1 files changed, 826 insertions, 0 deletions

diff --git a/media/webrtc/signaling/gtest/audioconduit_unittests.cpp b/media/webrtc/signaling/gtest/audioconduit_unittests.cpp
new file mode 100644
index 0000000000..2348adb7bd
--- /dev/null
+++ b/media/webrtc/signaling/gtest/audioconduit_unittests.cpp
@@ -0,0 +1,826 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+#include "AudioConduit.h"
+
+#include "MockCall.h"
+
+using namespace mozilla;
+
+namespace test {
+
+class MockChannelProxy : public webrtc::voe::ChannelProxy {
+ public:
+  void SetSendAudioLevelIndicationStatus(bool enable, int id) override {
+    mSetSendAudioLevelIndicationStatusEnabled = enable;
+  }
+
+  void SetReceiveAudioLevelIndicationStatus(bool enable, int id,
+                                            bool isLevelSsrc = true) override {
+    mSetReceiveAudioLevelIndicationStatusEnabled = enable;
+  }
+
+  void SetReceiveCsrcAudioLevelIndicationStatus(bool enable, int id) override {
+    mSetReceiveCsrcAudioLevelIndicationStatusEnabled = enable;
+  }
+
+  void SetSendMIDStatus(bool enable, int id) override {
+    mSetSendMIDStatusEnabled = enable;
+  }
+
+  void RegisterTransport(Transport* transport) override {}
+
+  void SetRtcpEventObserver(webrtc::RtcpEventObserver* observer) override {}
+
+  bool mSetSendAudioLevelIndicationStatusEnabled;
+  bool mSetReceiveAudioLevelIndicationStatusEnabled;
+  bool mSetReceiveCsrcAudioLevelIndicationStatusEnabled;
+  bool mSetSendMIDStatusEnabled;
+};
+
+class AudioConduitWithMockChannelProxy : public WebrtcAudioConduit {
+ public:
+  AudioConduitWithMockChannelProxy(RefPtr<WebRtcCallWrapper> aCall,
+                                   nsCOMPtr<nsISerialEventTarget> aStsThread)
+      : WebrtcAudioConduit(aCall, aStsThread) {}
+
+  MediaConduitErrorCode Init() override {
+    mRecvChannelProxy.reset(new MockChannelProxy);
+    mSendChannelProxy.reset(new MockChannelProxy);
+
+    return kMediaConduitNoError;
+  }
+
+  void DeleteChannels() override {
+    mRecvChannelProxy = nullptr;
+    mSendChannelProxy = nullptr;
+  }
+
+  const MockChannelProxy* GetRecvChannelProxy() {
+    return static_cast<MockChannelProxy*>(mRecvChannelProxy.get());
+  }
+
+  const MockChannelProxy* GetSendChannelProxy() {
+    return static_cast<MockChannelProxy*>(mSendChannelProxy.get());
+  }
+};
+
+class AudioConduitTest : public ::testing::Test {
+ public:
+  AudioConduitTest() : mCall(new MockCall()) {
+    mAudioConduit = new AudioConduitWithMockChannelProxy(
+        WebRtcCallWrapper::Create(UniquePtr<MockCall>(mCall)),
+        GetCurrentSerialEventTarget());
+    mAudioConduit->Init();
+  }
+
+  MockCall* mCall;
+  RefPtr<AudioConduitWithMockChannelProxy> mAudioConduit;
+};
+
+TEST_F(AudioConduitTest, TestConfigureSendMediaCodec) {
+  MediaConduitErrorCode ec;
+
+  // defaults
+  AudioCodecConfig codecConfig(114, "opus", 48000, 2, false);
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+
+  // null codec
+  ec = mAudioConduit->ConfigureSendMediaCodec(nullptr);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // empty codec name
+  codecConfig = AudioCodecConfig(114, "", 48000, 2, false);
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // long codec name
+  size_t longNameLength = WebrtcAudioConduit::CODEC_PLNAME_SIZE + 2;
+  char* longName = new char[longNameLength];
+  memset(longName, 'A', longNameLength - 2);
+  longName[longNameLength - 1] = 0;
+  codecConfig = AudioCodecConfig(114, longName, 48000, 2, false);
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+  delete[] longName;
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMono) {
+  MediaConduitErrorCode ec;
+
+  // opus mono
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 1, false);
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 1UL);
+    ASSERT_EQ(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusFEC) {
+  MediaConduitErrorCode ec;
+
+  // opus with inband Forward Error Correction
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, true);
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_NE(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("useinbandfec"), "1");
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMaxPlaybackRate) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mMaxPlaybackRate = 1234;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxplaybackrate"), "1234");
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMaxAverageBitrate) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mMaxAverageBitrate = 12345;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxaveragebitrate"), "12345");
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusDtx) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mDTXEnabled = true;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("usedtx"), "1");
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusCbr) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mCbrEnabled = true;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("cbr"), "1");
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusPtime) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mFrameSizeMs = 100;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("ptime"), "100");
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMinPtime) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mMinFrameSizeMs = 201;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("minptime"), "201");
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMaxPtime) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, false);
+  codecConfig.mMaxFrameSizeMs = 321;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("maxptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxptime"), "321");
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusAllParams) {
+  MediaConduitErrorCode ec;
+
+  AudioCodecConfig codecConfig = AudioCodecConfig(114, "opus", 48000, 2, true);
+  codecConfig.mMaxPlaybackRate = 5432;
+  codecConfig.mMaxAverageBitrate = 54321;
+  codecConfig.mDTXEnabled = true;
+  codecConfig.mCbrEnabled = true;
+  codecConfig.mFrameSizeMs = 999;
+  codecConfig.mMinFrameSizeMs = 123;
+  codecConfig.mMaxFrameSizeMs = 789;
+  ec = mAudioConduit->ConfigureSendMediaCodec(&codecConfig);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  mAudioConduit->StartTransmitting();
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioSendConfig.send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_NE(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("useinbandfec"), "1");
+    ASSERT_NE(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxplaybackrate"), "5432");
+    ASSERT_NE(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxaveragebitrate"), "54321");
+    ASSERT_NE(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("usedtx"), "1");
+    ASSERT_NE(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("cbr"), "1");
+    ASSERT_NE(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("ptime"), "999");
+    ASSERT_NE(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("minptime"), "123");
+    ASSERT_NE(f.parameters.find("maxptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxptime"), "789");
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveMediaCodecs) {
+  MediaConduitErrorCode ec;
+
+  // just default opus stereo
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "");
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+
+  // multiple codecs
+  codecs.clear();
+  codecs.emplace_back(new AudioCodecConfig(9, "g722", 16000, 2, false));
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "");
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 2U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(9);
+    ASSERT_EQ(f.name, "g722");
+    ASSERT_EQ(f.clockrate_hz, 16000);
+    ASSERT_EQ(f.num_channels, 2U);
+    ASSERT_EQ(f.parameters.size(), 0U);
+  }
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2U);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+  }
+
+  // no codecs
+  codecs.clear();
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // null codec
+  codecs.clear();
+  codecs.push_back(nullptr);
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // invalid codec name
+  codecs.clear();
+  codecs.emplace_back(new AudioCodecConfig(114, "", 48000, 2, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // invalid number of channels
+  codecs.clear();
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 42, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // long codec name
+  codecs.clear();
+  size_t longNameLength = WebrtcAudioConduit::CODEC_PLNAME_SIZE + 2;
+  char* longName = new char[longNameLength];
+  memset(longName, 'A', longNameLength - 2);
+  longName[longNameLength - 1] = 0;
+  codecs.emplace_back(new AudioCodecConfig(100, longName, 48000, 42, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+  delete[] longName;
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusMono) {
+  MediaConduitErrorCode ec;
+
+  // opus mono
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 1, false));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "");
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 1UL);
+    ASSERT_EQ(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusDtx) {
+  MediaConduitErrorCode ec;
+
+  // opus mono
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, false));
+  codecs[0]->mDTXEnabled = true;
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "");
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_NE(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("usedtx"), "1");
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusFEC) {
+  MediaConduitErrorCode ec;
+
+  // opus with inband Forward Error Correction
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, true));
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "");
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_NE(f.parameters.find("stereo"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_NE(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("useinbandfec"), "1");
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusMaxPlaybackRate) {
+  MediaConduitErrorCode ec;
+
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, false));
+
+  codecs[0]->mMaxPlaybackRate = 0;
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.count("maxplaybackrate"), 0U);
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+
+  codecs[0]->mMaxPlaybackRate = 8000;
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.at("maxplaybackrate"), "8000");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxaveragebitrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusMaxAverageBitrate) {
+  MediaConduitErrorCode ec;
+
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, false));
+
+  codecs[0]->mMaxAverageBitrate = 0;
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.count("maxaveragebitrate"), 0U);
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+
+  codecs[0]->mMaxAverageBitrate = 8000;
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.find("useinbandfec"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxplaybackrate"), f.parameters.end());
+    ASSERT_EQ(f.parameters.at("maxaveragebitrate"), "8000");
+    ASSERT_EQ(f.parameters.find("usedtx"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("cbr"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("ptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("minptime"), f.parameters.end());
+    ASSERT_EQ(f.parameters.find("maxptime"), f.parameters.end());
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusAllParameters) {
+  MediaConduitErrorCode ec;
+
+  std::vector<UniquePtr<mozilla::AudioCodecConfig>> codecs;
+  codecs.emplace_back(new AudioCodecConfig(114, "opus", 48000, 2, true));
+
+  codecs[0]->mMaxPlaybackRate = 8000;
+  codecs[0]->mMaxAverageBitrate = 9000;
+  codecs[0]->mDTXEnabled = true;
+  codecs[0]->mCbrEnabled = true;
+  codecs[0]->mFrameSizeMs = 10;
+  codecs[0]->mMinFrameSizeMs = 20;
+  codecs[0]->mMaxFrameSizeMs = 30;
+
+  ec = mAudioConduit->ConfigureRecvMediaCodecs(codecs);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        mCall->mAudioReceiveConfig.decoder_map.at(114);
+    ASSERT_EQ(f.name, "opus");
+    ASSERT_EQ(f.clockrate_hz, 48000);
+    ASSERT_EQ(f.num_channels, 2UL);
+    ASSERT_EQ(f.parameters.at("stereo"), "1");
+    ASSERT_EQ(f.parameters.at("useinbandfec"), "1");
+    ASSERT_EQ(f.parameters.at("maxplaybackrate"), "8000");
+    ASSERT_EQ(f.parameters.at("maxaveragebitrate"), "9000");
+    ASSERT_EQ(f.parameters.at("usedtx"), "1");
+    ASSERT_EQ(f.parameters.at("cbr"), "1");
+    ASSERT_EQ(f.parameters.at("ptime"), "10");
+    ASSERT_EQ(f.parameters.at("minptime"), "20");
+    ASSERT_EQ(f.parameters.at("maxptime"), "30");
+  }
+}
+
+TEST_F(AudioConduitTest, TestSetLocalRTPExtensions) {
+  MediaConduitErrorCode ec;
+
+  using LocalDirection = MediaSessionConduitLocalDirection;
+
+  RtpExtList extensions;
+
+  // Empty extensions
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kRecv, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kSend, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+
+  // Audio level
+  webrtc::RtpExtension extension;
+  extension.uri = webrtc::RtpExtension::kAudioLevelUri;
+  extensions.emplace_back(extension);
+
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kRecv, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StartReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StopReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.rtp.extensions.back().uri,
+            webrtc::RtpExtension::kAudioLevelUri);
+  ASSERT_TRUE(mAudioConduit->GetRecvChannelProxy()
+                  ->mSetReceiveAudioLevelIndicationStatusEnabled);
+
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kSend, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StartTransmitting();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StopTransmitting();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioSendConfig.rtp.extensions.back().uri,
+            webrtc::RtpExtension::kAudioLevelUri);
+  ASSERT_TRUE(mAudioConduit->GetSendChannelProxy()
+                  ->mSetSendAudioLevelIndicationStatusEnabled);
+
+  // Contributing sources audio level
+  extensions.clear();
+  extension.uri = webrtc::RtpExtension::kCsrcAudioLevelUri;
+  extensions.emplace_back(extension);
+
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kRecv, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StartReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StopReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.rtp.extensions.back().uri,
+            webrtc::RtpExtension::kCsrcAudioLevelUri);
+  ASSERT_TRUE(mAudioConduit->GetRecvChannelProxy()
+                  ->mSetReceiveCsrcAudioLevelIndicationStatusEnabled);
+
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kSend, extensions);
+  ASSERT_EQ(ec, kMediaConduitMalformedArgument);
+
+  // MId
+  extensions.clear();
+  extension.uri = webrtc::RtpExtension::kMIdUri;
+  extensions.emplace_back(extension);
+
+  // We do not support configuring receiving MId, but do not return an error
+  // in this case.
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kRecv, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+
+  ec = mAudioConduit->SetLocalRTPExtensions(LocalDirection::kSend, extensions);
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StartTransmitting();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StopTransmitting();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioSendConfig.rtp.extensions.back().uri,
+            webrtc::RtpExtension::kMIdUri);
+  ASSERT_TRUE(mAudioConduit->GetSendChannelProxy()->mSetSendMIDStatusEnabled);
+}
+
+TEST_F(AudioConduitTest, TestSetSyncGroup) {
+  MediaConduitErrorCode ec;
+
+  mAudioConduit->SetSyncGroup("test");
+  ec = mAudioConduit->StartReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ec = mAudioConduit->StopReceiving();
+  ASSERT_EQ(ec, kMediaConduitNoError);
+  ASSERT_EQ(mCall->mAudioReceiveConfig.sync_group, "test");
+}
+
+}  // End namespace test.