lgirdwood · Oct 15, 2026 · Oct 15, 2026 · Oct 15, 2026 · Oct 15, 2026 · Oct 15, 2026
@@ -107,6 +107,24 @@ if(NOT CONFIG_COMP_MODULE_SHARED_LIBRARY_BUILD)
 	if(CONFIG_COMP_VOLUME)
 		add_subdirectory(volume)
 	endif()
+	if(CONFIG_COMP_WEBRTC_VAD)
+		add_subdirectory(webrtc_vad)
+	endif()
+	if(CONFIG_COMP_WEBRTC_NS)
+		add_subdirectory(webrtc_ns)
+	endif()
+	if(CONFIG_COMP_WEBRTC_AEC)
+		add_subdirectory(webrtc_aec)
+	endif()
+	if(CONFIG_COMP_WEBRTC_NS2)
+		add_subdirectory(webrtc_ns2)
+	endif()
+	if(CONFIG_COMP_WEBRTC_AGC)
+		add_subdirectory(webrtc_agc)
+	endif()
+	if(CONFIG_COMP_WEBRTC_HPF)
+		add_subdirectory(webrtc_hpf)
+	endif()
 	if(CONFIG_DTS_CODEC)
 		add_subdirectory(codec)
 	endif()

@@ -218,6 +218,12 @@ rsource "tensorflow/Kconfig"
 rsource "tone/Kconfig"
 rsource "up_down_mixer/Kconfig"
 rsource "volume/Kconfig"
+rsource "webrtc_vad/Kconfig"
+rsource "webrtc_ns/Kconfig"
+rsource "webrtc_aec/Kconfig"
+rsource "webrtc_ns2/Kconfig"
+rsource "webrtc_agc/Kconfig"
+rsource "webrtc_hpf/Kconfig"
 # --- End Kconfig Sources (alphabetical order) ---
 
 rsource "level_multiplier/Kconfig"

@@ -0,0 +1,185 @@
+# SPDX-License-Identifier: BSD-3-Clause
+
+if(CONFIG_COMP_WEBRTC_AEC STREQUAL "m" AND DEFINED CONFIG_LLEXT)
+  add_subdirectory(llext ${PROJECT_BINARY_DIR}/webrtc_aec_llext)
+  add_dependencies(app webrtc_aec)
+else()
+  add_local_sources(sof webrtc_aec.c)
+  if(CONFIG_COMP_WEBRTC_AEC_STUB)
+    add_local_sources(sof webrtc_aec-stub.c)
+  else()
+    if(NOT DEFINED SOF_WEBRTC_APM_SRC_DIR)
+      set(SOF_WEBRTC_APM_SRC_DIR "${sof_top_dir}/../modules/audio/webrtc-apm"
+          CACHE PATH "webrtc-audio-processing 0.3.x source directory")
+    endif()
+    cmake_path(NORMAL_PATH SOF_WEBRTC_APM_SRC_DIR)
+
+    zephyr_include_directories(
+      "${SOF_WEBRTC_APM_SRC_DIR}"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/include"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/include"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/include"
+    )
+
+    zephyr_compile_definitions(
+      WEBRTC_ARCH_32_BITS
+      WEBRTC_ARCH_LITTLE_ENDIAN
+    )
+
+    set(_clang_fp_flags "")
+    if(CMAKE_C_COMPILER_ID MATCHES "Clang")
+      set(_clang_fp_flags "-Xclang;-target-feature;-Xclang;-fp")
+    endif()
+
+    if(CONFIG_WEBRTC_AEC_BACKEND_AEC3)
+      if(DEFINED CONFIG_SOC_TOOLCHAIN_NAME)
+        set(_xtensa_target "${CONFIG_SOC_TOOLCHAIN_NAME}")
+      else()
+        set(_xtensa_target "intel_ace30_ptl")
+      endif()
+      set(_sdk_cxx_dir "/home/lrg/zephyr-sdk-1.0.1/gnu/xtensa-${_xtensa_target}_zephyr-elf/xtensa-${_xtensa_target}_zephyr-elf")
+
+      target_include_directories(zephyr_interface SYSTEM BEFORE INTERFACE
+        $<$<COMPILE_LANGUAGE:CXX>:${_sdk_cxx_dir}/include/c++/14.3.0>
+        $<$<COMPILE_LANGUAGE:CXX>:${_sdk_cxx_dir}/include/c++/14.3.0/xtensa-${_xtensa_target}_zephyr-elf>
+        $<$<COMPILE_LANGUAGE:CXX>:${_sdk_cxx_dir}/include>
+      )
+
+      # Do not link heavy libstdc++.a to prevent out-of-range relocations and unwinder bloat.
+      # Minimal C++ runtime, new/delete, and exception stubs are provided in webrtc_aec3_shims.cpp.
+
+      add_local_sources(sof
+        webrtc_aec-aec3.c
+        webrtc_aec-math.c
+        webrtc_aec3_wrapper.cpp
+        webrtc_aec3_shims.cpp
+      )
+
+      file(GLOB _aec3_srcs
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec3/*.cc"
+      )
+      list(FILTER _aec3_srcs EXCLUDE REGEX ".*_(avx2|neon|mips|sse2)\\.cc$")
+
+      set(_aec3_support_srcs
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/api/audio/echo_canceller3_config.cc"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.cc"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/high_pass_filter.cc"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/utility/cascaded_biquad_filter.cc"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec3_buffer/audio_buffer.cc"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/logging/apm_data_dumper.cc"
+      )
+
+      add_local_sources(sof ${_aec3_srcs} ${_aec3_support_srcs})
+
+      set(_aec3_cxx_all
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec3_wrapper.cpp
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec3_shims.cpp
+        ${_aec3_srcs}
+        ${_aec3_support_srcs}
+      )
+
+      set_source_files_properties(
+        ${_aec3_cxx_all}
+        TARGET_DIRECTORY modules_sof
+        PROPERTIES COMPILE_OPTIONS "-std=c++17;-fno-exceptions;-fno-rtti;-O3;-fno-vectorize;-fno-slp-vectorize;${_clang_fp_flags};-DWEBRTC_POSIX;-DWEBRTC_ARCH_GENERIC;-DWEBRTC_APM_DEBUG_DUMP=0;-DRTC_DISABLE_METRICS;-DNDEBUG;-Wno-macro-redefined;-Wno-unused-function;-Wno-double-promotion"
+      )
+    elseif(CONFIG_WEBRTC_AEC_BACKEND_FLOAT)
+      add_local_sources(sof webrtc_aec-aec.c webrtc_aec-math.c)
+      add_local_sources(sof
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/echo_cancellation.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_core.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_rdft.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_resampler.c"
+      )
+    else()
+      add_local_sources(sof webrtc_aec-aecm.c)
+      add_local_sources(sof
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/echo_control_mobile.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/aecm_core.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/aecm_core_c.c"
+      )
+    endif()
+
+    add_local_sources(sof webrtc_aec-shims.c)
+    add_local_sources(sof
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/ring_buffer.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/utility/delay_estimator.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/complex_bit_reverse.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/complex_fft.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/cross_correlation.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/downsample_fast.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/real_fft.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/vector_scaling_operations.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/min_max_operations.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_sqrt_floor.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/randomization_functions.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_init.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_sqrt.c"
+    )
+
+    add_local_sources(sof
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/division_operations.c"
+      "${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(_webrtc_aec_srcs
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/ring_buffer.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/utility/delay_estimator.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/complex_bit_reverse.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/complex_fft.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/cross_correlation.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/downsample_fast.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/real_fft.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/vector_scaling_operations.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/min_max_operations.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_sqrt_floor.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/randomization_functions.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_init.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/spl_sqrt.c"
+      "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/common_audio/signal_processing/resample_48khz.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"
+    )
+
+    list(APPEND _webrtc_aec_srcs
+      ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec.c
+      ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-shims.c
+    )
+
+    if(CONFIG_WEBRTC_AEC_BACKEND_AEC3)
+      list(APPEND _webrtc_aec_srcs
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-aec3.c
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-math.c
+      )
+    elseif(CONFIG_WEBRTC_AEC_BACKEND_FLOAT)
+      list(APPEND _webrtc_aec_srcs
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-aec.c
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-math.c
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/echo_cancellation.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_core.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_rdft.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aec/aec_resampler.c"
+      )
+    else()
+      list(APPEND _webrtc_aec_srcs
+        ${CMAKE_CURRENT_SOURCE_DIR}/webrtc_aec-aecm.c
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/echo_control_mobile.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/aecm_core.c"
+        "${SOF_WEBRTC_APM_SRC_DIR}/webrtc/modules/audio_processing/aecm/aecm_core_c.c"
+      )
+    endif()
+
+    set_source_files_properties(
+      ${_webrtc_aec_srcs}
+      TARGET_DIRECTORY modules_sof
+      PROPERTIES COMPILE_OPTIONS "-O3;${_clang_fp_flags};-Wno-double-promotion;-Wno-implicit-function-declaration;-Wno-unused-function;-Dmalloc=webrtc_malloc;-Dcalloc=webrtc_calloc;-Drealloc=webrtc_realloc;-Dfree=webrtc_free"
+    )
+  endif()
+endif()
@@ -0,0 +1,98 @@
+# SPDX-License-Identifier: BSD-3-Clause
+
+config COMP_WEBRTC_AEC
+	tristate "WebRTC AECm — fixed-point Acoustic Echo Canceller"
+	help
+	  Select to include the WebRTC AECm (Mobile) acoustic echo canceller.
+	  AECm is a fixed-point Q15 implementation derived from the WebRTC
+	  audio processing library. It requires no floating-point unit and is
+	  well suited to Xtensa HiFi3/HiFi4 class DSPs.
+
+	  The module accepts two input streams (2 input pins):
+	    Pin 0 — microphone capture (same pipeline as the output)
+	    Pin 1 — playback echo reference (from the render pipeline)
+	  and produces one denoised microphone output stream.
+
+	  This module is intended as an open-source drop-in replacement for
+	  the google_rtc_audio_processing module that requires a proprietary
+	  library. Both implement the same two-input / one-output topology so
+	  topologies are interchangeable.
+
+	  The real backend requires webrtc-audio-processing 0.3.x source
+	  (west module modules/audio/webrtc-apm, same as COMP_WEBRTC_NS).
+	  For CI/testing without that source, select COMP_WEBRTC_AEC_STUB.
+
+if COMP_WEBRTC_AEC
+
+choice WEBRTC_AEC_BACKEND
+	prompt "WebRTC AEC processing backend"
+	default WEBRTC_AEC_BACKEND_FLOAT if COMP_WEBRTC_AEC = y
+	default WEBRTC_AEC_BACKEND_AECM
+
+config WEBRTC_AEC_BACKEND_FLOAT
+	bool "Classic Float AEC (Multidelay Block Frequency Domain with hardware FPU)"
+	depends on COMP_WEBRTC_AEC = y
+	help
+	  Single-precision floating-point AEC using partitioned-block
+	  frequency domain adaptive filtering (MDF) accelerated by the
+	  hardware FPU and HiFi4 VFPU.
+
+config WEBRTC_AEC_BACKEND_AEC3
+	bool "Modern WebRTC AEC3 (State-of-the-art C++17 Echo Canceller)"
+	depends on COMP_WEBRTC_AEC = y
+	help
+	  Modern WebRTC AEC3 with matched filter delay estimator, multi-partition
+	  linear adaptive filter, non-linear suppression, and full-band ERLE estimator.
+
+config WEBRTC_AEC_PERF_TELEMETRY
+	bool "Enable AEC3 cycle profiling telemetry via printk"
+	default y
+	depends on WEBRTC_AEC_BACKEND_AEC3
+	help
+	  Periodically emits [AEC3_P1] and [AEC3_P2] cycle count breakdowns
+	  every 200ms to monitor DSP cycle budgets.
+
+config WEBRTC_AEC_BACKEND_AECM
+	bool "AECm (Mobile fixed-point integer AEC)"
+	help
+	  Pure integer Q15 fixed-point echo cancellation. Low memory (~12 KB/ch)
+	  and low CPU (~1.5 MCPS).
+
+config COMP_WEBRTC_AEC_STUB
+	bool "WebRTC AEC stub backend (no WebRTC source dependency)"
+	help
+	  Build the webrtc_aec module against a dependency-free stub backend
+	  that passes microphone audio through without modification.
+endchoice
+
+config WEBRTC_AEC_FILTER_LEN_MS
+	int "AECm adaptive filter length in ms (32, 64 or 128)"
+	default 64
+	help
+	  Length of the adaptive echo-cancellation filter. Longer filters
+	  can handle larger acoustic echo delays (e.g. in rooms with long
+	  reverb tails) at the cost of increased memory and CPU.
+	  Valid values: 32, 64, 128.
+
+config WEBRTC_AEC_SUPPRESSION_LEVEL
+	int "AECm comfort-noise suppression level (0=off, 1=low, 2=medium)"
+	range 0 2
+	default 1
+	help
+	  Controls how aggressively residual echo / background noise is
+	  suppressed in the CNG (comfort noise generator) post-filter:
+	    0 - disabled (only echo suppression, no CNG)
+	    1 - low suppression (recommended)
+	    2 - medium suppression
+
+config WEBRTC_AEC_SAMPLE_RATE_HZ
+	int "AECm processing sample rate in Hz"
+	default 16000
+	help
+	  AECm operates at 8000 or 16000 Hz. When the pipeline runs at a
+	  higher rate (e.g. 48 kHz), the module downsamples before AEC
+	  processing and upsamples afterward. 16 kHz is recommended for
+	  best echo suppression quality.
+	  Valid values: 8000, 16000.
+
+endif # COMP_WEBRTC_AEC