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

webrtc/modules/audio_processing/audio_processing_unittest.cc

 /*
  *  Copyright (c) 2012 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 <math.h>
 #include <stdio.h>
 
 #include <algorithm>
 #include <limits>
 #include <memory>
 #include <queue>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/safe_minmax.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"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 #include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ skipping 1663 matching lines @@
   apm_->echo_cancellation()->set_stream_drift_samples(0);
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 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;
   bool first_init = true;
 
   while (ReadMessageFromFile(in_file, &event_msg)) {
     if (event_msg.type() == audioproc::Event::INIT) {
       const audioproc::Init msg = event_msg.init();
       int reverse_sample_rate = msg.sample_rate();
       if (msg.has_reverse_sample_rate()) {
         reverse_sample_rate = msg.reverse_sample_rate();
       }
       int output_sample_rate = msg.sample_rate();
       if (msg.has_output_sample_rate()) {
         output_sample_rate = msg.output_sample_rate();
       }
 
       Init(msg.sample_rate(),
            output_sample_rate,
            reverse_sample_rate,
            msg.num_input_channels(),
            msg.num_output_channels(),
            msg.num_reverse_channels(),
            false);
       if (first_init) {
-        // StartDebugRecording() writes an additional init message. Don't start
+        // AttachAecDump() 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));
+        auto aec_dump =
+            AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue);
+        EXPECT_TRUE(aec_dump);
+        apm_->AttachAecDump(std::move(aec_dump));
         first_init = false;
       }
 
     } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) {
       const audioproc::ReverseStream msg = event_msg.reverse_stream();
 
       if (msg.channel_size() > 0) {
         ASSERT_EQ(revframe_->num_channels_,
                   static_cast<size_t>(msg.channel_size()));
         for (int i = 0; i < msg.channel_size(); ++i) {
@@ skipping 36 matching lines @@
         memcpy(frame_->mutable_data(), msg.input_data().data(),
                msg.input_data().size());
         if (format == kFloatFormat) {
           // We're using an int16 input file; convert to float.
           ConvertToFloat(*frame_, float_cb_.get());
         }
       }
       ProcessStreamChooser(format);
     }
   }
-  EXPECT_NOERR(apm_->StopDebugRecording());
+  apm_->DetachAecDump();
   fclose(in_file);
 }
 
 void ApmTest::VerifyDebugDumpTest(Format format) {
   const std::string in_filename = test::ResourcePath("ref03", "aecdump");
   std::string format_string;
   switch (format) {
     case kIntFormat:
       format_string = "_int";
       break;
@@ skipping 59 matching lines @@
   VerifyDebugDumpTest(kIntFormat);
 }
 
 TEST_F(ApmTest, VerifyDebugDumpFloat) {
   VerifyDebugDumpTest(kFloatFormat);
 }
 #endif
 
 // 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));
+  {
+    auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue);
+    EXPECT_FALSE(aec_dump);
+  }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stopping without having started should be OK.
-  EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+  apm_->DetachAecDump();
 
-  EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str(), -1));
+  auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue);
+  EXPECT_TRUE(aec_dump);
+  apm_->AttachAecDump(std::move(aec_dump));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
-  EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+  apm_->DetachAecDump();
 
   // Verify the file has been written.
   FILE* fid = fopen(filename.c_str(), "r");
   ASSERT_TRUE(fid != NULL);
 
   // Clean it up.
   ASSERT_EQ(0, fclose(fid));
   ASSERT_EQ(0, remove(filename.c_str()));
 #else
-  EXPECT_EQ(apm_->kUnsupportedFunctionError,
-            apm_->StartDebugRecording(filename.c_str(), -1));
-  EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
-
   // Verify the file has NOT been written.
   ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 // 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());
+  apm_->DetachAecDump();
 
-  EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1));
+  auto aec_dump = AecDumpFactory::Create(fid, -1, &worker_queue);
+  EXPECT_TRUE(aec_dump);
+  apm_->AttachAecDump(std::move(aec_dump));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+  apm_->DetachAecDump();
 
   // Verify the file has been written.
   fid = fopen(filename.c_str(), "r");
   ASSERT_TRUE(fid != NULL);
 
   // Clean it up.
   ASSERT_EQ(0, fclose(fid));
   ASSERT_EQ(0, remove(filename.c_str()));
 #else
-  EXPECT_EQ(apm_->kUnsupportedFunctionError,
-            apm_->StartDebugRecording(fid, -1));
-  EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
-
   ASSERT_EQ(0, fclose(fid));
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
   audioproc::OutputData ref_data;
   OpenFileAndReadMessage(ref_filename_, &ref_data);
 
   Config config;
   config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
@@ skipping 933 matching lines @@
   // TODO(peah): Remove the testing for
   // apm->capture_nonlocked_.level_controller_enabled once the value in config_
   // is instead used to activate the level controller.
   EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
   EXPECT_NEAR(kTargetLcPeakLeveldBFS,
               apm->config_.level_controller.initial_peak_level_dbfs,
               std::numeric_limits<float>::epsilon());
 }
 
 }  // namespace webrtc