Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 775 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..4068a988f4
--- /dev/null
+++ b/media/webrtc/signaling/gtest/audioconduit_unittests.cpp
@@ -0,0 +1,775 @@
+/* -*- 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 "Canonicals.h"
+
+#include "MockCall.h"
+
+using namespace mozilla;
+using namespace testing;
+using namespace webrtc;
+
+namespace test {
+
+class AudioConduitTest : public ::testing::Test {
+ public:
+  AudioConduitTest()
+      : mCallWrapper(MockCallWrapper::Create()),
+        mAudioConduit(MakeRefPtr<WebrtcAudioConduit>(
+            mCallWrapper, GetCurrentSerialEventTarget())),
+        mControl(GetCurrentSerialEventTarget()) {
+    mControl.Update(
+        [&](auto& aControl) { mAudioConduit->InitControl(&mControl); });
+  }
+
+  ~AudioConduitTest() override {
+    mozilla::Unused << WaitFor(mAudioConduit->Shutdown());
+    mCallWrapper->Destroy();
+  }
+
+  MockCall* Call() { return mCallWrapper->GetMockCall(); }
+
+  const RefPtr<MockCallWrapper> mCallWrapper;
+  const RefPtr<WebrtcAudioConduit> mAudioConduit;
+  ConcreteControl mControl;
+};
+
+TEST_F(AudioConduitTest, TestConfigureSendMediaCodec) {
+  mControl.Update([&](auto& aControl) {
+    // defaults
+    aControl.mAudioSendCodec =
+        Some(AudioCodecConfig(114, "opus", 48000, 2, false));
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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());
+  }
+
+  mControl.Update([&](auto& aControl) {
+    // empty codec name
+    aControl.mAudioSendCodec = Some(AudioCodecConfig(114, "", 48000, 2, false));
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    // Invalid codec was ignored.
+    const webrtc::SdpAudioFormat& f =
+        Call()->mAudioSendConfig->send_codec_spec->format;
+    ASSERT_EQ(f.name, "opus");
+  }
+}
+
+TEST_F(AudioConduitTest, TestConfigureSendOpusMono) {
+  mControl.Update([&](auto& aControl) {
+    // opus mono
+    aControl.mAudioSendCodec =
+        Some(AudioCodecConfig(114, "opus", 48000, 1, false));
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    // opus with inband Forward Error Correction
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, true);
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mMaxPlaybackRate = 1234;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mMaxAverageBitrate = 12345;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mDTXEnabled = true;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mCbrEnabled = true;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mFrameSizeMs = 100;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mMinFrameSizeMs = 201;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    AudioCodecConfig codecConfig =
+        AudioCodecConfig(114, "opus", 48000, 2, false);
+    codecConfig.mMaxFrameSizeMs = 321;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    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;
+    aControl.mAudioSendCodec = Some(codecConfig);
+    aControl.mTransmitting = true;
+  });
+
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    // just default opus stereo
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, false));
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "");
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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());
+  }
+
+  mControl.Update([&](auto& aControl) {
+    // multiple codecs
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(9, "g722", 16000, 2, false));
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, false));
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "");
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 2U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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 =
+        Call()->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");
+  }
+
+  mControl.Update([&](auto& aControl) {
+    // no codecs
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    aControl.mAudioRecvCodecs = codecs;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 0U);
+
+  mControl.Update([&](auto& aControl) {
+    // invalid codec name
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "", 48000, 2, false));
+    aControl.mAudioRecvCodecs = codecs;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 0U);
+
+  mControl.Update([&](auto& aControl) {
+    // invalid number of channels
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 42, false));
+    aControl.mAudioRecvCodecs = codecs;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 0U);
+}
+
+TEST_F(AudioConduitTest, TestConfigureReceiveOpusMono) {
+  mControl.Update([&](auto& aControl) {
+    // opus mono
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 1, false));
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "");
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    // opus mono
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, false));
+    codecs[0].mDTXEnabled = true;
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "");
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  mControl.Update([&](auto& aControl) {
+    // opus with inband Forward Error Correction
+    std::vector<mozilla::AudioCodecConfig> codecs;
+    codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, true));
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "");
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  std::vector<mozilla::AudioCodecConfig> codecs;
+  codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, false));
+
+  mControl.Update([&](auto& aControl) {
+    codecs[0].mMaxPlaybackRate = 0;
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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());
+  }
+
+  mControl.Update([&](auto& aControl) {
+    codecs[0].mMaxPlaybackRate = 8000;
+    aControl.mAudioRecvCodecs = codecs;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  std::vector<mozilla::AudioCodecConfig> codecs;
+  codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, false));
+  mControl.Update([&](auto& aControl) {
+    codecs[0].mMaxAverageBitrate = 0;
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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());
+  }
+
+  mControl.Update([&](auto& aControl) {
+    codecs[0].mMaxAverageBitrate = 8000;
+    aControl.mAudioRecvCodecs = codecs;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  std::vector<mozilla::AudioCodecConfig> codecs;
+  codecs.emplace_back(AudioCodecConfig(114, "opus", 48000, 2, true));
+
+  mControl.Update([&](auto& aControl) {
+    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;
+
+    aControl.mAudioRecvCodecs = codecs;
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->decoder_map.size(), 1U);
+  {
+    const webrtc::SdpAudioFormat& f =
+        Call()->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) {
+  // Empty extensions
+  mControl.Update([&](auto& aControl) {
+    RtpExtList extensions;
+    aControl.mLocalRecvRtpExtensions = extensions;
+    aControl.mReceiving = true;
+    aControl.mLocalSendRtpExtensions = extensions;
+    aControl.mTransmitting = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  ASSERT_TRUE(Call()->mAudioSendConfig->rtp.extensions.empty());
+
+  // Audio level
+  mControl.Update([&](auto& aControl) {
+    RtpExtList extensions;
+    webrtc::RtpExtension extension;
+    extension.uri = webrtc::RtpExtension::kAudioLevelUri;
+    extensions.emplace_back(extension);
+    aControl.mLocalRecvRtpExtensions = extensions;
+    aControl.mLocalSendRtpExtensions = extensions;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  ASSERT_EQ(Call()->mAudioSendConfig->rtp.extensions.back().uri,
+            webrtc::RtpExtension::kAudioLevelUri);
+
+  // Contributing sources audio level
+  mControl.Update([&](auto& aControl) {
+    // We do not support configuring sending csrc-audio-level. It will be
+    // ignored.
+    RtpExtList extensions;
+    webrtc::RtpExtension extension;
+    extension.uri = webrtc::RtpExtension::kCsrcAudioLevelsUri;
+    extensions.emplace_back(extension);
+    aControl.mLocalRecvRtpExtensions = extensions;
+    aControl.mLocalSendRtpExtensions = extensions;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_TRUE(Call()->mAudioSendConfig);
+  ASSERT_TRUE(Call()->mAudioSendConfig->rtp.extensions.empty());
+
+  // Mid
+  mControl.Update([&](auto& aControl) {
+    // We do not support configuring receiving MId. It will be ignored.
+    RtpExtList extensions;
+    webrtc::RtpExtension extension;
+    extension.uri = webrtc::RtpExtension::kMidUri;
+    extensions.emplace_back(extension);
+    aControl.mLocalRecvRtpExtensions = extensions;
+    aControl.mLocalSendRtpExtensions = extensions;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioSendConfig->rtp.extensions.back().uri,
+            webrtc::RtpExtension::kMidUri);
+}
+
+TEST_F(AudioConduitTest, TestSyncGroup) {
+  mControl.Update([&](auto& aControl) {
+    aControl.mSyncGroup = "test";
+    aControl.mReceiving = true;
+  });
+  ASSERT_TRUE(Call()->mAudioReceiveConfig);
+  ASSERT_EQ(Call()->mAudioReceiveConfig->sync_group, "test");
+}
+
+}  // End namespace test.