Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/audio_processing_unittest.cc

Index: webrtc/modules/audio_processing/audio_processing_unittest.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index b52acce230c4d980feb25d8aa89a6f3092b619d1..dc6b098d942162775b69f2d56d54e7fb107592fb 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -20,6 +20,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
@@ -410,17 +412,17 @@ class ApmTest : public ::testing::Test {
 ApmTest::ApmTest()
     : output_path_(test::OutputPath()),
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
-      ref_filename_(test::ResourcePath("audio_processing/output_data_fixed",
-                                       "pb")),
+      ref_filename_(
+          test::ResourcePath("audio_processing/output_data_fixed", "pb")),
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 #if defined(WEBRTC_MAC)
       // A different file for Mac is needed because on this platform the AEC
       // constant |kFixedDelayMs| value is 20 and not 50 as it is on the rest.
-      ref_filename_(test::ResourcePath("audio_processing/output_data_mac",
-                                       "pb")),
+      ref_filename_(
+          test::ResourcePath("audio_processing/output_data_mac", "pb")),
 #else
-      ref_filename_(test::ResourcePath("audio_processing/output_data_float",
-                                       "pb")),
+      ref_filename_(
+          test::ResourcePath("audio_processing/output_data_float", "pb")),
 #endif
 #endif
       frame_(NULL),
@@ -581,6 +583,7 @@ void ApmTest::ProcessWithDefaultStreamParameters(AudioFrame* frame) {
 }
 
 int ApmTest::ProcessStreamChooser(Format format) {
+  rtc::Thread::Current()->SleepMs(1);
   if (format == kIntFormat) {
     return apm_->ProcessStream(frame_);
   }
@@ -594,6 +597,7 @@ int ApmTest::ProcessStreamChooser(Format format) {
 }
 
 int ApmTest::AnalyzeReverseStreamChooser(Format format) {
+  rtc::Thread::Current()->SleepMs(1);
   if (format == kIntFormat) {
     return apm_->ProcessReverseStream(revframe_);
   }
@@ -1706,6 +1710,7 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
                                const std::string& out_filename,
                                Format format,
                                int max_size_bytes) {
+  rtc::TaskQueue worker_queue("ApmTest_worker_queue");
   FILE* in_file = fopen(in_filename.c_str(), "rb");
   ASSERT_TRUE(in_file != NULL);
   audioproc::Event event_msg;
@@ -1733,8 +1738,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
       if (first_init) {
         // StartDebugRecording() writes an additional init message. Don't start
         // recording until after the first init to avoid the extra message.
-        EXPECT_NOERR(
-            apm_->StartDebugRecording(out_filename.c_str(), max_size_bytes));
+        EXPECT_NOERR(apm_->StartDebugRecording(out_filename.c_str(),
+                                               max_size_bytes, &worker_queue));
         first_init = false;
       }
 
@@ -1870,16 +1875,16 @@ TEST_F(ApmTest, VerifyDebugDumpFloat) {
 
 // TODO(andrew): expand test to verify output.
 TEST_F(ApmTest, DebugDump) {
+  rtc::TaskQueue worker_queue("ApmTest_worker_queue");
   const std::string filename =
       test::TempFilename(test::OutputPath(), "debug_aec");
-  EXPECT_EQ(apm_->kNullPointerError,
-            apm_->StartDebugRecording(static_cast<const char*>(NULL), -1));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stopping without having started should be OK.
   EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
 
-  EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str(), -1));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->StartDebugRecording(filename.c_str(), -1, &worker_queue));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
   EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
@@ -1903,18 +1908,17 @@ TEST_F(ApmTest, DebugDump) {
 
 // TODO(andrew): expand test to verify output.
 TEST_F(ApmTest, DebugDumpFromFileHandle) {
-  FILE* fid = NULL;
-  EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid, -1));
+  rtc::TaskQueue worker_queue("ApmTest_worker_queue");
   const std::string filename =
       test::TempFilename(test::OutputPath(), "debug_aec");
-  fid = fopen(filename.c_str(), "w");
+  FILE* fid = fopen(filename.c_str(), "w");
   ASSERT_TRUE(fid);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stopping without having started should be OK.
   EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
 
-  EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1));
+  EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1, &worker_queue));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());