-static void
-limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;		/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;	/* bytes for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int
 pcm_init_hw_params(struct snd_bebob *bebob,
 		   struct snd_pcm_substream *substream)
@@ -114,13 +101,6 @@ pcm_init_hw_params(struct snd_bebob *bebob,
 	struct snd_bebob_stream_formation *formations;
 	int err;
 
-	runtime->hw.info = SNDRV_PCM_INFO_BATCH |
-			   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			   SNDRV_PCM_INFO_INTERLEAVED |
-			   SNDRV_PCM_INFO_JOINT_DUPLEX |
-			   SNDRV_PCM_INFO_MMAP |
-			   SNDRV_PCM_INFO_MMAP_VALID;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		s = &bebob->tx_stream;
@@ -132,7 +112,6 @@ pcm_init_hw_params(struct snd_bebob *bebob,
 	}
 
 	limit_channels_and_rates(&runtime->hw, formations);
-	limit_period_and_buffer(&runtime->hw);
 
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				  hw_rule_channels, formations,
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 2dda74695069..7cb9e9713ac3 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -51,18 +51,6 @@ static int limit_channels_and_rates(struct snd_dice *dice,
 	return 0;
 }
 
-static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;			/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;    /* byte for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int init_hw_info(struct snd_dice *dice,
 			struct snd_pcm_substream *substream)
 {
@@ -74,13 +62,6 @@ static int init_hw_info(struct snd_dice *dice,
 	unsigned int count, size;
 	int err;
 
-	hw->info = SNDRV_PCM_INFO_MMAP |
-		   SNDRV_PCM_INFO_MMAP_VALID |
-		   SNDRV_PCM_INFO_BATCH |
-		   SNDRV_PCM_INFO_INTERLEAVED |
-		   SNDRV_PCM_INFO_JOINT_DUPLEX |
-		   SNDRV_PCM_INFO_BLOCK_TRANSFER;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		hw->formats = AM824_IN_PCM_FORMAT_BITS;
 		dir = AMDTP_IN_STREAM;
@@ -107,7 +88,6 @@ static int init_hw_info(struct snd_dice *dice,
 				       substream->pcm->device, size);
 	if (err < 0)
 		return err;
-	limit_period_and_buffer(hw);
 
 	return amdtp_am824_add_pcm_hw_constraints(stream, runtime);
 }
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
index f76cf5e383db..796f4b4645f5 100644
--- a/sound/firewire/digi00x/digi00x-pcm.c
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -58,31 +58,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params,
 static int pcm_init_hw_params(struct snd_dg00x *dg00x,
 			      struct snd_pcm_substream *substream)
 {
-	static const struct snd_pcm_hardware hardware = {
-		.info = SNDRV_PCM_INFO_BATCH |
-			SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			SNDRV_PCM_INFO_INTERLEAVED |
-			SNDRV_PCM_INFO_JOINT_DUPLEX |
-			SNDRV_PCM_INFO_MMAP |
-			SNDRV_PCM_INFO_MMAP_VALID,
-		.rates = SNDRV_PCM_RATE_44100 |
-			 SNDRV_PCM_RATE_48000 |
-			 SNDRV_PCM_RATE_88200 |
-			 SNDRV_PCM_RATE_96000,
-		.rate_min = 44100,
-		.rate_max = 96000,
-		.channels_min = 10,
-		.channels_max = 18,
-		.period_bytes_min = 4 * 18,
-		.period_bytes_max = 4 * 18 * 2048,
-		.buffer_bytes_max = 4 * 18 * 2048 * 2,
-		.periods_min = 2,
-		.periods_max = UINT_MAX,
-	};
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_pcm_hardware *hw = &runtime->hw;
 	struct amdtp_stream *s;
 	int err;
 
-	substream->runtime->hw = hardware;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
@@ -92,6 +72,15 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x,
 		s = &dg00x->rx_stream;
 	}
 
+	hw->channels_min = 10;
+	hw->channels_max = 18;
+
+	hw->rates = SNDRV_PCM_RATE_44100 |
+		    SNDRV_PCM_RATE_48000 |
+		    SNDRV_PCM_RATE_88200 |
+		    SNDRV_PCM_RATE_96000;
+	snd_pcm_limit_hw_rates(runtime);
+
 	err = snd_pcm_hw_rule_add(substream->runtime, 0,
 				  SNDRV_PCM_HW_PARAM_CHANNELS,
 				  hw_rule_channels, NULL,
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index ad974b5a2561..d12a0e3a4219 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -91,18 +91,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw,
 	}
 }
 
-static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;		/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;	/* bytes for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int pcm_init_hw_params(struct snd_ff *ff,
 			      struct snd_pcm_substream *substream)
 {
@@ -111,13 +99,6 @@ static int pcm_init_hw_params(struct snd_ff *ff,
 	const unsigned int *pcm_channels;
 	int err;
 
-	runtime->hw.info = SNDRV_PCM_INFO_BATCH |
-			   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			   SNDRV_PCM_INFO_INTERLEAVED |
-			   SNDRV_PCM_INFO_JOINT_DUPLEX |
-			   SNDRV_PCM_INFO_MMAP |
-			   SNDRV_PCM_INFO_MMAP_VALID;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
 		s = &ff->tx_stream;
@@ -128,9 +109,7 @@ static int pcm_init_hw_params(struct snd_ff *ff,
 		pcm_channels = ff->spec->pcm_playback_channels;
 	}
 
-	/* limit rates */
 	limit_channels_and_rates(&runtime->hw, pcm_channels);
-	limit_period_and_buffer(&runtime->hw);
 
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				  hw_rule_channels, (void *)pcm_channels,
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index 346e2647ed1f..40faed5e6968 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -129,19 +129,6 @@ limit_channels(struct snd_pcm_hardware *hw, unsigned int *pcm_channels)
 	}
 }
 
-static void
-limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;		/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;	/* bytes for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int
 pcm_init_hw_params(struct snd_efw *efw,
 		   struct snd_pcm_substream *substream)
@@ -151,13 +138,6 @@ pcm_init_hw_params(struct snd_efw *efw,
 	unsigned int *pcm_channels;
 	int err;
 
-	runtime->hw.info = SNDRV_PCM_INFO_BATCH |
-			   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			   SNDRV_PCM_INFO_INTERLEAVED |
-			   SNDRV_PCM_INFO_JOINT_DUPLEX |
-			   SNDRV_PCM_INFO_MMAP |
-			   SNDRV_PCM_INFO_MMAP_VALID;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		s = &efw->tx_stream;
@@ -173,7 +153,6 @@ pcm_init_hw_params(struct snd_efw *efw,
 	snd_pcm_limit_hw_rates(runtime);
 
 	limit_channels(&runtime->hw, pcm_channels);
-	limit_period_and_buffer(&runtime->hw);
 
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				  hw_rule_channels, pcm_channels,
diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c
index e3ef89cee565..a2b50df70874 100644
--- a/sound/firewire/motu/motu-pcm.c
+++ b/sound/firewire/motu/motu-pcm.c
@@ -96,18 +96,6 @@ static void limit_channels_and_rates(struct snd_motu *motu,
 	snd_pcm_limit_hw_rates(runtime);
 }
 
-static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;			/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;    /* byte for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int init_hw_info(struct snd_motu *motu,
 			struct snd_pcm_substream *substream)
 {
@@ -117,13 +105,6 @@ static int init_hw_info(struct snd_motu *motu,
 	struct snd_motu_packet_format *formats;
 	int err;
 
-	hw->info = SNDRV_PCM_INFO_MMAP |
-		   SNDRV_PCM_INFO_MMAP_VALID |
-		   SNDRV_PCM_INFO_BATCH |
-		   SNDRV_PCM_INFO_INTERLEAVED |
-		   SNDRV_PCM_INFO_JOINT_DUPLEX |
-		   SNDRV_PCM_INFO_BLOCK_TRANSFER;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		hw->formats = SNDRV_PCM_FMTBIT_S32;
 		stream = &motu->tx_stream;
@@ -135,7 +116,6 @@ static int init_hw_info(struct snd_motu *motu,
 	}
 
 	limit_channels_and_rates(motu, runtime, formats);
-	limit_period_and_buffer(hw);
 
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				  motu_rate_constraint, formats,
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index bc1a3a36ab06..3dd46285c0e2 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -106,18 +106,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats)
 	}
 }
 
-static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
-{
-	hw->periods_min = 2;		/* SNDRV_PCM_INFO_BATCH */
-	hw->periods_max = UINT_MAX;
-
-	hw->period_bytes_min = 4 * hw->channels_max;	/* bytes for a frame */
-
-	/* Just to prevent from allocating much pages. */
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
-}
-
 static int init_hw_params(struct snd_oxfw *oxfw,
 			  struct snd_pcm_substream *substream)
 {
@@ -126,13 +114,6 @@ static int init_hw_params(struct snd_oxfw *oxfw,
 	struct amdtp_stream *stream;
 	int err;
 
-	runtime->hw.info = SNDRV_PCM_INFO_BATCH |
-			   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			   SNDRV_PCM_INFO_INTERLEAVED |
-			   SNDRV_PCM_INFO_JOINT_DUPLEX |
-			   SNDRV_PCM_INFO_MMAP |
-			   SNDRV_PCM_INFO_MMAP_VALID;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		stream = &oxfw->tx_stream;
@@ -144,7 +125,6 @@ static int init_hw_params(struct snd_oxfw *oxfw,
 	}
 
 	limit_channels_and_rates(&runtime->hw, formats);
-	limit_period_and_buffer(&runtime->hw);
 
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				  hw_rule_channels, formats,
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index 3c4482aa7231..6ec8ec634d4d 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -8,41 +8,14 @@
 
 #include "tascam.h"
 
-static void set_buffer_params(struct snd_pcm_hardware *hw)
-{
-	hw->period_bytes_min = 4 * hw->channels_min;
-	hw->period_bytes_max = hw->period_bytes_min * 2048;
-	hw->buffer_bytes_max = hw->period_bytes_max * 2;
-
-	hw->periods_min = 2;
-	hw->periods_max = UINT_MAX;
-}
-
 static int pcm_init_hw_params(struct snd_tscm *tscm,
 			      struct snd_pcm_substream *substream)
 {
-	static const struct snd_pcm_hardware hardware = {
-		.info = SNDRV_PCM_INFO_BATCH |
-			SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			SNDRV_PCM_INFO_INTERLEAVED |
-			SNDRV_PCM_INFO_JOINT_DUPLEX |
-			SNDRV_PCM_INFO_MMAP |
-			SNDRV_PCM_INFO_MMAP_VALID,
-		.rates = SNDRV_PCM_RATE_44100 |
-			 SNDRV_PCM_RATE_48000 |
-			 SNDRV_PCM_RATE_88200 |
-			 SNDRV_PCM_RATE_96000,
-		.rate_min = 44100,
-		.rate_max = 96000,
-		.channels_min = 10,
-		.channels_max = 18,
-	};
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_pcm_hardware *hw = &runtime->hw;
 	struct amdtp_stream *stream;
 	unsigned int pcm_channels;
 
-	runtime->hw = hardware;
-
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
 		stream = &tscm->tx_stream;
@@ -59,7 +32,11 @@ static int pcm_init_hw_params(struct snd_tscm *tscm,
 		pcm_channels += 2;
 	runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels;
 
-	set_buffer_params(&runtime->hw);
+	hw->rates = SNDRV_PCM_RATE_44100 |
+		    SNDRV_PCM_RATE_48000 |
+		    SNDRV_PCM_RATE_88200 |
+		    SNDRV_PCM_RATE_96000;
+	snd_pcm_limit_hw_rates(runtime);
 
 	return amdtp_tscm_add_pcm_hw_constraints(stream, runtime);
 }
-- 
cgit v1.2.3-58-ga151