Cast: Added support for AES-CTR crypto.

media/cast/test/end2end_unittest.cc

Index: media/cast/test/end2end_unittest.cc
diff --git a/media/cast/test/end2end_unittest.cc b/media/cast/test/end2end_unittest.cc
index e9cefaad1522e30486e71682195d8ce8f95c577e..d82b765d399e8b94d7771640aa6aba2800499d94 100644
--- a/media/cast/test/end2end_unittest.cc
+++ b/media/cast/test/end2end_unittest.cc
@@ -199,6 +199,7 @@ class TestReceiverAudioCallback :
     // Check all out allowed delays.
     double square_error = 0;
     double variance = 0;
+

[Alpha Left Google, 2013/11/07 01:10:11]

nit: please don't fix style in a functional CL in the future.

[pwestin, 2013/11/07 17:16:04]

Done.

     for (size_t i = 0; i < reference_audio_frame.samples.size() - delay; ++i) {
       size_t error = reference_audio_frame.samples[i] -
           output_audio_samples[i + delay];
@@ -247,6 +248,7 @@ class TestReceiverAudioCallback :
     CheckBasicAudioFrame(audio_frame, playout_time);
     ExpectedAudioFrame expected_audio_frame = expected_frame_.front();
     expected_frame_.pop_front();
+
     if (audio_frame->samples.size() == 0) return;  // No more checks needed.
 
     size_t max_delay = CalculateMaxResamplingDelay(48000, 32000,
@@ -284,8 +286,9 @@ class TestReceiverAudioCallback :
     size_t number_of_samples = audio_frame->data.size() / 2;
 
     for (size_t i = 0; i < number_of_samples; ++i) {
-      uint16 sample = (audio_frame->data[1 + i * sizeof(uint16)]) +
-            (static_cast<uint16>(audio_frame->data[i * sizeof(uint16)]) << 8);
+      uint16 sample =
+          static_cast<uint8>(audio_frame->data[1 + i * sizeof(uint16)]) +
+          (static_cast<uint16>(audio_frame->data[i * sizeof(uint16)]) << 8);
       output_audio_samples.push_back(static_cast<int16>(sample));
     }
     EXPECT_GE(ComputeBestSNR(expected_audio_frame.audio_frame,
@@ -541,7 +544,7 @@ TEST_F(End2EndTest, LoopNoLossPcm16) {
   SetupConfig(kPcm16, 32000, false, 1);
   Create();
   test_receiver_audio_callback_->SetExpectedResult(kAudioSamplingFrequency, 20,
-      25);
+      20);
 
   int video_start = 1;
   int audio_diff = kFrameTimerMs;
@@ -862,6 +865,91 @@ TEST_F(End2EndTest, ResetReferenceFrameId) {
             test_receiver_video_callback_->number_times_called());
 }
 
+TEST_F(End2EndTest, CryptoVideo) {
+  SetupConfig(kPcm16, 32000, false, 1);
+
+  video_sender_config_.aes_iv_mask = "1234567890abcdeffedcba0987654321";
+  video_sender_config_.aes_key = "deadbeefcafeb0b0b0b0cafedeadbeef";
+
+  video_receiver_config_.aes_iv_mask = video_sender_config_.aes_iv_mask;
+  video_receiver_config_.aes_key = video_sender_config_.aes_key;
+
+  Create();
+
+  int frames_counter = 0;
+  for (; frames_counter < 20; ++frames_counter) {
+    const base::TimeTicks send_time = testing_clock_.NowTicks();
+
+    SendVideoFrame(frames_counter, send_time);
+
+    test_receiver_video_callback_->AddExpectedResult(frames_counter,
+        video_sender_config_.width, video_sender_config_.height, send_time);
+
+    // GetRawVideoFrame will not return the frame until we are close to the
+    // time in which we should render the frame.
+    frame_receiver_->GetRawVideoFrame(
+        base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,
+                   test_receiver_video_callback_));
+    RunTasks(kFrameTimerMs);
+  }
+  RunTasks(2 * kFrameTimerMs + 1);  // Empty the pipeline.
+  EXPECT_EQ(frames_counter,
+            test_receiver_video_callback_->number_times_called());
+}
+
+
+TEST_F(End2EndTest, CryptoAudio) {
+  SetupConfig(kPcm16, 32000, false, 1);
+
+  audio_sender_config_.aes_iv_mask = "abcdeffedcba12345678900987654321";
+  audio_sender_config_.aes_key = "deadbeefcafecafedeadbeefb0b0b0b0";
+
+  audio_receiver_config_.aes_iv_mask = audio_sender_config_.aes_iv_mask;
+  audio_receiver_config_.aes_key = audio_sender_config_.aes_key;
+
+  Create();
+  test_receiver_audio_callback_->SetExpectedResult(32000, 18, 20);
+
+  int frames_counter = 0;
+  for (; frames_counter < 20; ++frames_counter) {
+    int num_10ms_blocks = 2;
+
+    const base::TimeTicks send_time = testing_clock_.NowTicks();
+
+    PcmAudioFrame* audio_frame = CreateAudioFrame(num_10ms_blocks,
+        kSoundFrequency, 32000);
+
+    if (frames_counter != 0) {
+      // Due to the re-sampler and NetEq in the webrtc AudioCodingModule the
+      // first samples will be 0 and then slowly ramp up to its real amplitude;
+      // ignore the first frame.
+      test_receiver_audio_callback_->AddExpectedResult(audio_frame,
+          num_10ms_blocks, send_time);
+    }
+    frame_input_->InsertRawAudioFrame(audio_frame, send_time,
+       base::Bind(FrameInput::DeleteAudioFrame, audio_frame));
+
+    RunTasks(num_10ms_blocks * 10);
+
+    if (frames_counter == 0) {
+      frame_receiver_->GetRawAudioFrame(num_10ms_blocks,
+          32000,
+          base::Bind(&TestReceiverAudioCallback::IgnoreAudioFrame,
+              test_receiver_audio_callback_));
+    } else {
+      frame_receiver_->GetRawAudioFrame(num_10ms_blocks,
+          32000,
+          base::Bind(&TestReceiverAudioCallback::CheckPcmAudioFrame,
+              test_receiver_audio_callback_));
+    }
+  }
+  RunTasks(2 * kFrameTimerMs + 1);  // Empty the pipeline.
+  EXPECT_EQ(frames_counter - 1,
+            test_receiver_audio_callback_->number_times_called());
+}
+
+
+
 // TODO(pwestin): Add repeatable packet loss test.
 // TODO(pwestin): Add test for misaligned send get calls.
 // TODO(pwestin): Add more tests that does not resample.