ASoC: SOF: Intel: power optimizations with HDaudio SPIB register

Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

	The use of the SPIB register helps reduce power consumption - though
	to a smaller degree than DMI_L1. This hardware capability is however
	incompatible with userspace-initiated rewinds typically used by
	PulseAudio.

	In the past (2015..2017) Intel suggested an API extension to let
	applications disable rewinds. At the time the feedback was that such a
	capability was too Intel-specific and SPIB remained unused except for
	loading DSP code. We now see devices with smaller batteries being
	released, and it's time to revisit Linux support for SPIB to extend
	battery life.

	In this update the rewinds are disabled via an opt-in kernel
	parameter. In the previous reviews, there was consensus that a Kconfig
	option was too complicated for distributions to set, and we are
	missing a TBD API to expose such capabilities to user-space.

	The debate on whether or not to use rewinds, and the impact of
	disabling rewinds, will likely be closed when Intel releases the
	'deep-buffer' support, currently under development [2][3]. With this
	solution, rewinds will not be needed, ever. When an application deals
	with content that is not latency-sensitive (e.g. music playback), it
	will be able to reduce power consumption by selecting a different PCM
	device with increased buffering capabilities.  Low-latency streams
	will be handled by the 'regular' path. In other words, the impossible
	compromise between power and latency will be handled with different
	PCM devices/profiles for the same endpoint, and we can push the design
	of capability negotiation to a later time when all the building blocks
	(firmware topology, kernel, userspace) are ready - we still have
	firmware xruns, DPCM race conditions to solve, and a need to describe
	these alternate PCM devices with UCM using 'modifiers'.

12 files changed, 86 insertions, 3 deletions

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 5fbb79e30819..ff7e638221c5 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -300,7 +300,7 @@ typedef int __bitwise snd_pcm_subformat_t;
 #define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME    0x04000000  /* report estimated link audio time */
 #define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000  /* report synchronized audio/system time */
 #define SNDRV_PCM_INFO_EXPLICIT_SYNC	0x10000000	/* needs explicit sync of pointers and data */
-
+#define SNDRV_PCM_INFO_NO_REWINDS	0x20000000	/* hardware can only support monotonic changes of appl_ptr */
 #define SNDRV_PCM_INFO_DRAIN_TRIGGER	0x40000000		/* internal kernel flag - trigger in drain */
 #define SNDRV_PCM_INFO_FIFO_IN_FRAMES	0x80000000	/* internal kernel flag - FIFO size is in frames */
 
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4f4b4739f987..f2090025236b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2128,11 +2128,28 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t old_appl_ptr = runtime->control->appl_ptr;
+	snd_pcm_sframes_t diff;
 	int ret;
 
 	if (old_appl_ptr == appl_ptr)
 		return 0;
 
+	if (appl_ptr >= runtime->boundary)
+		return -EINVAL;
+	/*
+	 * check if a rewind is requested by the application
+	 */
+	if (substream->runtime->info & SNDRV_PCM_INFO_NO_REWINDS) {
+		diff = appl_ptr - old_appl_ptr;
+		if (diff >= 0) {
+			if (diff > runtime->buffer_size)
+				return -EINVAL;
+		} else {
+			if (runtime->boundary + diff > runtime->buffer_size)
+				return -EINVAL;
+		}
+	}
+
 	runtime->control->appl_ptr = appl_ptr;
 	if (substream->ops->ack) {
 		ret = substream->ops->ack(substream);
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index 1baf0fddeb3d..8778f46f1d37 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -78,6 +78,7 @@ const struct snd_sof_dsp_ops sof_apl_ops = {
 	.pcm_hw_free	= hda_dsp_stream_hw_free,
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
+	.pcm_ack	= hda_dsp_pcm_ack,
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
 	/* probe callbacks */
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index d455272bfc8e..04daaa6100f1 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -283,6 +283,7 @@ const struct snd_sof_dsp_ops sof_cnl_ops = {
 	.pcm_hw_free	= hda_dsp_stream_hw_free,
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
+	.pcm_ack	= hda_dsp_pcm_ack,
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
 	/* probe callbacks */
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index cc8ddef37f37..974383cd0440 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -32,6 +32,10 @@ static bool hda_always_enable_dmi_l1;
 module_param_named(always_enable_dmi_l1, hda_always_enable_dmi_l1, bool, 0444);
 MODULE_PARM_DESC(always_enable_dmi_l1, "SOF HDA always enable DMI l1");
 
+static bool hda_disable_rewinds = IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_DISABLE_REWINDS);
+module_param_named(disable_rewinds, hda_disable_rewinds, bool, 0444);
+MODULE_PARM_DESC(disable_rewinds, "SOF HDA disable rewinds");
+
 u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate)
 {
 	switch (rate) {
@@ -120,8 +124,11 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev,
 		return ret;
 	}
 
-	/* disable SPIB, to enable buffer wrap for stream */
-	hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0);
+	/* enable SPIB when rewinds are disabled */
+	if (hda_disable_rewinds)
+		hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_ENABLE, 0);
+	else
+		hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0);
 
 	/* update no_stream_position flag for ipc params */
 	if (hda && hda->no_ipc_position) {
@@ -140,6 +147,29 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev,
 	return 0;
 }
 
+/* update SPIB register with appl position */
+int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream)
+{
+	struct hdac_stream *hstream = substream->runtime->private_data;
+	struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	ssize_t appl_pos, buf_size;
+	u32 spib;
+
+	appl_pos = frames_to_bytes(runtime, runtime->control->appl_ptr);
+	buf_size = frames_to_bytes(runtime, runtime->buffer_size);
+
+	spib = appl_pos % buf_size;
+
+	/* Allowable value for SPIB is 1 byte to max buffer size */
+	if (!spib)
+		spib = buf_size;
+
+	sof_io_write(sdev, hext_stream->spib_addr, spib);
+
+	return 0;
+}
+
 int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev,
 			struct snd_pcm_substream *substream, int cmd)
 {
@@ -235,6 +265,13 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
 	}
 
 	/*
+	 * if we want the .ack to work, we need to prevent the control from being mapped.
+	 * The status can still be mapped.
+	 */
+	if (hda_disable_rewinds)
+		runtime->hw.info |= SNDRV_PCM_INFO_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR;
+
+	/*
 	 * All playback streams are DMI L1 capable, capture streams need
 	 * pause push/release to be disabled
 	 */
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 1d845c2cbc33..b6f037815344 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -655,6 +655,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev,
 					SOF_HDA_REG_PP_PPCTL, mask, 0);
 	spin_unlock_irq(&bus->reg_lock);
 
+	hda_dsp_stream_spib_config(sdev, link_dev, HDA_DSP_SPIB_DISABLE, 0);
+
 	stream->substream = NULL;
 
 	return 0;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 646f5d4dc882..8ed4031ca007 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -534,6 +534,7 @@ int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev,
 			struct snd_pcm_substream *substream, int cmd);
 snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev,
 				      struct snd_pcm_substream *substream);
+int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream);
 
 /*
  * DSP Stream Operations.
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index 6c5422157ec8..343c1af7c453 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -77,6 +77,7 @@ const struct snd_sof_dsp_ops sof_icl_ops = {
 	.pcm_hw_free	= hda_dsp_stream_hw_free,
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
+	.pcm_ack	= hda_dsp_pcm_ack,
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
 	/* probe callbacks */
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 237e92e790b7..7f7929c5cb88 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -113,6 +113,7 @@ const struct snd_sof_dsp_ops sof_tgl_ops = {
 	.pcm_hw_free	= hda_dsp_stream_hw_free,
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
+	.pcm_ack	= hda_dsp_pcm_ack,
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
 	/* probe callbacks */
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index a0648a13e3eb..0226a53148c9 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -487,6 +487,16 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev,
 	return 0;
 }
 
+/* pcm ack */
+static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev,
+					   struct snd_pcm_substream *substream)
+{
+	if (sof_ops(sdev) && sof_ops(sdev)->pcm_ack)
+		return sof_ops(sdev)->pcm_ack(sdev, substream);
+
+	return 0;
+}
+
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
 static inline int
 snd_sof_probe_compr_assign(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 0ceb1a9cbf73..1bf7e60be772 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -916,6 +916,14 @@ static void sof_pcm_remove(struct snd_soc_component *component)
 	snd_soc_tplg_component_remove(component);
 }
 
+static int sof_pcm_ack(struct snd_soc_component *component,
+		       struct snd_pcm_substream *substream)
+{
+	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
+
+	return snd_sof_pcm_platform_ack(sdev, substream);
+}
+
 void snd_sof_new_platform_drv(struct snd_sof_dev *sdev)
 {
 	struct snd_soc_component_driver *pd = &sdev->plat_drv;
@@ -934,6 +942,7 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev)
 	pd->hw_free = sof_pcm_hw_free;
 	pd->trigger = sof_pcm_trigger;
 	pd->pointer = sof_pcm_pointer;
+	pd->ack = sof_pcm_ack;
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
 	pd->compress_ops = &sof_probe_compressed_ops;
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index a9c5197617f1..16caf5c74035 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -207,6 +207,9 @@ struct snd_sof_dsp_ops {
 	snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev,
 					 struct snd_pcm_substream *substream); /* optional */
 
+	/* pcm ack */
+	int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */
+
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
 	/* Except for probe_pointer, all probe ops are mandatory */
 	int (*probe_assign)(struct snd_sof_dev *sdev,