Change existing aec dump tests to use webrtc::AecDump.

Currently the debug dump functionality of WebRTC (a log of all
AudioProcessing operations) was tested by the following tests:

1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing
   from a debug dump, and verifies that the same debug dump is
   recorded.
2. DebugDumpTest.* which is a comprehensive test of the debug dump
   operations. AudioProcessing configuration is changed, and the dump
   is scanned for the change.
3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that
   debug dumping can be started and files written.

This CL replaces the debug dump mechanism in all these tests to
webrtc::AecDump. Some of the tests are adapted to the chenges of the
new API to AecDump {Start,Stop}DebugRecording: the old functions
signal errors when a file cannot be opened. With AecDump, the
AecDumpFactory instead returns a nullptr.

The CL also changes audioproc_f to use AecDump.

BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2864373002
Cr-Commit-Position: refs/heads/master@{#18605}

webrtc/modules/audio_processing/BUILD.gn

@@ -595,6 +595,9 @@ if (rtc_include_tests) {
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_unittest_proto",
+        "../../base:rtc_task_queue",
+        "aec_dump",
+        "aec_dump:aec_dump_unittests",
       ]
       sources += [
         "aec3/adaptive_fir_filter_unittest.cc",
@@ -745,10 +748,13 @@ if (rtc_include_tests) {
         ":audioproc_test_utils",
         "../../base:protobuf_utils",
         "../../base:rtc_base_approved",
+        "../../base:rtc_task_queue",
         "../../common_audio:common_audio",
         "../../system_wrappers",
         "../../system_wrappers:system_wrappers_default",
         "../../test:test_support",
+        "aec_dump",
+        "aec_dump:aec_dump_impl",
         "//testing/gtest",
         "//third_party/gflags:gflags",
       ]