Before this change, the APM limiter used in FrameCombiner (a sub-component of AudioMixer) only gets to process the data when the number of non-muted streams is >1. If this number varies between <=1 and >1, the limiter's view of the data will have gaps during the periods with <= 1 active stream. This leads to discontinuities in the applied gain. These discontinuities cause clicks in the output audio. This change activates APM limiter processing based on the number of audio streams, independently of their mutedness status. BUG=chromium:695993 Review-Url: https://codereview.webrtc.org/2776113002 Cr-Commit-Position: refs/heads/master@{#17442}
90 lines
2.1 KiB
Plaintext
90 lines
2.1 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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public_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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public_deps = [
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"../../api:audio_mixer_api",
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]
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deps = [
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":audio_frame_manipulator",
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"../..:webrtc_common",
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"../../audio/utility:audio_frame_operations",
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"../../base:rtc_base_approved",
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"../../system_wrappers",
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"../audio_processing",
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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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"../../audio/utility",
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"../../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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"../../api:audio_mixer_api",
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"../../audio/utility:audio_frame_operations",
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"../../base:rtc_base",
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"../../base:rtc_base_approved",
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"../../test:test_support",
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"//testing/gmock",
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]
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}
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}
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