audio: webrtc: add 32/48 kHz AGC support, AEC DP scheduling, and Kconfig overlays

app/stub_build_all_ipc4.conf

@@ -2,6 +2,12 @@ CONFIG_COMP_STUBS=y
 CONFIG_COMP_ARIA=y
 CONFIG_COMP_CHAIN_DMA=y
 CONFIG_COMP_UP_DOWN_MIXER=y
+CONFIG_COMP_WEBRTC_VAD=m
+CONFIG_COMP_WEBRTC_NS=m
+CONFIG_COMP_WEBRTC_AEC=m
+CONFIG_COMP_WEBRTC_NS2=m
+CONFIG_COMP_WEBRTC_AGC=m
+CONFIG_COMP_WEBRTC_HPF=m
 CONFIG_IPC_MAJOR_4=y
 CONFIG_SAMPLES=y
 CONFIG_SAMPLE_SMART_AMP=y

app/webrtc_overlay.conf

@@ -0,0 +1,44 @@
+# SPDX-License-Identifier: BSD-3-Clause
+#
+# WebRTC Audio Processing Modules Kconfig overlay (LLEXT real backends)
+
+# WebRTC NS (classic spectral Wiener filter)
+CONFIG_COMP_WEBRTC_NS=m
+CONFIG_COMP_WEBRTC_NS_STUB=n
+CONFIG_WEBRTC_NS_LEVEL=2
+CONFIG_WEBRTC_NS_SAMPLE_RATE_HZ=16000
+
+# WebRTC VAD (libfvad Voice Activity Detector)
+CONFIG_COMP_WEBRTC_VAD=m
+CONFIG_COMP_WEBRTC_VAD_STUB=n
+CONFIG_WEBRTC_VAD_MODE=2
+CONFIG_WEBRTC_VAD_FRAME_MS=10
+
+# WebRTC AEC (Acoustic Echo Canceller)
+CONFIG_COMP_WEBRTC_AEC=m
+CONFIG_WEBRTC_AEC_BACKEND_AECM=y
+CONFIG_WEBRTC_AEC_FILTER_LEN_MS=64
+CONFIG_WEBRTC_AEC_SUPPRESSION_LEVEL=1
+CONFIG_WEBRTC_AEC_SAMPLE_RATE_HZ=16000
+
+# WebRTC NS2 (RNNoise neural network noise suppression)
+CONFIG_COMP_WEBRTC_NS2=m
+CONFIG_COMP_WEBRTC_NS2_STUB=n
+CONFIG_WEBRTC_NS2_CHANNELS_MAX=2
+CONFIG_WEBRTC_NS2_VAD_THRESHOLD_PCT=60
+CONFIG_WEBRTC_NS2_VAD_NOTIFY=y
+
+# WebRTC HPF (High-Pass Filter 80 Hz biquad)
+CONFIG_COMP_WEBRTC_HPF=m
+CONFIG_COMP_WEBRTC_HPF_STUB=n
+
+# WebRTC AGC (Automatic Gain Control & Limiter)
+CONFIG_COMP_WEBRTC_AGC=m
+CONFIG_COMP_WEBRTC_AGC_STUB=n
+CONFIG_WEBRTC_AGC_TARGET_DBFS=3
+CONFIG_WEBRTC_AGC_COMPRESSION_GAIN_DB=9
+CONFIG_WEBRTC_AGC_LIMITER=y
+
+# Heap and stack sizing for full WebRTC suite
+CONFIG_HEAP_MEM_POOL_SIZE=131072
+CONFIG_SOF_ZEPHYR_HEAP_SIZE=0x80000

app/webrtc_stubs_overlay.conf

@@ -0,0 +1,16 @@
+# SPDX-License-Identifier: BSD-3-Clause
+#
+# WebRTC Audio Processing Modules Kconfig overlay (LLEXT pass-through stub backends)
+
+CONFIG_COMP_WEBRTC_VAD=m
+CONFIG_COMP_WEBRTC_VAD_STUB=y
+CONFIG_COMP_WEBRTC_NS=m
+CONFIG_COMP_WEBRTC_NS_STUB=y
+CONFIG_COMP_WEBRTC_AEC=m
+CONFIG_COMP_WEBRTC_AEC_STUB=y
+CONFIG_COMP_WEBRTC_NS2=m
+CONFIG_COMP_WEBRTC_NS2_STUB=y
+CONFIG_COMP_WEBRTC_AGC=m
+CONFIG_COMP_WEBRTC_AGC_STUB=y
+CONFIG_COMP_WEBRTC_HPF=m
+CONFIG_COMP_WEBRTC_HPF_STUB=y

src/audio/webrtc_aec/Kconfig

@@ -26,6 +26,7 @@ if COMP_WEBRTC_AEC
 
 choice WEBRTC_AEC_BACKEND
 	prompt "WebRTC AEC processing backend"
+	default COMP_WEBRTC_AEC_STUB if COMP_STUBS
 	default WEBRTC_AEC_BACKEND_FLOAT if COMP_WEBRTC_AEC = y
 	default WEBRTC_AEC_BACKEND_AECM
 

src/audio/webrtc_aec/webrtc_aec.c

@@ -641,14 +641,35 @@ static int webrtc_aec_get_config(struct processing_module *mod, uint32_t config_
 	return -EINVAL;
 }
 
+static bool webrtc_aec_is_ready_to_process(struct processing_module *mod,
+					   struct sof_source **sources,
+					   int num_of_sources,
+					   struct sof_sink **sinks,
+					   int num_of_sinks)
+{
+	struct webrtc_aec_comp_data *cd = module_get_private_data(mod);
+	struct sof_source *mic;
+	struct sof_sink *out;
+
+	if (!cd || !cd->configured || num_of_sources <= cd->mic_src || num_of_sinks < 1)
+		return false;
+
+	mic = sources[cd->mic_src];
+	out = sinks[0];
+
+	return source_get_data_available(mic) >= source_get_min_available(mic) &&
+	       sink_get_free_size(out) >= sink_get_min_free_space(out);
+}
+
 static const struct module_interface webrtc_aec_interface = {
-	.init              = webrtc_aec_init,
-	.prepare           = webrtc_aec_prepare,
-	.process           = webrtc_aec_process,
-	.set_configuration = webrtc_aec_set_config,
-	.get_configuration = webrtc_aec_get_config,
-	.reset             = webrtc_aec_reset,
-	.free              = webrtc_aec_free,
+	.init                = webrtc_aec_init,
+	.prepare             = webrtc_aec_prepare,
+	.is_ready_to_process = webrtc_aec_is_ready_to_process,
+	.process             = webrtc_aec_process,
+	.set_configuration   = webrtc_aec_set_config,
+	.get_configuration   = webrtc_aec_get_config,
+	.reset               = webrtc_aec_reset,
+	.free                = webrtc_aec_free,
 };
 
 #if CONFIG_COMP_WEBRTC_AEC_MODULE
@@ -657,9 +678,19 @@ static const struct module_interface webrtc_aec_interface = {
 #include <module/module/llext.h>
 #include <rimage/sof/user/manifest.h>
 
-static const struct sof_man_module_manifest mod_manifest __section(".module") __used =
-	SOF_LLEXT_MODULE_MANIFEST("WRTCAEC", &webrtc_aec_interface, 2,
-				  SOF_REG_UUID(webrtc_aec), 40);
+static const struct sof_man_module_manifest mod_manifest __section(".module") __used = {
+	.module = {
+		.name = "WRTCAEC",
+		.uuid = SOF_REG_UUID(webrtc_aec),
+		.entry_point = (uint32_t)(&webrtc_aec_interface),
+		.instance_max_count = 40,
+		.type = {
+			.load_type = SOF_MAN_MOD_TYPE_LLEXT,
+			.domain_dp = 1,
+		},
+		.affinity_mask = 2,
+	}
+};
 
 SOF_LLEXT_BUILDINFO;
 

src/audio/webrtc_aec/webrtc_aec.toml

@@ -9,7 +9,7 @@
 	uuid = UUIDREG_STR_WEBRTC_AEC
 	affinity_mask = "0x1"
 	instance_count = "40"
-	domain_types = "0"
+	domain_types = "1"
 	load_type = LOAD_TYPE
 	module_type = "9"
 	auto_start = "0"

src/audio/webrtc_agc/CMakeLists.txt

@@ -37,6 +37,7 @@ else()
       "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/division_operations.c"
       "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_by_2_internal.c"
       "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_fractional.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_48khz.c"
     )
 
     set_source_files_properties(

src/audio/webrtc_agc/Kconfig

@@ -11,7 +11,7 @@ if COMP_WEBRTC_AGC
 
 config COMP_WEBRTC_AGC_STUB
 	bool "Build with pass-through stub backend"
-	default n
+	default y if COMP_STUBS
 	help
 	  Select to build with the pass-through stub backend instead of the real AGC.
 

src/audio/webrtc_agc/README.md

@@ -8,7 +8,7 @@ Wraps WebRTC's digital Automatic Gain Control (`digital_agc.c`) behind the SOF `
 
 - **Adaptive Speech Leveling**: Analyzes short-term speech envelope and applies dynamic digital gain to boost quiet talkers without amplifying background stationary noise.
 - **Integrated Saturation Limiter**: S-curve saturation limiter with fast lookahead attack and smooth release decay, guaranteeing zero digital overs.
-- **Framing FIFO Engine**: Transparently decouples pipeline period sizes (e.g., 1 ms, 2 ms, or 5 ms) by buffering streaming audio into standard 10 ms WebRTC analysis frames (160 samples @ 16 kHz, 80 samples @ 8 kHz).
+- **Framing FIFO Engine**: Transparently decouples pipeline period sizes (e.g., 1 ms, 2 ms, or 5 ms) by buffering streaming audio into standard 10 ms WebRTC analysis frames (80/160/320/480 samples @ 8/16/32/48 kHz).
 - **Pure Fixed-Point Implementation**: Pure integer Q15/Q31 arithmetic with zero floating-point unit dependencies (`CONFIG_FPU=n`).
 - **ALSA IPC4 Switch Control**: Runtime bypass/enable toggling via standard ALSA mixer controls (`SOF_IPC4_SWITCH_CONTROL_PARAM_ID`).
 - **Multichannel Support**: Accommodates up to 4 audio channels with per-channel state instances.

src/audio/webrtc_agc/llext/CMakeLists.txt

@@ -28,6 +28,8 @@ else()
             ${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/dot_product_with_scale.c
             ${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_by_2.c
             ${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_by_2_internal.c
+            ${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_fractional.c
+            ${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_48khz.c
     INCLUDES "${SOF_WEBRTC_APM_SRC_DIR}"
              "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/agc/legacy"
              "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/include"