Merge series "simple-audio-card codec2codec support" from Samuel Holland <samuel@sholland.org>:

We are currently using simple-audio-card on the Allwinner A64 SoC.
The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the
CPU, the modem, and Bluetooth. Adding support for the secondary AIFs
requires adding codec2codec DAI links.

Since the modem and bt-sco codec DAI drivers only have one set of
possible PCM parameters (namely, 8kHz mono S16LE), there's no real
need for a machine driver to specify the DAI link configuration. The
parameters for these "simple" DAI links can be chosen automatically.

This series adds codec2codec DAI link support to simple-audio-card.
Codec to codec links are automatically detected when all DAIs in the
link belong to codec components.

I tried to reuse as much code as possible, so the first two patches
refactor a couple of helper functions to be more generic.

The last patch adds the new feature and its documentation.

Changes in v4:
  - Rebased on top of asoc/for-next, several changes to patch 2
  - Removed unused variable from patch 3
Changes in v3:
  - Update use of for_each_rtd_components for v5.6
Changes in v2:
  - Drop patch 1 as it was merged
  - Automatically detect codec2codec links instead of using a DT property

Samuel Holland (3):
  ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates
  ASoC: pcm: Export parameter intersection logic
  ASoC: simple-card: Add support for codec2codec DAI links

 Documentation/sound/soc/codec-to-codec.rst |  9 +++-
 include/sound/pcm.h                        |  9 +++-
 include/sound/soc.h                        |  3 ++
 sound/core/pcm_misc.c                      | 18 +++----
 sound/soc/generic/simple-card-utils.c      | 48 ++++++++++++++++++
 sound/soc/soc-pcm.c                        | 59 ++++++++++++++--------
 6 files changed, 114 insertions(+), 32 deletions(-)

--
2.24.1

2 files changed, 11 insertions, 1 deletions

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2628246b76fa..f7a95b711100 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
 	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
 }
 
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
+
+static inline int
+snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+	return snd_pcm_hw_limit_rates(&runtime->hw);
+}
+
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
 unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 81e5d17be935..9543d9246ca4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
 
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream);
+
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt);