Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 95 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/modules/audio_mixer/BUILD.gn b/third_party/libwebrtc/webrtc/modules/audio_mixer/BUILD.gn
new file mode 100644
index 0000000000..00e3e609b0
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/modules/audio_mixer/BUILD.gn
@@ -0,0 +1,95 @@
+# 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",
+    "..:module_api",
+    "../..:webrtc_common",
+    "../../api:array_view",
+    "../../audio/utility:audio_frame_operations",
+    "../../rtc_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 = [
+    "..:module_api",
+    "../../audio/utility",
+    "../../rtc_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",
+      "..:module_api",
+      "../../api:array_view",
+      "../../api:audio_mixer_api",
+      "../../audio/utility:audio_frame_operations",
+      "../../rtc_base:rtc_base_approved",
+      "../../rtc_base:rtc_task_queue",
+      "../../test:test_support",
+      "//testing/gmock",
+    ]
+  }
+}