Merge two files with AudioEncoderOpus tests

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 33afa5fcc57b39b8247948d2c9b92292d94b0a0c..71c89bf099ca8b694556a847304ef8c337d52b66 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
@@ -10,32 +10,97 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 
 namespace webrtc {
 
+namespace {
+const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
+}  // namespace
+
 class AudioEncoderOpusTest : public ::testing::Test {
  protected:
-  // The constructor simply creates an Opus encoder with default configuration.
-  AudioEncoderOpusTest()
-      : opus_(new AudioEncoderOpus(AudioEncoderOpus::Config())) {}
-
-  // Repeatedly sets packet loss rates in the range [from, to], increasing by
-  // 0.01 in each step. The function verifies that the actual loss rate is
-  // |expected_return|.
-  void TestSetPacketLossRate(double from, double to, double expected_return) {
-    ASSERT_TRUE(opus_);
-    for (double loss = from; loss <= to;
-         (to >= from) ? loss += 0.01 : loss -= 0.01) {
-      opus_->SetProjectedPacketLossRate(loss);
-      EXPECT_DOUBLE_EQ(expected_return, opus_->packet_loss_rate());
-    }
+  void CreateCodec(int num_channels) {
+    codec_inst_.channels = num_channels;
+    encoder_.reset(new AudioEncoderOpus(codec_inst_));
+    auto expected_app =
+        num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
+    EXPECT_EQ(expected_app, encoder_->application());
   }
 
-  rtc::scoped_ptr<AudioEncoderOpus> opus_;
+  CodecInst codec_inst_ = kOpusSettings;
+  rtc::scoped_ptr<AudioEncoderOpus> encoder_;
 };
 
+TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  CreateCodec(1);
+}
+
+TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  CreateCodec(2);
+}
+
+TEST_F(AudioEncoderOpusTest, ChangeApplicationMode) {
+  CreateCodec(2);
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+}
+
+TEST_F(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  CreateCodec(2);
+
+  // Trigger a reset.
+  encoder_->Reset();
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+
+  // Now change to kVoip.
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+
+  // Trigger a reset again.
+  encoder_->Reset();
+  // Verify that the mode is still kVoip.
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+}
+
+TEST_F(AudioEncoderOpusTest, ToggleDtx) {
+  CreateCodec(2);
+  // Enable DTX
+  EXPECT_TRUE(encoder_->SetDtx(true));
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+  // Turn off DTX.
+  EXPECT_TRUE(encoder_->SetDtx(false));
+}
+
+TEST_F(AudioEncoderOpusTest, SetBitrate) {
+  CreateCodec(1);
+  // Constants are replicated from audio_encoder_opus.cc.
+  const int kMinBitrateBps = 500;
+  const int kMaxBitrateBps = 512000;
+  // Set a too low bitrate.
+  encoder_->SetTargetBitrate(kMinBitrateBps - 1);
+  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  // Set a too high bitrate.
+  encoder_->SetTargetBitrate(kMaxBitrateBps + 1);
+  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  // Set the minimum rate.
+  encoder_->SetTargetBitrate(kMinBitrateBps);
+  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  // Set the maximum rate.
+  encoder_->SetTargetBitrate(kMaxBitrateBps);
+  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  // Set rates from 1000 up to 32000 bps.
+  for (int rate = 1000; rate <= 32000; rate += 1000) {
+    encoder_->SetTargetBitrate(rate);
+    EXPECT_EQ(rate, encoder_->GetTargetBitrate());
+  }
+}
+
 namespace {
+
 // These constants correspond to those used in
 // AudioEncoderOpus::SetProjectedPacketLossRate.
 const double kPacketLossRate20 = 0.20;
@@ -45,36 +110,52 @@ const double kPacketLossRate1 = 0.01;
 const double kLossRate20Margin = 0.02;
 const double kLossRate10Margin = 0.01;
 const double kLossRate5Margin = 0.01;
+
+// Repeatedly sets packet loss rates in the range [from, to], increasing by
+// 0.01 in each step. The function verifies that the actual loss rate is
+// |expected_return|.
+void TestSetPacketLossRate(AudioEncoderOpus* encoder,
+                           double from,
+                           double to,
+                           double expected_return) {
+  for (double loss = from; loss <= to;
+       (to >= from) ? loss += 0.01 : loss -= 0.01) {
+    encoder->SetProjectedPacketLossRate(loss);
+    EXPECT_DOUBLE_EQ(expected_return, encoder->packet_loss_rate());
+  }
+}
+
 }  // namespace
 
 TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) {
+  CreateCodec(1);
+
   // Note that the order of the following calls is critical.
-  TestSetPacketLossRate(0.0, 0.0, 0.0);
-  TestSetPacketLossRate(kPacketLossRate1,
+  TestSetPacketLossRate(encoder_.get(), 0.0, 0.0, 0.0);
+  TestSetPacketLossRate(encoder_.get(), kPacketLossRate1,
                         kPacketLossRate5 + kLossRate5Margin - 0.01,
                         kPacketLossRate1);
-  TestSetPacketLossRate(kPacketLossRate5 + kLossRate5Margin,
+  TestSetPacketLossRate(encoder_.get(), kPacketLossRate5 + kLossRate5Margin,
                         kPacketLossRate10 + kLossRate10Margin - 0.01,
                         kPacketLossRate5);
-  TestSetPacketLossRate(kPacketLossRate10 + kLossRate10Margin,
+  TestSetPacketLossRate(encoder_.get(), kPacketLossRate10 + kLossRate10Margin,
                         kPacketLossRate20 + kLossRate20Margin - 0.01,
                         kPacketLossRate10);
-  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
-                        1.0,
-                        kPacketLossRate20);
-  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
+  TestSetPacketLossRate(encoder_.get(), kPacketLossRate20 + kLossRate20Margin,
+                        1.0, kPacketLossRate20);
+  TestSetPacketLossRate(encoder_.get(), kPacketLossRate20 + kLossRate20Margin,
                         kPacketLossRate20 - kLossRate20Margin,
                         kPacketLossRate20);
-  TestSetPacketLossRate(kPacketLossRate20 - kLossRate20Margin - 0.01,
-                        kPacketLossRate10 - kLossRate10Margin,
-                        kPacketLossRate10);
-  TestSetPacketLossRate(kPacketLossRate10 - kLossRate10Margin - 0.01,
-                        kPacketLossRate5 - kLossRate5Margin,
-                        kPacketLossRate5);
-  TestSetPacketLossRate(kPacketLossRate5 - kLossRate5Margin - 0.01,
-                        kPacketLossRate1,
-                        kPacketLossRate1);
-  TestSetPacketLossRate(0.0, 0.0, 0.0);
+  TestSetPacketLossRate(
+      encoder_.get(), kPacketLossRate20 - kLossRate20Margin - 0.01,
+      kPacketLossRate10 - kLossRate10Margin, kPacketLossRate10);
+  TestSetPacketLossRate(encoder_.get(),
+                        kPacketLossRate10 - kLossRate10Margin - 0.01,
+                        kPacketLossRate5 - kLossRate5Margin, kPacketLossRate5);
+  TestSetPacketLossRate(encoder_.get(),
+                        kPacketLossRate5 - kLossRate5Margin - 0.01,
+                        kPacketLossRate1, kPacketLossRate1);
+  TestSetPacketLossRate(encoder_.get(), 0.0, 0.0, 0.0);
 }
 
 }  // namespace webrtc