Merge branch 'for-next' into for-linus

Author: tiwai

Documentation/sound/kernel-api/writing-an-alsa-driver.rst

@@ -3565,13 +3565,17 @@ given size.
 The second argument (type) and the third argument (device pointer) are
 dependent on the bus. For normal devices, pass the device pointer
 (typically identical as ``card->dev``) to the third argument with
-``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the
+``SNDRV_DMA_TYPE_DEV`` type.
+
+For the continuous buffer unrelated to the
 bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type.
 You can pass NULL to the device pointer in that case, which is the
 default mode implying to allocate with ``GFP_KERNEL`` flag.
-If you need a different GFP flag, you can pass it by encoding the flag
-into the device pointer via a special macro
-:c:func:`snd_dma_continuous_data()`.
+If you need a restricted (lower) address, set up the coherent DMA mask
+bits for the device, and pass the device pointer, like the normal
+device memory allocations.  For this type, it's still allowed to pass
+NULL to the device pointer, too, if no address restriction is needed.
+
 For the scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the
 device pointer (see the `Non-Contiguous Buffers`_ section).
 
@@ -3811,15 +3815,6 @@ arguments here.  Since each vmalloc call should succeed at any time,
 we don't need to pre-allocate the buffers like other continuous
 pages.
 
-If you need the 32bit DMA allocation, pass the device pointer encoded
-by :c:func:`snd_dma_continuous_data()` with ``GFP_KERNEL|__GFP_DMA32``
-argument.
-
-::
-
-  snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC,
-          snd_dma_continuous_data(GFP_KERNEL | __GFP_DMA32), 0, 0);
-
 Proc Interface
 ==============
 

drivers/usb/gadget/function/u_uac1_legacy.c

@@ -158,8 +158,8 @@ size_t u_audio_playback(struct gaudio *card, void *buf, size_t count)
 	snd_pcm_sframes_t frames;
 
 try_again:
-	if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
-		runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+	if (runtime->state == SNDRV_PCM_STATE_XRUN ||
+		runtime->state == SNDRV_PCM_STATE_SUSPENDED) {
 		result = snd_pcm_kernel_ioctl(substream,
 				SNDRV_PCM_IOCTL_PREPARE, NULL);
 		if (result < 0) {

include/sound/hda_codec.h

@@ -258,7 +258,6 @@ struct hda_codec {
 	unsigned int link_down_at_suspend:1; /* link down at runtime suspend */
 	unsigned int relaxed_resume:1;	/* don't resume forcibly for jack */
 	unsigned int forced_resume:1; /* forced resume for jack */
-	unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */
 
 #ifdef CONFIG_PM
 	unsigned long power_on_acct;
@@ -293,8 +292,6 @@ struct hda_codec {
 #define dev_to_hda_codec(_dev)	container_of(_dev, struct hda_codec, core.dev)
 #define hda_codec_dev(_dev)	(&(_dev)->core.dev)
 
-#define hdac_to_hda_priv(_hdac) \
-			container_of(_hdac, struct hdac_hda_priv, codec.core)
 #define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
 
 #define list_for_each_codec(c, bus) \

include/sound/hda_register.h

@@ -260,23 +260,25 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 
 #define AZX_REG_ML_LCAP			0x00
 #define AZX_REG_ML_LCTL			0x04
+
+#define AZX_ML_LCTL_CPA			BIT(23)
+#define AZX_ML_LCTL_CPA_SHIFT		23
+#define AZX_ML_LCTL_SPA			BIT(16)
+#define AZX_ML_LCTL_SPA_SHIFT		16
+#define AZX_ML_LCTL_SCF			GENMASK(3, 0)
+
 #define AZX_REG_ML_LOSIDV		0x08
+
+/* bit0 is reserved, with BIT(1) mapping to stream1 */
+#define AZX_ML_LOSIDV_STREAM_MASK	0xFFFE
+
 #define AZX_REG_ML_LSDIID		0x0C
 #define AZX_REG_ML_LPSOO		0x10
 #define AZX_REG_ML_LPSIO		0x12
 #define AZX_REG_ML_LWALFC		0x18
 #define AZX_REG_ML_LOUTPAY		0x20
 #define AZX_REG_ML_LINPAY		0x30
 
-/* bit0 is reserved, with BIT(1) mapping to stream1 */
-#define ML_LOSIDV_STREAM_MASK		0xFFFE
-
-#define ML_LCTL_SCF_MASK			0xF
-#define AZX_MLCTL_SPA				(0x1 << 16)
-#define AZX_MLCTL_CPA				(0x1 << 23)
-#define AZX_MLCTL_SPA_SHIFT			16
-#define AZX_MLCTL_CPA_SHIFT			23
-
 /* registers for DMA Resume Capability Structure */
 #define AZX_DRSM_CAP_ID			0x5
 #define AZX_REG_DRSM_CTL		0x4

include/sound/hdaudio.h

@@ -10,6 +10,7 @@
 #include <linux/interrupt.h>
 #include <linux/io.h>
 #include <linux/io-64-nonatomic-lo-hi.h>
+#include <linux/iopoll.h>
 #include <linux/pm_runtime.h>
 #include <linux/timecounter.h>
 #include <sound/core.h>
@@ -550,6 +551,7 @@ void snd_hdac_stream_init(struct hdac_bus *bus, struct hdac_stream *azx_dev,
 			  int idx, int direction, int tag);
 struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus,
 					   struct snd_pcm_substream *substream);
+void snd_hdac_stream_release_locked(struct hdac_stream *azx_dev);
 void snd_hdac_stream_release(struct hdac_stream *azx_dev);
 struct hdac_stream *snd_hdac_get_stream(struct hdac_bus *bus,
 					int dir, int stream_tag);
@@ -560,8 +562,8 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev);
 int snd_hdac_stream_set_params(struct hdac_stream *azx_dev,
 				unsigned int format_val);
 void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start);
-void snd_hdac_stream_clear(struct hdac_stream *azx_dev);
 void snd_hdac_stream_stop(struct hdac_stream *azx_dev);
+void snd_hdac_stop_streams(struct hdac_bus *bus);
 void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus);
 void snd_hdac_stream_reset(struct hdac_stream *azx_dev);
 void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set,
@@ -589,6 +591,12 @@ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
 	snd_hdac_reg_readw((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg)
 #define snd_hdac_stream_readb(dev, reg) \
 	snd_hdac_reg_readb((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg)
+#define snd_hdac_stream_readb_poll(dev, reg, val, cond, delay_us, timeout_us) \
+	readb_poll_timeout((dev)->sd_addr + AZX_REG_ ## reg, val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_stream_readl_poll(dev, reg, val, cond, delay_us, timeout_us) \
+	readl_poll_timeout((dev)->sd_addr + AZX_REG_ ## reg, val, cond, \
+			   delay_us, timeout_us)
 
 /* update a register, pass without AZX_REG_ prefix */
 #define snd_hdac_stream_updatel(dev, reg, mask, val) \

include/sound/hdaudio_ext.h

@@ -11,9 +11,6 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 		      const struct hdac_ext_bus_ops *ext_ops);
 
 void snd_hdac_ext_bus_exit(struct hdac_bus *bus);
-int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr,
-				struct hdac_device *hdev, int type);
-void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev);
 void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus);
 
 #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \
@@ -80,12 +77,9 @@ struct hdac_ext_stream {
 #define stream_to_hdac_ext_stream(s) \
 	container_of(s, struct hdac_ext_stream, hstream)
 
-void snd_hdac_ext_stream_init(struct hdac_bus *bus,
-			      struct hdac_ext_stream *hext_stream, int idx,
-			      int direction, int tag);
 int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
 				 int num_stream, int dir);
-void snd_hdac_stream_free_all(struct hdac_bus *bus);
+void snd_hdac_ext_stream_free_all(struct hdac_bus *bus);
 void snd_hdac_link_free_all(struct hdac_bus *bus);
 struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus,
 					   struct snd_pcm_substream *substream,
@@ -188,12 +182,6 @@ void snd_hdac_ext_bus_link_power(struct hdac_device *codec, bool enable);
 #define snd_hdac_adsp_readq_poll(chip, reg, val, cond, delay_us, timeout_us) \
 	readq_poll_timeout((chip)->dsp_ba + (reg), val, cond, \
 			   delay_us, timeout_us)
-#define snd_hdac_stream_readb_poll(strm, reg, val, cond, delay_us, timeout_us) \
-	readb_poll_timeout((strm)->sd_addr + AZX_REG_ ## reg, val, cond, \
-			   delay_us, timeout_us)
-#define snd_hdac_stream_readl_poll(strm, reg, val, cond, delay_us, timeout_us) \
-	readl_poll_timeout((strm)->sd_addr + AZX_REG_ ## reg, val, cond, \
-			   delay_us, timeout_us)
 
 struct hdac_ext_device;
 

include/sound/memalloc.h

@@ -26,9 +26,6 @@ struct snd_dma_device {
 	struct device *dev;		/* generic device */
 };
 
-#define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
-
-
 /*
  * buffer types
  */

include/sound/pcm.h

@@ -346,6 +346,8 @@ static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy
 
 struct snd_pcm_runtime {
 	/* -- Status -- */
+	snd_pcm_state_t state;		/* stream state */
+	snd_pcm_state_t suspended_state; /* suspended stream state */
 	struct snd_pcm_substream *trigger_master;
 	struct timespec64 trigger_tstamp;	/* trigger timestamp */
 	bool trigger_tstamp_latched;     /* trigger timestamp latched in low-level driver/hardware */
@@ -678,11 +680,25 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
  */
 static inline int snd_pcm_running(struct snd_pcm_substream *substream)
 {
-	return (substream->runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
-		(substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING &&
+	return (substream->runtime->state == SNDRV_PCM_STATE_RUNNING ||
+		(substream->runtime->state == SNDRV_PCM_STATE_DRAINING &&
 		 substream->stream == SNDRV_PCM_STREAM_PLAYBACK));
 }
 
+/**
+ * __snd_pcm_set_state - Change the current PCM state
+ * @runtime: PCM runtime to set
+ * @state: the current state to set
+ *
+ * Call within the stream lock
+ */
+static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime,
+				       snd_pcm_state_t state)
+{
+	runtime->state = state;
+	runtime->status->state = state; /* copy for mmap */
+}
+
 /**
  * bytes_to_samples - Unit conversion of the size from bytes to samples
  * @runtime: PCM runtime instance

include/uapi/sound/asequencer.h

@@ -3,22 +3,6 @@
  *  Main header file for the ALSA sequencer
  *  Copyright (c) 1998-1999 by Frank van de Pol <[email protected]>
  *            (c) 1998-1999 by Jaroslav Kysela <[email protected]>
- *
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
  */
 #ifndef _UAPI__SOUND_ASEQUENCER_H
 #define _UAPI__SOUND_ASEQUENCER_H

include/uapi/sound/asoc.h

@@ -5,10 +5,6 @@
  * Copyright (C) 2012 Texas Instruments Inc.
  * Copyright (C) 2015 Intel Corporation.
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
  * Simple file API to load FW that includes mixers, coefficients, DAPM graphs,
  * algorithms, equalisers, DAIs, widgets etc.
 */