Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/aec_dump/BUILD.gn

@@ -0,0 +1,110 @@
+# Copyright (c) 2017 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")  # This contains def of 'rtc_enable_protobuf'
+
+rtc_source_set("aec_dump") {
+  visibility = [ "*" ]
+  sources = [ "aec_dump_factory.h" ]
+
+  deps = [
+    "..:aec_dump_interface",
+    "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base/system:file_wrapper",
+    "../../../rtc_base/system:rtc_export",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("mock_aec_dump") {
+    testonly = true
+    sources = [
+      "mock_aec_dump.cc",
+      "mock_aec_dump.h",
+    ]
+
+    deps = [
+      "..:aec_dump_interface",
+      "..:audioproc_test_utils",
+      "../",
+      "../../../test:test_support",
+    ]
+  }
+
+  rtc_library("mock_aec_dump_unittests") {
+    testonly = true
+    configs += [ "..:apm_debug_dump" ]
+    sources = [ "aec_dump_integration_test.cc" ]
+
+    deps = [
+      ":mock_aec_dump",
+      "..:api",
+      "..:audioproc_test_utils",
+      "../",
+      "../../../rtc_base:rtc_base_approved",
+      "//testing/gtest",
+    ]
+  }
+}
+
+if (rtc_enable_protobuf) {
+  rtc_library("aec_dump_impl") {
+    sources = [
+      "aec_dump_impl.cc",
+      "aec_dump_impl.h",
+      "capture_stream_info.cc",
+      "capture_stream_info.h",
+      "write_to_file_task.cc",
+      "write_to_file_task.h",
+    ]
+
+    deps = [
+      ":aec_dump",
+      "..:aec_dump_interface",
+      "../../../api/audio:audio_frame_api",
+      "../../../api/task_queue",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:ignore_wundef",
+      "../../../rtc_base:protobuf_utils",
+      "../../../rtc_base:rtc_base_approved",
+      "../../../rtc_base:rtc_task_queue",
+      "../../../rtc_base/system:file_wrapper",
+      "../../../system_wrappers",
+    ]
+
+    deps += [ "../:audioproc_debug_proto" ]
+  }
+
+  if (rtc_include_tests) {
+    rtc_library("aec_dump_unittests") {
+      testonly = true
+      defines = []
+      deps = [
+        ":aec_dump",
+        ":aec_dump_impl",
+        "..:audioproc_debug_proto",
+        "../",
+        "../../../rtc_base:task_queue_for_test",
+        "../../../test:fileutils",
+        "../../../test:test_support",
+        "//testing/gtest",
+      ]
+      sources = [ "aec_dump_unittest.cc" ]
+    }
+  }
+}
+
+rtc_library("null_aec_dump_factory") {
+  assert_no_deps = [ ":aec_dump_impl" ]
+  sources = [ "null_aec_dump_factory.cc" ]
+
+  deps = [
+    ":aec_dump",
+    "..:aec_dump_interface",
+  ]
+}

webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
+namespace webrtc {
+
+class RTC_EXPORT AecDumpFactory {
+ public:
+  // The |worker_queue| may not be null and must outlive the created
+  // AecDump instance. |max_log_size_bytes == -1| means the log size
+  // will be unlimited. |handle| may not be null. The AecDump takes
+  // responsibility for |handle| and closes it in the destructor. A
+  // non-null return value indicates that the file has been
+  // sucessfully opened.
+  static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+  static std::unique_ptr<AecDump> Create(std::string file_name,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+  static std::unique_ptr<AecDump> Create(FILE* handle,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_

webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/include/aec_dump.h"
+
+namespace webrtc {
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+}  // namespace webrtc