Simplifying audio network adaptor by moving receiver frame length range to ctor.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 6a4c47c0e33a189bcb910a5290dfd7facad3905f..170dc7f685c076abc447f4ef6310a754b8056ea9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -34,6 +34,7 @@ AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
   config.payload_type = codec_inst.pltype;
   config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
                                                 : AudioEncoderOpus::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
   return config;
 }
 
@@ -222,9 +223,33 @@ TEST(AudioEncoderOpusTest, PacketLossRateOptimized) {
   // clang-format on
 }
 
+namespace {
+
+void CheckArrayView(const std::vector<int>& expect_values,
+                    const rtc::ArrayView<const int>& array_view) {

[kwiberg-webrtc, 2016/10/24 11:58:35]

Pass ArrayView by value.

+  EXPECT_THAT(expect_values, ::testing::ElementsAreArray(array_view.data(),
+                                                         array_view.size()));

[kwiberg-webrtc, 2016/10/24 11:58:35]

The actual value should be the first argument, and the matcher the second
argument:
https://github.com/google/googletest/blob/master/googlemock/docs/CheatSheet.m...

So

  EXPECT_THAT(array_view,
              ElementsAreArray(expected.data(), expected.size()));

+}
+
+}  // namespace
+
+TEST(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
+  auto states = CreateCodec(2);
+  // Before calling to |SetReceiverFrameLengthRange|,
+  // |supported_frame_lengths_ms| should contain only the frame length being
+  // used.
+  CheckArrayView({states.encoder->next_frame_length_ms()},
+                 states.encoder->supported_frame_lengths_ms());

[kwiberg-webrtc, 2016/10/24 11:58:35]

I don't think you need the separate function:

EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
            ElementsAre(states.encoder->next_frame_length_ms()));

+  states.encoder->SetReceiverFrameLengthRange(0, 12345);
+  CheckArrayView({20, 60}, states.encoder->supported_frame_lengths_ms());

[kwiberg-webrtc, 2016/10/24 11:58:35]

EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
            ElementsAre(20, 60));

+  states.encoder->SetReceiverFrameLengthRange(21, 60);
+  CheckArrayView({60}, states.encoder->supported_frame_lengths_ms());
+  states.encoder->SetReceiverFrameLengthRange(20, 59);
+  CheckArrayView({20}, states.encoder->supported_frame_lengths_ms());
+}
+
 TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetUplinkBandwidth) {
   auto states = CreateCodec(2);
-  printf("passed!\n");
   states.encoder->EnableAudioNetworkAdaptor("", nullptr);
 
   auto config = CreateEncoderRuntimeConfig();
@@ -292,24 +317,6 @@ TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetRtt) {
 }
 
 TEST(AudioEncoderOpusTest,
-     InvokeAudioNetworkAdaptorOnSetReceiverFrameLengthRange) {
-  auto states = CreateCodec(2);
-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
-
-  auto config = CreateEncoderRuntimeConfig();
-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
-      .WillOnce(Return(config));
-
-  constexpr int kMinFrameLength = 10;
-  constexpr int kMaxFrameLength = 60;
-  EXPECT_CALL(**states.mock_audio_network_adaptor,
-              SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength));
-  states.encoder->SetReceiverFrameLengthRange(kMinFrameLength, kMaxFrameLength);
-
-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
-}
-
-TEST(AudioEncoderOpusTest,
      PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
   auto states = CreateCodec(2);