Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 107 insertions, 0 deletions

diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
new file mode 100644
index 0000000000..4b65acdeef
--- /dev/null
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -0,0 +1,107 @@
+/*
+ *  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.
+ */
+
+// Unit tests for FilePlayer.
+
+#include "webrtc/modules/utility/interface/file_player.h"
+
+#include <stdio.h>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/md5digest.h"
+#include "webrtc/base/stringencode.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+DEFINE_bool(file_player_output, false, "Generate reference files.");
+
+namespace webrtc {
+
+class FilePlayerTest : public ::testing::Test {
+ protected:
+  static const uint32_t kId = 0;
+  static const FileFormats kFileFormat = kFileFormatWavFile;
+  static const int kSampleRateHz = 8000;
+
+  FilePlayerTest()
+      : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)),
+        output_file_(NULL) {}
+
+  void SetUp() override {
+    if (FLAGS_file_player_output) {
+      std::string output_file =
+          webrtc::test::OutputPath() + "file_player_unittest_out.pcm";
+      output_file_ = fopen(output_file.c_str(), "wb");
+      ASSERT_TRUE(output_file_ != NULL);
+    }
+  }
+
+  void TearDown() override {
+    if (output_file_)
+      fclose(output_file_);
+  }
+
+  ~FilePlayerTest() { FilePlayer::DestroyFilePlayer(player_); }
+
+  void PlayFileAndCheck(const std::string& input_file,
+                        const std::string& ref_checksum,
+                        int output_length_ms) {
+    const float kScaling = 1;
+    ASSERT_EQ(0,
+              player_->StartPlayingFile(
+                  input_file.c_str(), false, 0, kScaling, 0, 0, NULL));
+    rtc::Md5Digest checksum;
+    for (int i = 0; i < output_length_ms / 10; ++i) {
+      int16_t out[10 * kSampleRateHz / 1000] = {0};
+      size_t num_samples;
+      EXPECT_EQ(0,
+                player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz));
+      checksum.Update(out, num_samples * sizeof(out[0]));
+      if (FLAGS_file_player_output) {
+        ASSERT_EQ(num_samples,
+                  fwrite(out, sizeof(out[0]), num_samples, output_file_));
+      }
+    }
+    char checksum_result[rtc::Md5Digest::kSize];
+    EXPECT_EQ(rtc::Md5Digest::kSize,
+              checksum.Finish(checksum_result, rtc::Md5Digest::kSize));
+    EXPECT_EQ(ref_checksum,
+              rtc::hex_encode(checksum_result, sizeof(checksum_result)));
+  }
+
+  FilePlayer* player_;
+  FILE* output_file_;
+};
+
+TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
+  const std::string kFileName =
+      test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
+  // The file is longer than this, but keeping the output shorter limits the
+  // runtime for the test.
+  const int kOutputLengthMs = 10000;
+  const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a";
+
+  PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
+}
+
+TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcm16File)) {
+  const std::string kFileName =
+      test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
+  // The file is longer than this, but keeping the output shorter limits the
+  // runtime for the test.
+  const int kOutputLengthMs = 10000;
+  const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71";
+
+  PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
+}
+
+}  // namespace webrtc