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Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
146 lines
3.8 KiB
Text
146 lines
3.8 KiB
Text
# 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_library("audio_mixer_impl") {
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visibility = [ "*" ]
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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
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# 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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"../../api:array_view",
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"../../api:rtp_packet_info",
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"../../api:scoped_refptr",
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"../../api/audio:audio_frame_api",
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"../../api/audio:audio_mixer_api",
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"../../api/audio:audio_processing",
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"../../api/audio:audio_processing",
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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:event_tracer",
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"../../rtc_base:logging",
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"../../rtc_base:macromagic",
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"../../rtc_base:race_checker",
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"../../rtc_base:refcount",
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"../../rtc_base:safe_conversions",
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"../../rtc_base/synchronization:mutex",
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"../../system_wrappers",
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"../../system_wrappers:metrics",
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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_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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"../../api/audio:audio_frame_api",
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"../../audio/utility:audio_frame_operations",
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"../../rtc_base:checks",
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]
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}
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if (rtc_include_tests) {
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rtc_library("audio_mixer_test_utils") {
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testonly = true
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sources = [
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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:array_view",
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"../../api/audio:audio_frame_api",
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"../../rtc_base:checks",
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"../../rtc_base:safe_conversions",
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]
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}
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rtc_library("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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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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deps = [
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":audio_frame_manipulator",
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":audio_mixer_impl",
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":audio_mixer_test_utils",
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"../../api:array_view",
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"../../api:rtp_packet_info",
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"../../api/audio:audio_mixer_api",
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"../../api/units:timestamp",
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"../../audio/utility:audio_frame_operations",
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"../../rtc_base:checks",
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"../../rtc_base:stringutils",
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"../../rtc_base:task_queue_for_test",
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"../../system_wrappers:metrics",
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"../../test:test_support",
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]
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}
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if (!build_with_chromium) {
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rtc_executable("audio_mixer_test") {
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testonly = true
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sources = [ "audio_mixer_test.cc" ]
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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:stringutils",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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]
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}
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}
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}
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