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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /media/webrtc/signaling/gtest/audioconduit_unittests.cpp | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | media/webrtc/signaling/gtest/audioconduit_unittests.cpp | 781 |
1 files changed, 781 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..af94cd2ba8 --- /dev/null +++ b/media/webrtc/signaling/gtest/audioconduit_unittests.cpp @@ -0,0 +1,781 @@ +/* -*- 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 "ConcreteConduitControl.h" +#include "WaitFor.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()) { + 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; + ConcreteConduitControl 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()->mAudioReceiveConfig->rtp.extensions.empty()); + 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_EQ(Call()->mAudioReceiveConfig->rtp.extensions.back().uri, + webrtc::RtpExtension::kAudioLevelUri); + 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_EQ(Call()->mAudioReceiveConfig->rtp.extensions.back().uri, + webrtc::RtpExtension::kCsrcAudioLevelsUri); + 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_TRUE(Call()->mAudioReceiveConfig->rtp.extensions.empty()); + 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. |