2bac896d5e
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
120 lines
3.0 KiB
Plaintext
120 lines
3.0 KiB
Plaintext
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../webrtc.gni")
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group("audio_mixer") {
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deps = [
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":audio_frame_manipulator",
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":audio_mixer_impl",
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]
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}
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rtc_static_library("audio_mixer_impl") {
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sources = [
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"audio_mixer_impl.cc",
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"audio_mixer_impl.h",
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"default_output_rate_calculator.cc",
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"default_output_rate_calculator.h",
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"frame_combiner.cc",
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"frame_combiner.h",
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"output_rate_calculator.h",
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]
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public = [
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"audio_mixer_impl.h",
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"default_output_rate_calculator.h", # For creating a mixer with limiter disabled.
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"frame_combiner.h",
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]
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configs += [ "../audio_processing:apm_debug_dump" ]
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deps = [
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":audio_frame_manipulator",
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"..:module_api",
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"../..:webrtc_common",
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"../../:typedefs",
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"../../api:array_view",
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"../../api/audio:audio_mixer_api",
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"../../audio/utility:audio_frame_operations",
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"../../common_audio",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../system_wrappers",
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"../../system_wrappers:field_trial_api",
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"../../system_wrappers:metrics_api",
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"../audio_processing",
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"../audio_processing:apm_logging",
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"../audio_processing:audio_frame_view",
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"../audio_processing/agc2:fixed_digital",
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]
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}
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rtc_static_library("audio_frame_manipulator") {
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visibility = [
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":*",
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"../../modules:*",
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]
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sources = [
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"audio_frame_manipulator.cc",
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"audio_frame_manipulator.h",
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]
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deps = [
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"..:module_api",
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"../../audio/utility:audio_frame_operations",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("audio_mixer_unittests") {
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testonly = true
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sources = [
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"audio_frame_manipulator_unittest.cc",
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"audio_mixer_impl_unittest.cc",
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"frame_combiner_unittest.cc",
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"gain_change_calculator.cc",
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"gain_change_calculator.h",
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"sine_wave_generator.cc",
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"sine_wave_generator.h",
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]
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deps = [
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":audio_frame_manipulator",
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":audio_mixer_impl",
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"..:module_api",
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"../../api:array_view",
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"../../api/audio:audio_mixer_api",
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"../../audio/utility:audio_frame_operations",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:rtc_task_queue_for_test",
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"../../test:test_support",
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]
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}
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rtc_executable("audio_mixer_test") {
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testonly = true
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sources = [
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"audio_mixer_test.cc",
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]
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deps = [
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":audio_mixer_impl",
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"../../api/audio:audio_mixer_api",
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"../../common_audio",
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"../../rtc_base:rtc_base_approved",
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"../../system_wrappers:system_wrappers_default",
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]
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}
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}
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