Files
platform-external-webrtc/webrtc/modules/audio_mixer/BUILD.gn
aleloi 2c9306ed50 Send data from mixer to APM limiter more often.
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}
2017-03-29 11:25:16 +00:00

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# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../webrtc.gni")
group("audio_mixer") {
public_deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
]
}
rtc_static_library("audio_mixer_impl") {
sources = [
"audio_mixer_impl.cc",
"audio_mixer_impl.h",
"default_output_rate_calculator.cc",
"default_output_rate_calculator.h",
"frame_combiner.cc",
"frame_combiner.h",
"output_rate_calculator.h",
]
public = [
"audio_mixer_impl.h",
"default_output_rate_calculator.h", # For creating a mixer with limiter disabled.
"frame_combiner.h",
]
public_deps = [
"../../api:audio_mixer_api",
]
deps = [
":audio_frame_manipulator",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_base_approved",
"../../system_wrappers",
"../audio_processing",
]
}
rtc_static_library("audio_frame_manipulator") {
visibility = [
":*",
"../../modules:*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
]
deps = [
"../../audio/utility",
"../../base:rtc_base_approved",
]
}
if (rtc_include_tests) {
rtc_source_set("audio_mixer_unittests") {
testonly = true
sources = [
"audio_frame_manipulator_unittest.cc",
"audio_mixer_impl_unittest.cc",
"frame_combiner_unittest.cc",
"gain_change_calculator.cc",
"gain_change_calculator.h",
"sine_wave_generator.cc",
"sine_wave_generator.h",
]
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
"../../api:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gmock",
]
}
}