google_rtc_audio_processing: Major rework

Lots of work on AEC, with an eye to getting it working on main, where
it had bitrotten, and pulling in various features (in particular IPC3
pipeline state management and DP scheduling) that had merged in other
branches.

Dynamically configure stream formats (sample format/rate and channel
count) from the connected streams at prepare() time instead of relying
on build-time tuning.

Port the code to use the source/sink API, in as sophisticated a manner
as I can find.  Copies unroll cleanly into just a few instructions per
sample, including integer/float conversions and de-/interleaving.

Support both 16 and 32 bit sample formats, with a fairly clever
inlining scheme to share as much code as possible between them.  The
component will select "copy" function pointers at prepare() time.

Support an is_ready_to_process() method to enable AEC's use as a
async component in a DP scheduler.

The large buffers required by this component (input and output staging
and an internally-managed pool/heap block) are now static symbols
instead of dynamic memory from the heap.  These are very large, taking
up about half of what is available to the linker on MTL.  Relying on
heap allocation is just dangerous in this context.

This fully decouples AEC from the playback stream.  It will run
without an active reference happily, feeding zeros to the processing,
and pick up in stride when the pipeline starts.  This requires adding
a trigger handler for pipeline control in IPC3, which will propagate
certain triggers across pipeline boundaries when shutting down
playback streams, breaking active capture.

Fixes a few bugs and misfeatures also:

+ Chunk the copies by full buffer strides between AEC processing calls
  instead of testing at each copied frame.

+ Copy the reference and mic streams in tandem, preventing them from
  becoming out of sync if the devices weren't themselves synchronized.

+ Copy the AEC results to the output stream after the call to
  ProcessCapture() instead of before.  This was a hidden latency bug
  in the original code, I think.

+ Cleans up the Kconfig to remove stale variables and guard all
  the component-specific tunables under the top-level component
  variable.  Also uses a default instead of a select to couple to
  CONFIG_STUBS, allowing AEC to be manually tuned.

Signed-off-by: Andy Ross <[email protected]>

app/stub_build_all_ipc3.conf

@@ -1,5 +1,6 @@
 CONFIG_COMP_STUBS=y
 CONFIG_COMP_GOOGLE_RTC_AUDIO_PROCESSING=y
+CONFIG_GOOGLE_RTC_AUDIO_PROCESSING_MOCK=y
 CONFIG_COMP_TONE=n
 CONFIG_COMP_CROSSOVER=y
 CONFIG_COMP_DRC=y

src/arch/host/configs/library_defconfig

@@ -4,6 +4,7 @@ CONFIG_COMP_DCBLOCK=y
 CONFIG_COMP_DRC=y
 CONFIG_COMP_FIR=y
 CONFIG_COMP_GOOGLE_RTC_AUDIO_PROCESSING=y
+CONFIG_GOOGLE_RTC_AUDIO_PROCESSING_MOCK=y
 CONFIG_COMP_IIR=y
 CONFIG_COMP_MFCC=y
 CONFIG_COMP_MODULE_ADAPTER=y

src/audio/google/Kconfig

@@ -15,7 +15,6 @@ config COMP_GOOGLE_HOTWORD_DETECT
 config COMP_GOOGLE_RTC_AUDIO_PROCESSING
 	bool "Google Real Time Communication Audio processing"
 	select COMP_BLOB
-	select GOOGLE_RTC_AUDIO_PROCESSING_MOCK if COMP_STUBS
 	default n
 	help
 	  Select for Google real-time communication audio processing. It
@@ -24,6 +23,8 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING
 	  This component takes raw microphones input and playback reference
 	  and outputs an echo-free microphone signal.
 
+if COMP_GOOGLE_RTC_AUDIO_PROCESSING
+
 config COMP_GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ
 	depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
 	int "Sample rate for Google Real Time Communication Audio processing"
@@ -32,22 +33,15 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ
 	  Sets the sample rate for the memory buffer for the Google real-time
 	  communication audio processing.
 
-config COMP_GOOGLE_RTC_AUDIO_PROCESSING_NUM_CHANNELS
-	depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
-	int "Number of channels to process for Google Real Time Communication Audio processing"
-	default 2 if MT8195
-	default 1
-	help
-	  Sets the number of input/mic channels to process in the
-	  Google real-time communication audio processing.
-
-config COMP_GOOGLE_RTC_AUDIO_PROCESSING_NUM_AEC_REFERENCE_CHANNELS
-	depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
-	int "Number of AEC reference channels for Google Real Time Communication Audio processing"
+config COMP_GOOGLE_RTC_AUDIO_PROCESSING_CHANNEL_MAX
+	int "Max number of AEC channels"
 	default 2
 	help
-	  Sets the number AEC reference channels in the Google real-time
-	  communication audio processing.
+	  Sets the maximum number source/sink channels Google Real
+	  Time Communication Audio Processing will use for.  This is a
+	  computation and memory budget tunable.  Channel counts are
+	  retrieved at runtime, but channels higher than this number
+	  are ignored (on input) or cleared (output).
 
 config COMP_GOOGLE_RTC_AUDIO_PROCESSING_MEMORY_BUFFER_SIZE_BYTES
 	depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
@@ -75,10 +69,12 @@ config COMP_GOOGLE_RTC_AUDIO_PROCESSING_MIC_HEADROOM_LINEAR
 
 config GOOGLE_RTC_AUDIO_PROCESSING_MOCK
 	bool "Google Real Time Communication Audio processing mock"
-	default n
+	default y if COMP_STUBS
 	depends on COMP_GOOGLE_RTC_AUDIO_PROCESSING
 	help
 	  Mock Google real-time communication audio processing.
 	  It allows for compilation check and basic audio flow checking.
 
+endif # COMP_GOOGLE_RTC_AUDIO_PROCESSING
+
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