Doudou Kisabaka fe6595f006 Include all RTP packet infos from the mix list when updating the audio frame for mixing.
Users of the mixer can use this information to determine which sources were included in the frame.

Bug: webrtc:12745
Change-Id: I11a8e3b1f4e8f95eb870336cad8dd082330bdf02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217768
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34035}
2021-05-18 11:05:37 +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") {
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
]
}
rtc_library("audio_mixer_impl") {
visibility = [ "*" ]
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",
]
configs += [ "../audio_processing:apm_debug_dump" ]
deps = [
":audio_frame_manipulator",
"../../api:array_view",
"../../api:rtp_packet_info",
"../../api:scoped_refptr",
"../../api/audio:audio_frame_api",
"../../api/audio:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:safe_conversions",
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../../system_wrappers:metrics",
"../audio_processing:api",
"../audio_processing:apm_logging",
"../audio_processing:audio_frame_view",
"../audio_processing/agc2:fixed_digital",
]
}
rtc_library("audio_frame_manipulator") {
visibility = [
":*",
"../../modules:*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
]
deps = [
"../../api/audio:audio_frame_api",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
}
if (rtc_include_tests) {
rtc_library("audio_mixer_test_utils") {
testonly = true
sources = [
"gain_change_calculator.cc",
"gain_change_calculator.h",
"sine_wave_generator.cc",
"sine_wave_generator.h",
]
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
}
rtc_library("audio_mixer_unittests") {
testonly = true
sources = [
"audio_frame_manipulator_unittest.cc",
"audio_mixer_impl_unittest.cc",
"frame_combiner_unittest.cc",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
":audio_mixer_test_utils",
"../../api:array_view",
"../../api:rtp_packet_info",
"../../api/audio:audio_mixer_api",
"../../api/units:timestamp",
"../../audio/utility:audio_frame_operations",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:task_queue_for_test",
"../../test:test_support",
]
}
if (!build_with_chromium) {
rtc_executable("audio_mixer_test") {
testonly = true
sources = [ "audio_mixer_test.cc" ]
deps = [
":audio_mixer_impl",
"../../api/audio:audio_mixer_api",
"../../common_audio",
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
}
}
}