mirror of
https://github.com/mollyim/webrtc.git
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a136ed4085
Setting the transport cc flag was only possible post creation for audio receive streams, while video receive streams need to be recreated. This CL moves the setter for transport_cc() to where the getter is and adds boiler plate implementations for the video streams. For audio streams this splits "SetUseTransportCcAndNackHistory" into two methods, SetTransportCc and SetNackHistory. Bug: none Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37038}
3889 lines
156 KiB
C++
3889 lines
156 KiB
C++
/*
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* Copyright (c) 2008 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "media/engine/webrtc_voice_engine.h"
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#include <memory>
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#include <utility>
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#include "absl/memory/memory.h"
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#include "absl/strings/match.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/rtp_parameters.h"
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#include "api/scoped_refptr.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/transport/field_trial_based_config.h"
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#include "call/call.h"
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#include "media/base/fake_media_engine.h"
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#include "media/base/fake_network_interface.h"
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#include "media/base/fake_rtp.h"
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#include "media/base/media_constants.h"
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#include "media/engine/fake_webrtc_call.h"
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#include "modules/audio_device/include/mock_audio_device.h"
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#include "modules/audio_processing/include/mock_audio_processing.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/byte_order.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "test/gtest.h"
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#include "test/mock_audio_decoder_factory.h"
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#include "test/mock_audio_encoder_factory.h"
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#include "test/scoped_key_value_config.h"
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using ::testing::_;
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using ::testing::ContainerEq;
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using ::testing::Contains;
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using ::testing::Field;
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using ::testing::Return;
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using ::testing::ReturnPointee;
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using ::testing::SaveArg;
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using ::testing::StrictMock;
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namespace {
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using webrtc::BitrateConstraints;
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constexpr uint32_t kMaxUnsignaledRecvStreams = 4;
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const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1);
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const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
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const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2);
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const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
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const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
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const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
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const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
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const cricket::AudioCodec kRed48000Codec(112, "RED", 48000, 32000, 2);
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const cricket::AudioCodec kTelephoneEventCodec1(106,
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"telephone-event",
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8000,
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0,
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1);
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const cricket::AudioCodec kTelephoneEventCodec2(107,
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"telephone-event",
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32000,
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0,
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1);
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const uint32_t kSsrc0 = 0;
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const uint32_t kSsrc1 = 1;
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const uint32_t kSsrcX = 0x99;
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const uint32_t kSsrcY = 0x17;
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const uint32_t kSsrcZ = 0x42;
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const uint32_t kSsrcW = 0x02;
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const uint32_t kSsrcs4[] = {11, 200, 30, 44};
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constexpr int kRtpHistoryMs = 5000;
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constexpr webrtc::AudioProcessing::Config::GainController1::Mode
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kDefaultAgcMode =
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#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
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webrtc::AudioProcessing::Config::GainController1::kFixedDigital;
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#else
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webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
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#endif
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constexpr webrtc::AudioProcessing::Config::NoiseSuppression::Level
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kDefaultNsLevel =
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webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh;
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void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
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RTC_DCHECK(adm);
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// Setup.
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EXPECT_CALL(*adm, Init()).WillOnce(Return(0));
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EXPECT_CALL(*adm, RegisterAudioCallback(_)).WillOnce(Return(0));
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#if defined(WEBRTC_WIN)
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EXPECT_CALL(
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*adm,
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SetPlayoutDevice(
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::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
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webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
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.WillOnce(Return(0));
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#else
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EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0));
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#endif // #if defined(WEBRTC_WIN)
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EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0));
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EXPECT_CALL(*adm, StereoPlayoutIsAvailable(::testing::_)).WillOnce(Return(0));
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EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0));
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#if defined(WEBRTC_WIN)
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EXPECT_CALL(
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*adm,
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SetRecordingDevice(
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::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
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webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
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.WillOnce(Return(0));
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#else
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EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
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#endif // #if defined(WEBRTC_WIN)
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EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0));
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EXPECT_CALL(*adm, StereoRecordingIsAvailable(::testing::_))
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.WillOnce(Return(0));
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EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0));
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EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
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EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
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EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
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// Teardown.
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EXPECT_CALL(*adm, StopPlayout()).WillOnce(Return(0));
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EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0));
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EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0));
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EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0));
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}
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} // namespace
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// Tests that our stub library "works".
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TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
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for (bool use_null_apm : {false, true}) {
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std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
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webrtc::CreateDefaultTaskQueueFactory();
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rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
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webrtc::test::MockAudioDeviceModule::CreateStrict();
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AdmSetupExpectations(adm.get());
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rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm =
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use_null_apm ? nullptr
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: rtc::make_ref_counted<
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StrictMock<webrtc::test::MockAudioProcessing>>();
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webrtc::AudioProcessing::Config apm_config;
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if (!use_null_apm) {
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EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config));
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EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config));
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EXPECT_CALL(*apm, DetachAecDump());
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}
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{
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webrtc::FieldTrialBasedConfig trials;
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cricket::WebRtcVoiceEngine engine(
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task_queue_factory.get(), adm.get(),
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webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
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webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
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nullptr, trials);
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engine.Init();
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}
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}
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}
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class FakeAudioSink : public webrtc::AudioSinkInterface {
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public:
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void OnData(const Data& audio) override {}
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};
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class FakeAudioSource : public cricket::AudioSource {
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void SetSink(Sink* sink) override {}
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};
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class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
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public:
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WebRtcVoiceEngineTestFake()
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: use_null_apm_(GetParam()),
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task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
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adm_(webrtc::test::MockAudioDeviceModule::CreateStrict()),
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apm_(use_null_apm_
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? nullptr
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: rtc::make_ref_counted<
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StrictMock<webrtc::test::MockAudioProcessing>>()),
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call_(&field_trials_) {
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// AudioDeviceModule.
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AdmSetupExpectations(adm_.get());
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if (!use_null_apm_) {
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// AudioProcessing.
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EXPECT_CALL(*apm_, GetConfig())
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.WillRepeatedly(ReturnPointee(&apm_config_));
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EXPECT_CALL(*apm_, ApplyConfig(_))
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.WillRepeatedly(SaveArg<0>(&apm_config_));
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EXPECT_CALL(*apm_, DetachAecDump());
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}
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// Default Options.
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// TODO(kwiberg): We should use mock factories here, but a bunch of
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// the tests here probe the specific set of codecs provided by the builtin
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// factories. Those tests should probably be moved elsewhere.
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auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
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auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
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engine_.reset(new cricket::WebRtcVoiceEngine(
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task_queue_factory_.get(), adm_.get(), encoder_factory, decoder_factory,
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nullptr, apm_, nullptr, field_trials_));
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engine_->Init();
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send_parameters_.codecs.push_back(kPcmuCodec);
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recv_parameters_.codecs.push_back(kPcmuCodec);
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if (!use_null_apm_) {
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// Default Options.
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VerifyEchoCancellationSettings(/*enabled=*/true);
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EXPECT_TRUE(IsHighPassFilterEnabled());
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EXPECT_TRUE(apm_config_.noise_suppression.enabled);
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EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
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VerifyGainControlEnabledCorrectly();
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VerifyGainControlDefaultSettings();
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}
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}
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bool SetupChannel() {
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channel_ = engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
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cricket::AudioOptions(),
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webrtc::CryptoOptions());
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return (channel_ != nullptr);
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}
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bool SetupRecvStream() {
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if (!SetupChannel()) {
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return false;
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}
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return AddRecvStream(kSsrcX);
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}
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bool SetupSendStream() {
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return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX));
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}
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bool SetupSendStream(const cricket::StreamParams& sp) {
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if (!SetupChannel()) {
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return false;
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}
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if (!channel_->AddSendStream(sp)) {
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return false;
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}
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if (!use_null_apm_) {
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EXPECT_CALL(*apm_, set_output_will_be_muted(false));
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}
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return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_);
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}
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bool AddRecvStream(uint32_t ssrc) {
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EXPECT_TRUE(channel_);
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return channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc));
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}
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void SetupForMultiSendStream() {
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EXPECT_TRUE(SetupSendStream());
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// Remove stream added in Setup.
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EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
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EXPECT_TRUE(channel_->RemoveSendStream(kSsrcX));
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// Verify the channel does not exist.
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EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX));
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}
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void DeliverPacket(const void* data, int len) {
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rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
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channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
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rtc::Thread::Current()->ProcessMessages(0);
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}
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void TearDown() override { delete channel_; }
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const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
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const auto* send_stream = call_.GetAudioSendStream(ssrc);
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EXPECT_TRUE(send_stream);
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return *send_stream;
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}
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const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) {
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const auto* recv_stream = call_.GetAudioReceiveStream(ssrc);
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EXPECT_TRUE(recv_stream);
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return *recv_stream;
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}
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const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) {
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return GetSendStream(ssrc).GetConfig();
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}
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const webrtc::AudioReceiveStreamInterface::Config& GetRecvStreamConfig(
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uint32_t ssrc) {
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return GetRecvStream(ssrc).GetConfig();
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}
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void SetSend(bool enable) {
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ASSERT_TRUE(channel_);
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if (enable) {
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EXPECT_CALL(*adm_, RecordingIsInitialized())
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.Times(::testing::AtMost(1))
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.WillOnce(Return(false));
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EXPECT_CALL(*adm_, Recording())
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.Times(::testing::AtMost(1))
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.WillOnce(Return(false));
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EXPECT_CALL(*adm_, InitRecording())
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.Times(::testing::AtMost(1))
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.WillOnce(Return(0));
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}
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channel_->SetSend(enable);
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}
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void SetSendParameters(const cricket::AudioSendParameters& params) {
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ASSERT_TRUE(channel_);
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EXPECT_TRUE(channel_->SetSendParameters(params));
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}
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void SetAudioSend(uint32_t ssrc,
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bool enable,
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cricket::AudioSource* source,
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const cricket::AudioOptions* options = nullptr) {
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ASSERT_TRUE(channel_);
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if (!use_null_apm_) {
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EXPECT_CALL(*apm_, set_output_will_be_muted(!enable));
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}
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EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source));
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}
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void TestInsertDtmf(uint32_t ssrc,
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bool caller,
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const cricket::AudioCodec& codec) {
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EXPECT_TRUE(SetupChannel());
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if (caller) {
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// If this is a caller, local description will be applied and add the
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// send stream.
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EXPECT_TRUE(
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channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
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}
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// Test we can only InsertDtmf when the other side supports telephone-event.
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SetSendParameters(send_parameters_);
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SetSend(true);
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EXPECT_FALSE(channel_->CanInsertDtmf());
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EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
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send_parameters_.codecs.push_back(codec);
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SetSendParameters(send_parameters_);
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EXPECT_TRUE(channel_->CanInsertDtmf());
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if (!caller) {
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// If this is callee, there's no active send channel yet.
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EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
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EXPECT_TRUE(
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channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
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}
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// Check we fail if the ssrc is invalid.
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EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111));
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// Test send.
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cricket::FakeAudioSendStream::TelephoneEvent telephone_event =
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GetSendStream(kSsrcX).GetLatestTelephoneEvent();
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EXPECT_EQ(-1, telephone_event.payload_type);
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EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
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telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent();
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EXPECT_EQ(codec.id, telephone_event.payload_type);
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EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency);
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EXPECT_EQ(2, telephone_event.event_code);
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EXPECT_EQ(123, telephone_event.duration_ms);
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}
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void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
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// For a caller, the answer will be applied in set remote description
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// where SetSendParameters() is called.
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EXPECT_TRUE(SetupChannel());
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EXPECT_TRUE(
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channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
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send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
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SetSendParameters(send_parameters_);
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const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
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EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
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}
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void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
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// For a callee, the answer will be applied in set local description
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// where SetExtmapAllowMixed() and AddSendStream() are called.
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EXPECT_TRUE(SetupChannel());
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channel_->SetExtmapAllowMixed(extmap_allow_mixed);
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EXPECT_TRUE(
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channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
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const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
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EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
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}
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// Test that send bandwidth is set correctly.
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// `codec` is the codec under test.
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// `max_bitrate` is a parameter to set to SetMaxSendBandwidth().
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// `expected_result` is the expected result from SetMaxSendBandwidth().
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// `expected_bitrate` is the expected audio bitrate afterward.
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void TestMaxSendBandwidth(const cricket::AudioCodec& codec,
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int max_bitrate,
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bool expected_result,
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int expected_bitrate) {
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cricket::AudioSendParameters parameters;
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parameters.codecs.push_back(codec);
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parameters.max_bandwidth_bps = max_bitrate;
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if (expected_result) {
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SetSendParameters(parameters);
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} else {
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EXPECT_FALSE(channel_->SetSendParameters(parameters));
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}
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EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX));
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}
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// Sets the per-stream maximum bitrate limit for the specified SSRC.
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bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) {
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webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc);
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EXPECT_EQ(1UL, parameters.encodings.size());
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parameters.encodings[0].max_bitrate_bps = bitrate;
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return channel_->SetRtpSendParameters(ssrc, parameters).ok();
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}
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void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
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cricket::AudioSendParameters send_parameters;
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send_parameters.codecs.push_back(codec);
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send_parameters.max_bandwidth_bps = bitrate;
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SetSendParameters(send_parameters);
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}
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void CheckSendCodecBitrate(int32_t ssrc,
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const char expected_name[],
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int expected_bitrate) {
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const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec;
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|
EXPECT_EQ(expected_name, spec->format.name);
|
|
EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps);
|
|
}
|
|
|
|
absl::optional<int> GetCodecBitrate(int32_t ssrc) {
|
|
return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;
|
|
}
|
|
|
|
const absl::optional<std::string>& GetAudioNetworkAdaptorConfig(
|
|
int32_t ssrc) {
|
|
return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
|
|
}
|
|
|
|
void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec,
|
|
int global_max,
|
|
int stream_max,
|
|
bool expected_result,
|
|
int expected_codec_bitrate) {
|
|
// Clear the bitrate limit from the previous test case.
|
|
EXPECT_TRUE(SetMaxBitrateForStream(kSsrcX, -1));
|
|
|
|
// Attempt to set the requested bitrate limits.
|
|
SetGlobalMaxBitrate(codec, global_max);
|
|
EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrcX, stream_max));
|
|
|
|
// Verify that reading back the parameters gives results
|
|
// consistent with the Set() result.
|
|
webrtc::RtpParameters resulting_parameters =
|
|
channel_->GetRtpSendParameters(kSsrcX);
|
|
EXPECT_EQ(1UL, resulting_parameters.encodings.size());
|
|
EXPECT_EQ(expected_result ? stream_max : -1,
|
|
resulting_parameters.encodings[0].max_bitrate_bps);
|
|
|
|
// Verify that the codec settings have the expected bitrate.
|
|
EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX));
|
|
}
|
|
|
|
void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps,
|
|
int expected_min_bitrate_bps,
|
|
const char* start_bitrate_kbps,
|
|
int expected_start_bitrate_bps,
|
|
const char* max_bitrate_kbps,
|
|
int expected_max_bitrate_bps) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
auto& codecs = send_parameters_.codecs;
|
|
codecs.clear();
|
|
codecs.push_back(kOpusCodec);
|
|
codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps;
|
|
codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps;
|
|
codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps;
|
|
EXPECT_CALL(*call_.GetMockTransportControllerSend(),
|
|
SetSdpBitrateParameters(
|
|
AllOf(Field(&BitrateConstraints::min_bitrate_bps,
|
|
expected_min_bitrate_bps),
|
|
Field(&BitrateConstraints::start_bitrate_bps,
|
|
expected_start_bitrate_bps),
|
|
Field(&BitrateConstraints::max_bitrate_bps,
|
|
expected_max_bitrate_bps))));
|
|
|
|
SetSendParameters(send_parameters_);
|
|
}
|
|
|
|
void TestSetSendRtpHeaderExtensions(const std::string& ext) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// Ensure extensions are off by default.
|
|
EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure unknown extensions won't cause an error.
|
|
send_parameters_.extensions.push_back(
|
|
webrtc::RtpExtension("urn:ietf:params:unknownextention", 1));
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure extensions stay off with an empty list of headers.
|
|
send_parameters_.extensions.clear();
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure extension is set properly.
|
|
const int id = 1;
|
|
send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id));
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
|
|
EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri);
|
|
EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id);
|
|
|
|
// Ensure extension is set properly on new stream.
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
|
|
EXPECT_NE(call_.GetAudioSendStream(kSsrcX),
|
|
call_.GetAudioSendStream(kSsrcY));
|
|
EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size());
|
|
EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri);
|
|
EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id);
|
|
|
|
// Ensure all extensions go back off with an empty list.
|
|
send_parameters_.codecs.push_back(kPcmuCodec);
|
|
send_parameters_.extensions.clear();
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
|
|
EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size());
|
|
}
|
|
|
|
void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
|
|
// Ensure extensions are off by default.
|
|
EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure unknown extensions won't cause an error.
|
|
recv_parameters_.extensions.push_back(
|
|
webrtc::RtpExtension("urn:ietf:params:unknownextention", 1));
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure extensions stay off with an empty list of headers.
|
|
recv_parameters_.extensions.clear();
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
|
|
|
|
// Ensure extension is set properly.
|
|
const int id = 2;
|
|
recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id));
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
|
|
EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].uri);
|
|
EXPECT_EQ(id, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].id);
|
|
|
|
// Ensure extension is set properly on new stream.
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_NE(call_.GetAudioReceiveStream(kSsrcX),
|
|
call_.GetAudioReceiveStream(kSsrcY));
|
|
EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size());
|
|
EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].uri);
|
|
EXPECT_EQ(id, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].id);
|
|
|
|
// Ensure all extensions go back off with an empty list.
|
|
recv_parameters_.extensions.clear();
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
|
|
EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size());
|
|
}
|
|
|
|
webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
|
|
webrtc::AudioSendStream::Stats stats;
|
|
stats.local_ssrc = 12;
|
|
stats.payload_bytes_sent = 345;
|
|
stats.header_and_padding_bytes_sent = 56;
|
|
stats.packets_sent = 678;
|
|
stats.packets_lost = 9012;
|
|
stats.fraction_lost = 34.56f;
|
|
stats.codec_name = "codec_name_send";
|
|
stats.codec_payload_type = 42;
|
|
stats.jitter_ms = 12;
|
|
stats.rtt_ms = 345;
|
|
stats.audio_level = 678;
|
|
stats.apm_statistics.delay_median_ms = 234;
|
|
stats.apm_statistics.delay_standard_deviation_ms = 567;
|
|
stats.apm_statistics.echo_return_loss = 890;
|
|
stats.apm_statistics.echo_return_loss_enhancement = 1234;
|
|
stats.apm_statistics.residual_echo_likelihood = 0.432f;
|
|
stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f;
|
|
stats.ana_statistics.bitrate_action_counter = 321;
|
|
stats.ana_statistics.channel_action_counter = 432;
|
|
stats.ana_statistics.dtx_action_counter = 543;
|
|
stats.ana_statistics.fec_action_counter = 654;
|
|
stats.ana_statistics.frame_length_increase_counter = 765;
|
|
stats.ana_statistics.frame_length_decrease_counter = 876;
|
|
stats.ana_statistics.uplink_packet_loss_fraction = 987.0;
|
|
return stats;
|
|
}
|
|
void SetAudioSendStreamStats() {
|
|
for (auto* s : call_.GetAudioSendStreams()) {
|
|
s->SetStats(GetAudioSendStreamStats());
|
|
}
|
|
}
|
|
void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info,
|
|
bool is_sending) {
|
|
const auto stats = GetAudioSendStreamStats();
|
|
EXPECT_EQ(info.ssrc(), stats.local_ssrc);
|
|
EXPECT_EQ(info.payload_bytes_sent, stats.payload_bytes_sent);
|
|
EXPECT_EQ(info.header_and_padding_bytes_sent,
|
|
stats.header_and_padding_bytes_sent);
|
|
EXPECT_EQ(info.packets_sent, stats.packets_sent);
|
|
EXPECT_EQ(info.packets_lost, stats.packets_lost);
|
|
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
|
|
EXPECT_EQ(info.codec_name, stats.codec_name);
|
|
EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
|
|
EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
|
|
EXPECT_EQ(info.rtt_ms, stats.rtt_ms);
|
|
EXPECT_EQ(info.audio_level, stats.audio_level);
|
|
EXPECT_EQ(info.apm_statistics.delay_median_ms,
|
|
stats.apm_statistics.delay_median_ms);
|
|
EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms,
|
|
stats.apm_statistics.delay_standard_deviation_ms);
|
|
EXPECT_EQ(info.apm_statistics.echo_return_loss,
|
|
stats.apm_statistics.echo_return_loss);
|
|
EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement,
|
|
stats.apm_statistics.echo_return_loss_enhancement);
|
|
EXPECT_EQ(info.apm_statistics.residual_echo_likelihood,
|
|
stats.apm_statistics.residual_echo_likelihood);
|
|
EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max,
|
|
stats.apm_statistics.residual_echo_likelihood_recent_max);
|
|
EXPECT_EQ(info.ana_statistics.bitrate_action_counter,
|
|
stats.ana_statistics.bitrate_action_counter);
|
|
EXPECT_EQ(info.ana_statistics.channel_action_counter,
|
|
stats.ana_statistics.channel_action_counter);
|
|
EXPECT_EQ(info.ana_statistics.dtx_action_counter,
|
|
stats.ana_statistics.dtx_action_counter);
|
|
EXPECT_EQ(info.ana_statistics.fec_action_counter,
|
|
stats.ana_statistics.fec_action_counter);
|
|
EXPECT_EQ(info.ana_statistics.frame_length_increase_counter,
|
|
stats.ana_statistics.frame_length_increase_counter);
|
|
EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter,
|
|
stats.ana_statistics.frame_length_decrease_counter);
|
|
EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction,
|
|
stats.ana_statistics.uplink_packet_loss_fraction);
|
|
}
|
|
|
|
webrtc::AudioReceiveStreamInterface::Stats GetAudioReceiveStreamStats()
|
|
const {
|
|
webrtc::AudioReceiveStreamInterface::Stats stats;
|
|
stats.remote_ssrc = 123;
|
|
stats.payload_bytes_rcvd = 456;
|
|
stats.header_and_padding_bytes_rcvd = 67;
|
|
stats.packets_rcvd = 768;
|
|
stats.packets_lost = 101;
|
|
stats.codec_name = "codec_name_recv";
|
|
stats.codec_payload_type = 42;
|
|
stats.jitter_ms = 901;
|
|
stats.jitter_buffer_ms = 234;
|
|
stats.jitter_buffer_preferred_ms = 567;
|
|
stats.delay_estimate_ms = 890;
|
|
stats.audio_level = 1234;
|
|
stats.total_samples_received = 5678901;
|
|
stats.concealed_samples = 234;
|
|
stats.concealment_events = 12;
|
|
stats.jitter_buffer_delay_seconds = 34;
|
|
stats.jitter_buffer_emitted_count = 77;
|
|
stats.expand_rate = 5.67f;
|
|
stats.speech_expand_rate = 8.90f;
|
|
stats.secondary_decoded_rate = 1.23f;
|
|
stats.secondary_discarded_rate = 0.12f;
|
|
stats.accelerate_rate = 4.56f;
|
|
stats.preemptive_expand_rate = 7.89f;
|
|
stats.decoding_calls_to_silence_generator = 12;
|
|
stats.decoding_calls_to_neteq = 345;
|
|
stats.decoding_normal = 67890;
|
|
stats.decoding_plc = 1234;
|
|
stats.decoding_codec_plc = 1236;
|
|
stats.decoding_cng = 5678;
|
|
stats.decoding_plc_cng = 9012;
|
|
stats.decoding_muted_output = 3456;
|
|
stats.capture_start_ntp_time_ms = 7890;
|
|
return stats;
|
|
}
|
|
void SetAudioReceiveStreamStats() {
|
|
for (auto* s : call_.GetAudioReceiveStreams()) {
|
|
s->SetStats(GetAudioReceiveStreamStats());
|
|
}
|
|
}
|
|
void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) {
|
|
const auto stats = GetAudioReceiveStreamStats();
|
|
EXPECT_EQ(info.ssrc(), stats.remote_ssrc);
|
|
EXPECT_EQ(info.payload_bytes_rcvd, stats.payload_bytes_rcvd);
|
|
EXPECT_EQ(info.header_and_padding_bytes_rcvd,
|
|
stats.header_and_padding_bytes_rcvd);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd),
|
|
stats.packets_rcvd);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),
|
|
stats.packets_lost);
|
|
EXPECT_EQ(info.codec_name, stats.codec_name);
|
|
EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms),
|
|
stats.jitter_buffer_ms);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_preferred_ms),
|
|
stats.jitter_buffer_preferred_ms);
|
|
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.delay_estimate_ms),
|
|
stats.delay_estimate_ms);
|
|
EXPECT_EQ(info.audio_level, stats.audio_level);
|
|
EXPECT_EQ(info.total_samples_received, stats.total_samples_received);
|
|
EXPECT_EQ(info.concealed_samples, stats.concealed_samples);
|
|
EXPECT_EQ(info.concealment_events, stats.concealment_events);
|
|
EXPECT_EQ(info.jitter_buffer_delay_seconds,
|
|
stats.jitter_buffer_delay_seconds);
|
|
EXPECT_EQ(info.jitter_buffer_emitted_count,
|
|
stats.jitter_buffer_emitted_count);
|
|
EXPECT_EQ(info.expand_rate, stats.expand_rate);
|
|
EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate);
|
|
EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate);
|
|
EXPECT_EQ(info.secondary_discarded_rate, stats.secondary_discarded_rate);
|
|
EXPECT_EQ(info.accelerate_rate, stats.accelerate_rate);
|
|
EXPECT_EQ(info.preemptive_expand_rate, stats.preemptive_expand_rate);
|
|
EXPECT_EQ(info.decoding_calls_to_silence_generator,
|
|
stats.decoding_calls_to_silence_generator);
|
|
EXPECT_EQ(info.decoding_calls_to_neteq, stats.decoding_calls_to_neteq);
|
|
EXPECT_EQ(info.decoding_normal, stats.decoding_normal);
|
|
EXPECT_EQ(info.decoding_plc, stats.decoding_plc);
|
|
EXPECT_EQ(info.decoding_codec_plc, stats.decoding_codec_plc);
|
|
EXPECT_EQ(info.decoding_cng, stats.decoding_cng);
|
|
EXPECT_EQ(info.decoding_plc_cng, stats.decoding_plc_cng);
|
|
EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output);
|
|
EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms);
|
|
}
|
|
void VerifyVoiceSendRecvCodecs(const cricket::VoiceMediaInfo& info) const {
|
|
EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size());
|
|
for (const cricket::AudioCodec& codec : send_parameters_.codecs) {
|
|
ASSERT_EQ(info.send_codecs.count(codec.id), 1U);
|
|
EXPECT_EQ(info.send_codecs.find(codec.id)->second,
|
|
codec.ToCodecParameters());
|
|
}
|
|
EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size());
|
|
for (const cricket::AudioCodec& codec : recv_parameters_.codecs) {
|
|
ASSERT_EQ(info.receive_codecs.count(codec.id), 1U);
|
|
EXPECT_EQ(info.receive_codecs.find(codec.id)->second,
|
|
codec.ToCodecParameters());
|
|
}
|
|
}
|
|
|
|
void VerifyGainControlEnabledCorrectly() {
|
|
EXPECT_TRUE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_EQ(kDefaultAgcMode, apm_config_.gain_controller1.mode);
|
|
}
|
|
|
|
void VerifyGainControlDefaultSettings() {
|
|
EXPECT_EQ(3, apm_config_.gain_controller1.target_level_dbfs);
|
|
EXPECT_EQ(9, apm_config_.gain_controller1.compression_gain_db);
|
|
EXPECT_TRUE(apm_config_.gain_controller1.enable_limiter);
|
|
}
|
|
|
|
void VerifyEchoCancellationSettings(bool enabled) {
|
|
constexpr bool kDefaultUseAecm =
|
|
#if defined(WEBRTC_ANDROID)
|
|
true;
|
|
#else
|
|
false;
|
|
#endif
|
|
EXPECT_EQ(apm_config_.echo_canceller.enabled, enabled);
|
|
EXPECT_EQ(apm_config_.echo_canceller.mobile_mode, kDefaultUseAecm);
|
|
}
|
|
|
|
bool IsHighPassFilterEnabled() {
|
|
return apm_config_.high_pass_filter.enabled;
|
|
}
|
|
|
|
protected:
|
|
rtc::AutoThread main_thread_;
|
|
const bool use_null_apm_;
|
|
webrtc::test::ScopedKeyValueConfig field_trials_;
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm_;
|
|
rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_;
|
|
cricket::FakeCall call_;
|
|
std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
|
|
cricket::VoiceMediaChannel* channel_ = nullptr;
|
|
cricket::AudioSendParameters send_parameters_;
|
|
cricket::AudioRecvParameters recv_parameters_;
|
|
FakeAudioSource fake_source_;
|
|
webrtc::AudioProcessing::Config apm_config_;
|
|
};
|
|
|
|
INSTANTIATE_TEST_SUITE_P(TestBothWithAndWithoutNullApm,
|
|
WebRtcVoiceEngineTestFake,
|
|
::testing::Values(false, true));
|
|
|
|
// Tests that we can create and destroy a channel.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CreateMediaChannel) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
}
|
|
|
|
// Test that we can add a send stream and that it has the correct defaults.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CreateSendStream) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
|
|
const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
|
|
EXPECT_EQ(kSsrcX, config.rtp.ssrc);
|
|
EXPECT_EQ("", config.rtp.c_name);
|
|
EXPECT_EQ(0u, config.rtp.extensions.size());
|
|
EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
|
|
config.send_transport);
|
|
}
|
|
|
|
// Test that we can add a receive stream and that it has the correct defaults.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
const webrtc::AudioReceiveStreamInterface::Config& config =
|
|
GetRecvStreamConfig(kSsrcX);
|
|
EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc);
|
|
EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
|
|
EXPECT_FALSE(config.rtp.transport_cc);
|
|
EXPECT_EQ(0u, config.rtp.extensions.size());
|
|
EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
|
|
config.rtcp_send_transport);
|
|
EXPECT_EQ("", config.sync_group);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
|
|
const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
|
|
bool opus_found = false;
|
|
for (const cricket::AudioCodec& codec : codecs) {
|
|
if (codec.name == "opus") {
|
|
EXPECT_TRUE(HasTransportCc(codec));
|
|
opus_found = true;
|
|
}
|
|
}
|
|
EXPECT_TRUE(opus_found);
|
|
}
|
|
|
|
// Test that we set our inbound codecs properly, including changing PT.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs.push_back(kTelephoneEventCodec2);
|
|
parameters.codecs[0].id = 106; // collide with existing CN 32k
|
|
parameters.codecs[2].id = 126;
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}},
|
|
{106, {"ISAC", 16000, 1}},
|
|
{126, {"telephone-event", 8000, 1}},
|
|
{107, {"telephone-event", 32000, 1}}})));
|
|
}
|
|
|
|
// Test that we fail to set an unknown inbound codec.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1));
|
|
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
|
|
}
|
|
|
|
// Test that we fail if we have duplicate types in the inbound list.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs[1].id = kIsacCodec.id;
|
|
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
|
|
}
|
|
|
|
// Test that we can decode OPUS without stereo parameters.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}},
|
|
{103, {"ISAC", 16000, 1}},
|
|
{111, {"opus", 48000, 2}}})));
|
|
}
|
|
|
|
// Test that we can decode OPUS with stereo = 0.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[2].params["stereo"] = "0";
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}},
|
|
{103, {"ISAC", 16000, 1}},
|
|
{111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
|
|
}
|
|
|
|
// Test that we can decode OPUS with stereo = 1.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[2].params["stereo"] = "1";
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}},
|
|
{103, {"ISAC", 16000, 1}},
|
|
{111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
|
|
}
|
|
|
|
// Test that changes to recv codecs are applied to all streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs.push_back(kTelephoneEventCodec2);
|
|
parameters.codecs[0].id = 106; // collide with existing CN 32k
|
|
parameters.codecs[2].id = 126;
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
for (const auto& ssrc : {kSsrcX, kSsrcY}) {
|
|
EXPECT_TRUE(AddRecvStream(ssrc));
|
|
EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}},
|
|
{106, {"ISAC", 16000, 1}},
|
|
{126, {"telephone-event", 8000, 1}},
|
|
{107, {"telephone-event", 32000, 1}}})));
|
|
}
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs[0].id = 106; // collide with existing CN 32k
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map;
|
|
ASSERT_EQ(1u, dm.count(106));
|
|
EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106));
|
|
}
|
|
|
|
// Test that we can apply the same set of codecs again while playing.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
channel_->SetPlayout(true);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
// Remapping a payload type to a different codec should fail.
|
|
parameters.codecs[0] = kOpusCodec;
|
|
parameters.codecs[0].id = kIsacCodec.id;
|
|
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(GetRecvStream(kSsrcX).started());
|
|
}
|
|
|
|
// Test that we can add a codec while playing.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
channel_->SetPlayout(true);
|
|
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(GetRecvStream(kSsrcX).started());
|
|
}
|
|
|
|
// Test that we accept adding the same codec with a different payload type.
|
|
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847
|
|
TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
++parameters.codecs[0].id;
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
}
|
|
|
|
// Test that we do allow setting Opus/Red by default.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs[1].params[""] = "111/111";
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{111, {"opus", 48000, 2}},
|
|
{112, {"red", 48000, 2, {{"", "111/111"}}}}})));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// Test that when autobw is enabled, bitrate is kept as the default
|
|
// value. autobw is enabled for the following tests because the target
|
|
// bitrate is <= 0.
|
|
|
|
// ISAC, default bitrate == 32000.
|
|
TestMaxSendBandwidth(kIsacCodec, 0, true, 32000);
|
|
|
|
// PCMU, default bitrate == 64000.
|
|
TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000);
|
|
|
|
// opus, default bitrate == 32000 in mono.
|
|
TestMaxSendBandwidth(kOpusCodec, -1, true, 32000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// ISAC, default bitrate == 32000.
|
|
TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000);
|
|
// Rates above the max (56000) should be capped.
|
|
TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000);
|
|
|
|
// opus, default bitrate == 64000.
|
|
TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000);
|
|
TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000);
|
|
// Rates above the max (510000) should be capped.
|
|
TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// Test that we can only set a maximum bitrate for a fixed-rate codec
|
|
// if it's bigger than the fixed rate.
|
|
|
|
// PCMU, fixed bitrate == 64000.
|
|
TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000);
|
|
TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
const int kDesiredBitrate = 128000;
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs = engine_->send_codecs();
|
|
parameters.max_bandwidth_bps = kDesiredBitrate;
|
|
SetSendParameters(parameters);
|
|
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
|
|
|
|
EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX));
|
|
}
|
|
|
|
// Test that bitrate cannot be set for CBR codecs.
|
|
// Bitrate is ignored if it is higher than the fixed bitrate.
|
|
// Bitrate less then the fixed bitrate is an error.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// PCMU, default bitrate == 64000.
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
|
|
|
|
send_parameters_.max_bandwidth_bps = 128000;
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
|
|
|
|
send_parameters_.max_bandwidth_bps = 128;
|
|
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
|
|
EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
|
|
}
|
|
|
|
// Test that the per-stream bitrate limit and the global
|
|
// bitrate limit both apply.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// opus, default bitrate == 32000.
|
|
SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000);
|
|
SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000);
|
|
SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000);
|
|
SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000);
|
|
|
|
// CBR codecs allow both maximums to exceed the bitrate.
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000);
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000);
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000);
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000);
|
|
|
|
// CBR codecs don't allow per stream maximums to be too low.
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000);
|
|
SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000);
|
|
}
|
|
|
|
// Test that an attempt to set RtpParameters for a stream that does not exist
|
|
// fails.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
webrtc::RtpParameters nonexistent_parameters =
|
|
channel_->GetRtpSendParameters(kSsrcX);
|
|
EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
|
|
|
|
nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
|
|
EXPECT_FALSE(
|
|
channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
|
|
// This test verifies that setting RtpParameters succeeds only if
|
|
// the structure contains exactly one encoding.
|
|
// TODO(skvlad): Update this test when we start supporting setting parameters
|
|
// for each encoding individually.
|
|
|
|
EXPECT_TRUE(SetupSendStream());
|
|
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
// Two or more encodings should result in failure.
|
|
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
|
|
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
// Zero encodings should also fail.
|
|
parameters.encodings.clear();
|
|
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
}
|
|
|
|
// Changing the SSRC through RtpParameters is not allowed.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
parameters.encodings[0].ssrc = 0xdeadbeef;
|
|
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
}
|
|
|
|
// Test that a stream will not be sending if its encoding is made
|
|
// inactive through SetRtpSendParameters.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSend(true);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
// Get current parameters and change "active" to false.
|
|
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
ASSERT_EQ(1u, parameters.encodings.size());
|
|
ASSERT_TRUE(parameters.encodings[0].active);
|
|
parameters.encodings[0].active = false;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Now change it back to active and verify we resume sending.
|
|
// This should occur even when other parameters are updated.
|
|
parameters.encodings[0].active = true;
|
|
parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000);
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRtpParametersAdaptivePtime) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
// Get current parameters and change "adaptive_ptime" to true.
|
|
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
ASSERT_EQ(1u, parameters.encodings.size());
|
|
ASSERT_FALSE(parameters.encodings[0].adaptive_ptime);
|
|
parameters.encodings[0].adaptive_ptime = true;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
EXPECT_EQ(16000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
|
|
|
|
parameters.encodings[0].adaptive_ptime = false;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
EXPECT_FALSE(GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
EXPECT_EQ(32000, GetSendStreamConfig(kSsrcX).min_bitrate_bps);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
DisablingAdaptivePtimeDoesNotRemoveAudioNetworkAdaptorFromOptions) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
send_parameters_.options.audio_network_adaptor = true;
|
|
send_parameters_.options.audio_network_adaptor_config = {"1234"};
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
|
|
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
parameters.encodings[0].adaptive_ptime = false;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, AdaptivePtimeFieldTrial) {
|
|
webrtc::test::ScopedKeyValueConfig override_field_trials(
|
|
field_trials_, "WebRTC-Audio-AdaptivePtime/enabled:true/");
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
}
|
|
|
|
// Test that SetRtpSendParameters configures the correct encoding channel for
|
|
// each SSRC.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
|
|
SetupForMultiSendStream();
|
|
// Create send streams.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
}
|
|
// Configure one stream to be limited by the stream config, another to be
|
|
// limited by the global max, and the third one with no per-stream limit
|
|
// (still subject to the global limit).
|
|
SetGlobalMaxBitrate(kOpusCodec, 32000);
|
|
EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000));
|
|
EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000));
|
|
EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1));
|
|
|
|
EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
|
|
EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1]));
|
|
EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
|
|
|
|
// Remove the global cap; the streams should switch to their respective
|
|
// maximums (or remain unchanged if there was no other limit on them.)
|
|
SetGlobalMaxBitrate(kOpusCodec, -1);
|
|
EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
|
|
EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1]));
|
|
EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
|
|
}
|
|
|
|
// Test that GetRtpSendParameters returns the currently configured codecs.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
SetSendParameters(parameters);
|
|
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
ASSERT_EQ(2u, rtp_parameters.codecs.size());
|
|
EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
|
|
EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
|
|
}
|
|
|
|
// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
|
|
cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX);
|
|
params.cname = "rtcpcname";
|
|
EXPECT_TRUE(SetupSendStream(params));
|
|
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
DetectRtpSendParameterHeaderExtensionsChange) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
rtp_parameters.header_extensions.emplace_back();
|
|
|
|
EXPECT_NE(0u, rtp_parameters.header_extensions.size());
|
|
|
|
webrtc::RTCError result =
|
|
channel_->SetRtpSendParameters(kSsrcX, rtp_parameters);
|
|
EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
|
|
}
|
|
|
|
// Test that GetRtpSendParameters returns an SSRC.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
ASSERT_EQ(1u, rtp_parameters.encodings.size());
|
|
EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc);
|
|
}
|
|
|
|
// Test that if we set/get parameters multiple times, we get the same results.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
SetSendParameters(parameters);
|
|
|
|
webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX);
|
|
|
|
// We should be able to set the params we just got.
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params).ok());
|
|
|
|
// ... And this shouldn't change the params returned by GetRtpSendParameters.
|
|
webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX);
|
|
EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX));
|
|
}
|
|
|
|
// Test that max_bitrate_bps in send stream config gets updated correctly when
|
|
// SetRtpSendParameters is called.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters send_parameters;
|
|
send_parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(send_parameters);
|
|
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
// Expect empty on parameters.encodings[0].max_bitrate_bps;
|
|
EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps);
|
|
|
|
constexpr int kMaxBitrateBps = 6000;
|
|
rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
|
|
|
const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
|
|
EXPECT_EQ(max_bitrate, kMaxBitrateBps);
|
|
}
|
|
|
|
// Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to
|
|
// a value <= 0, setting the parameters returns false.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
EXPECT_EQ(1UL, rtp_parameters.encodings.size());
|
|
EXPECT_EQ(webrtc::kDefaultBitratePriority,
|
|
rtp_parameters.encodings[0].bitrate_priority);
|
|
|
|
rtp_parameters.encodings[0].bitrate_priority = 0;
|
|
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
|
rtp_parameters.encodings[0].bitrate_priority = -1.0;
|
|
EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
|
}
|
|
|
|
// Test that the bitrate_priority in the send stream config gets updated when
|
|
// SetRtpSendParameters is set for the VoiceMediaChannel.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
|
|
|
|
EXPECT_EQ(1UL, rtp_parameters.encodings.size());
|
|
EXPECT_EQ(webrtc::kDefaultBitratePriority,
|
|
rtp_parameters.encodings[0].bitrate_priority);
|
|
double new_bitrate_priority = 2.0;
|
|
rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority;
|
|
EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
|
|
|
|
// The priority should get set for both the audio channel's rtp parameters
|
|
// and the audio send stream's audio config.
|
|
EXPECT_EQ(
|
|
new_bitrate_priority,
|
|
channel_->GetRtpSendParameters(kSsrcX).encodings[0].bitrate_priority);
|
|
EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority);
|
|
}
|
|
|
|
// Test that GetRtpReceiveParameters returns the currently configured codecs.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
webrtc::RtpParameters rtp_parameters =
|
|
channel_->GetRtpReceiveParameters(kSsrcX);
|
|
ASSERT_EQ(2u, rtp_parameters.codecs.size());
|
|
EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
|
|
EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
|
|
}
|
|
|
|
// Test that GetRtpReceiveParameters returns an SSRC.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
webrtc::RtpParameters rtp_parameters =
|
|
channel_->GetRtpReceiveParameters(kSsrcX);
|
|
ASSERT_EQ(1u, rtp_parameters.encodings.size());
|
|
EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc);
|
|
}
|
|
|
|
// Test that if we set/get parameters multiple times, we get the same results.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
webrtc::RtpParameters initial_params =
|
|
channel_->GetRtpReceiveParameters(kSsrcX);
|
|
|
|
// ... And this shouldn't change the params returned by
|
|
// GetRtpReceiveParameters.
|
|
webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX);
|
|
EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrcX));
|
|
}
|
|
|
|
// Test that GetRtpReceiveParameters returns parameters correctly when SSRCs
|
|
// aren't signaled. It should return an empty "RtpEncodingParameters" when
|
|
// configured to receive an unsignaled stream and no packets have been received
|
|
// yet, and start returning the SSRC once a packet has been received.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) {
|
|
ASSERT_TRUE(SetupChannel());
|
|
// Call necessary methods to configure receiving a default stream as
|
|
// soon as it arrives.
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
|
|
// Call GetDefaultRtpReceiveParameters before configured to receive an
|
|
// unsignaled stream. Should return nothing.
|
|
EXPECT_EQ(webrtc::RtpParameters(),
|
|
channel_->GetDefaultRtpReceiveParameters());
|
|
|
|
// Set a sink for an unsignaled stream.
|
|
std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink());
|
|
channel_->SetDefaultRawAudioSink(std::move(fake_sink));
|
|
|
|
// Call GetDefaultRtpReceiveParameters before the SSRC is known.
|
|
webrtc::RtpParameters rtp_parameters =
|
|
channel_->GetDefaultRtpReceiveParameters();
|
|
ASSERT_EQ(1u, rtp_parameters.encodings.size());
|
|
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
|
|
|
|
// Receive PCMU packet (SSRC=1).
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
// The `ssrc` member should still be unset.
|
|
rtp_parameters = channel_->GetDefaultRtpReceiveParameters();
|
|
ASSERT_EQ(1u, rtp_parameters.encodings.size());
|
|
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
|
|
}
|
|
|
|
// Test that we apply codecs properly.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kCn8000Codec);
|
|
parameters.codecs[0].id = 96;
|
|
parameters.codecs[0].bitrate = 22000;
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, send_codec_spec.payload_type);
|
|
EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps);
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000);
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
|
|
EXPECT_FALSE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that we use Opus/Red by default when it is
|
|
// listed as the first codec and there is an fmtp line.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRed) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs[0].params[""] = "111/111";
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(112, send_codec_spec.red_payload_type);
|
|
}
|
|
|
|
// Test that we do not use Opus/Red by default when it is
|
|
// listed as the first codec but there is no fmtp line.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedNoFmtp) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
|
|
}
|
|
|
|
// Test that we do not use Opus/Red by default.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedDefault) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs[1].params[""] = "111/111";
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
|
|
}
|
|
|
|
// Test that the RED fmtp line must match the payload type.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpMismatch) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs[0].params[""] = "8/8";
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
|
|
}
|
|
|
|
// Test that the RED fmtp line must show 2..32 payloads.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kRed48000Codec);
|
|
parameters.codecs[0].params[""] = "111";
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
|
|
for (int i = 1; i < 32; i++) {
|
|
parameters.codecs[0].params[""] += "/111";
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec2 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec2.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec2.format.name.c_str());
|
|
EXPECT_EQ(112, send_codec_spec2.red_payload_type);
|
|
}
|
|
parameters.codecs[0].params[""] += "/111";
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec3 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, send_codec_spec3.payload_type);
|
|
EXPECT_STRCASEEQ("opus", send_codec_spec3.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec3.red_payload_type);
|
|
}
|
|
|
|
// Test that WebRtcVoiceEngine reconfigures, rather than recreates its
|
|
// AudioSendStream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kCn8000Codec);
|
|
parameters.codecs[0].id = 96;
|
|
parameters.codecs[0].bitrate = 48000;
|
|
const int initial_num = call_.GetNumCreatedSendStreams();
|
|
SetSendParameters(parameters);
|
|
EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
|
|
// Calling SetSendCodec again with same codec which is already set.
|
|
// In this case media channel shouldn't send codec to VoE.
|
|
SetSendParameters(parameters);
|
|
EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
|
|
}
|
|
|
|
// TODO(ossu): Revisit if these tests need to be here, now that these kinds of
|
|
// tests should be available in AudioEncoderOpusTest.
|
|
|
|
// Test that if clockrate is not 48000 for opus, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].clockrate = 50000;
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that if channels=0 for opus, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].channels = 0;
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that if channels=0 for opus, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].channels = 0;
|
|
parameters.codecs[0].params["stereo"] = "1";
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that if channel is 1 for opus and there's no stereo, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].channels = 1;
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that if channel is 1 for opus and stereo=0, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].channels = 1;
|
|
parameters.codecs[0].params["stereo"] = "0";
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that if channel is 1 for opus and stereo=1, we fail.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].channels = 1;
|
|
parameters.codecs[0].params["stereo"] = "1";
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that with bitrate=0 and no stereo, bitrate is 32000.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 32000);
|
|
}
|
|
|
|
// Test that with bitrate=0 and stereo=0, bitrate is 32000.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].params["stereo"] = "0";
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 32000);
|
|
}
|
|
|
|
// Test that with bitrate=invalid and stereo=0, bitrate is 32000.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].params["stereo"] = "0";
|
|
// bitrate that's out of the range between 6000 and 510000 will be clamped.
|
|
parameters.codecs[0].bitrate = 5999;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 6000);
|
|
|
|
parameters.codecs[0].bitrate = 510001;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 510000);
|
|
}
|
|
|
|
// Test that with bitrate=0 and stereo=1, bitrate is 64000.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 0;
|
|
parameters.codecs[0].params["stereo"] = "1";
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 64000);
|
|
}
|
|
|
|
// Test that with bitrate=invalid and stereo=1, bitrate is 64000.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].params["stereo"] = "1";
|
|
// bitrate that's out of the range between 6000 and 510000 will be clamped.
|
|
parameters.codecs[0].bitrate = 5999;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 6000);
|
|
|
|
parameters.codecs[0].bitrate = 510001;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 510000);
|
|
}
|
|
|
|
// Test that with bitrate=N and stereo unset, bitrate is N.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 96000;
|
|
SetSendParameters(parameters);
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, spec.payload_type);
|
|
EXPECT_EQ(96000, spec.target_bitrate_bps);
|
|
EXPECT_EQ("opus", spec.format.name);
|
|
EXPECT_EQ(2u, spec.format.num_channels);
|
|
EXPECT_EQ(48000, spec.format.clockrate_hz);
|
|
}
|
|
|
|
// Test that with bitrate=N and stereo=0, bitrate is N.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 30000;
|
|
parameters.codecs[0].params["stereo"] = "0";
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 30000);
|
|
}
|
|
|
|
// Test that with bitrate=N and without any parameters, bitrate is N.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 30000;
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 30000);
|
|
}
|
|
|
|
// Test that with bitrate=N and stereo=1, bitrate is N.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].bitrate = 30000;
|
|
parameters.codecs[0].params["stereo"] = "1";
|
|
SetSendParameters(parameters);
|
|
CheckSendCodecBitrate(kSsrcX, "opus", 30000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) {
|
|
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
|
|
200000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) {
|
|
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
SetSendCodecsWithoutBitratesUsesCorrectDefaults) {
|
|
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) {
|
|
SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
|
|
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
|
|
200000);
|
|
send_parameters_.max_bandwidth_bps = 100000;
|
|
// Setting max bitrate should keep previous min bitrate
|
|
// Setting max bitrate should not reset start bitrate.
|
|
EXPECT_CALL(*call_.GetMockTransportControllerSend(),
|
|
SetSdpBitrateParameters(
|
|
AllOf(Field(&BitrateConstraints::min_bitrate_bps, 100000),
|
|
Field(&BitrateConstraints::start_bitrate_bps, -1),
|
|
Field(&BitrateConstraints::max_bitrate_bps, 200000))));
|
|
SetSendParameters(send_parameters_);
|
|
}
|
|
|
|
// Test that we can enable NACK with opus as callee.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
|
|
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
|
|
EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
|
|
SetSendParameters(parameters);
|
|
// NACK should be enabled even with no send stream.
|
|
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
|
|
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
|
|
}
|
|
|
|
// Test that we can enable NACK on receive streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
|
|
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
|
|
EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
|
|
SetSendParameters(parameters);
|
|
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
|
|
}
|
|
|
|
// Test that we can disable NACK on receive streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
|
|
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
|
|
SetSendParameters(parameters);
|
|
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
|
|
|
|
parameters.codecs.clear();
|
|
parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(parameters);
|
|
EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
|
|
}
|
|
|
|
// Test that NACK is enabled on a new receive stream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
|
|
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
|
|
SetSendParameters(parameters);
|
|
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
|
|
EXPECT_TRUE(AddRecvStream(kSsrcZ));
|
|
EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioSendParameters send_parameters;
|
|
send_parameters.codecs.push_back(kOpusCodec);
|
|
EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty());
|
|
SetSendParameters(send_parameters);
|
|
|
|
cricket::AudioRecvParameters recv_parameters;
|
|
recv_parameters.codecs.push_back(kIsacCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr);
|
|
EXPECT_FALSE(call_.GetAudioReceiveStream(kSsrcX)->transport_cc());
|
|
|
|
send_parameters.codecs = engine_->send_codecs();
|
|
SetSendParameters(send_parameters);
|
|
ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr);
|
|
EXPECT_TRUE(call_.GetAudioReceiveStream(kSsrcX)->transport_cc());
|
|
}
|
|
|
|
// Test that we can switch back and forth between Opus and ISAC with CN.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
cricket::AudioSendParameters opus_parameters;
|
|
opus_parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(opus_parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
|
|
}
|
|
|
|
cricket::AudioSendParameters isac_parameters;
|
|
isac_parameters.codecs.push_back(kIsacCodec);
|
|
isac_parameters.codecs.push_back(kCn16000Codec);
|
|
isac_parameters.codecs.push_back(kOpusCodec);
|
|
SetSendParameters(isac_parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(103, spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
|
|
}
|
|
|
|
SetSendParameters(opus_parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, spec.payload_type);
|
|
EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
|
|
}
|
|
}
|
|
|
|
// Test that we handle various ways of specifying bitrate.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec); // bitrate == 32000
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(103, spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
|
|
EXPECT_EQ(32000, spec.target_bitrate_bps);
|
|
}
|
|
|
|
parameters.codecs[0].bitrate = 0; // bitrate == default
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(103, spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
|
|
EXPECT_EQ(32000, spec.target_bitrate_bps);
|
|
}
|
|
parameters.codecs[0].bitrate = 28000; // bitrate == 28000
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(103, spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
|
|
EXPECT_EQ(28000, spec.target_bitrate_bps);
|
|
}
|
|
|
|
parameters.codecs[0] = kPcmuCodec; // bitrate == 64000
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(0, spec.payload_type);
|
|
EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str());
|
|
EXPECT_EQ(64000, spec.target_bitrate_bps);
|
|
}
|
|
|
|
parameters.codecs[0].bitrate = 0; // bitrate == default
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(0, spec.payload_type);
|
|
EXPECT_STREQ("PCMU", spec.format.name.c_str());
|
|
EXPECT_EQ(64000, spec.target_bitrate_bps);
|
|
}
|
|
|
|
parameters.codecs[0] = kOpusCodec;
|
|
parameters.codecs[0].bitrate = 0; // bitrate == default
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(111, spec.payload_type);
|
|
EXPECT_STREQ("opus", spec.format.name.c_str());
|
|
EXPECT_EQ(32000, spec.target_bitrate_bps);
|
|
}
|
|
}
|
|
|
|
// Test that we fail if no codecs are specified.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
}
|
|
|
|
// Test that we can set send codecs even with telephone-event codec as the first
|
|
// one on the list.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs[0].id = 98; // DTMF
|
|
parameters.codecs[1].id = 96;
|
|
SetSendParameters(parameters);
|
|
const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that CanInsertDtmf() is governed by the send flag
|
|
TEST_P(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs[0].id = 98; // DTMF
|
|
parameters.codecs[1].id = 96;
|
|
SetSendParameters(parameters);
|
|
EXPECT_FALSE(channel_->CanInsertDtmf());
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
SetSend(false);
|
|
EXPECT_FALSE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that payload type range is limited for telephone-event codec.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kTelephoneEventCodec2);
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs[0].id = 0; // DTMF
|
|
parameters.codecs[1].id = 96;
|
|
SetSendParameters(parameters);
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
parameters.codecs[0].id = 128; // DTMF
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
EXPECT_FALSE(channel_->CanInsertDtmf());
|
|
parameters.codecs[0].id = 127;
|
|
SetSendParameters(parameters);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
parameters.codecs[0].id = -1; // DTMF
|
|
EXPECT_FALSE(channel_->SetSendParameters(parameters));
|
|
EXPECT_FALSE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that we can set send codecs even with CN codec as the first
|
|
// one on the list.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs[0].id = 98; // wideband CN
|
|
parameters.codecs[1].id = 96;
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(98, send_codec_spec.cng_payload_type);
|
|
}
|
|
|
|
// Test that we set VAD and DTMF types correctly as caller.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
// TODO(juberti): cn 32000
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs.push_back(kCn8000Codec);
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs[0].id = 96;
|
|
parameters.codecs[2].id = 97; // wideband CN
|
|
parameters.codecs[4].id = 98; // DTMF
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(97, send_codec_spec.cng_payload_type);
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that we set VAD and DTMF types correctly as callee.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
// TODO(juberti): cn 32000
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs.push_back(kCn8000Codec);
|
|
parameters.codecs.push_back(kTelephoneEventCodec2);
|
|
parameters.codecs[0].id = 96;
|
|
parameters.codecs[2].id = 97; // wideband CN
|
|
parameters.codecs[4].id = 98; // DTMF
|
|
SetSendParameters(parameters);
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
|
|
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(97, send_codec_spec.cng_payload_type);
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
// Test that we only apply VAD if we have a CN codec that matches the
|
|
// send codec clockrate.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
// Set ISAC(16K) and CN(16K). VAD should be activated.
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs[1].id = 97;
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(97, send_codec_spec.cng_payload_type);
|
|
}
|
|
// Set PCMU(8K) and CN(16K). VAD should not be activated.
|
|
parameters.codecs[0] = kPcmuCodec;
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
|
|
}
|
|
// Set PCMU(8K) and CN(8K). VAD should be activated.
|
|
parameters.codecs[1] = kCn8000Codec;
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(13, send_codec_spec.cng_payload_type);
|
|
}
|
|
// Set ISAC(16K) and CN(8K). VAD should not be activated.
|
|
parameters.codecs[0] = kIsacCodec;
|
|
SetSendParameters(parameters);
|
|
{
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
|
|
}
|
|
}
|
|
|
|
// Test that we perform case-insensitive matching of codec names.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioSendParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs.push_back(kCn8000Codec);
|
|
parameters.codecs.push_back(kTelephoneEventCodec1);
|
|
parameters.codecs[0].name = "iSaC";
|
|
parameters.codecs[0].id = 96;
|
|
parameters.codecs[2].id = 97; // wideband CN
|
|
parameters.codecs[4].id = 98; // DTMF
|
|
SetSendParameters(parameters);
|
|
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
|
|
EXPECT_EQ(96, send_codec_spec.payload_type);
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(97, send_codec_spec.cng_payload_type);
|
|
SetSend(true);
|
|
EXPECT_TRUE(channel_->CanInsertDtmf());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
SupportsTransportSequenceNumberHeaderExtension) {
|
|
const std::vector<webrtc::RtpExtension> header_extensions =
|
|
GetDefaultEnabledRtpHeaderExtensions(*engine_);
|
|
EXPECT_THAT(header_extensions,
|
|
Contains(::testing::Field(
|
|
"uri", &RtpExtension::uri,
|
|
webrtc::RtpExtension::kTransportSequenceNumberUri)));
|
|
}
|
|
|
|
// Test support for audio level header extension.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
|
|
TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
|
|
}
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
|
|
TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
|
|
}
|
|
|
|
// Test support for transport sequence number header extension.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) {
|
|
TestSetSendRtpHeaderExtensions(
|
|
webrtc::RtpExtension::kTransportSequenceNumberUri);
|
|
}
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) {
|
|
TestSetRecvRtpHeaderExtensions(
|
|
webrtc::RtpExtension::kTransportSequenceNumberUri);
|
|
}
|
|
|
|
// Test that we can create a channel and start sending on it.
|
|
TEST_P(WebRtcVoiceEngineTestFake, Send) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSendParameters(send_parameters_);
|
|
SetSend(true);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
SetSend(false);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
}
|
|
|
|
// Test that a channel will send if and only if it has a source and is enabled
|
|
// for sending.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSendParameters(send_parameters_);
|
|
SetAudioSend(kSsrcX, true, nullptr);
|
|
SetSend(true);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
SetAudioSend(kSsrcX, true, &fake_source_);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
SetAudioSend(kSsrcX, true, nullptr);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
}
|
|
|
|
// Test that a channel is muted/unmuted.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).muted());
|
|
SetAudioSend(kSsrcX, true, nullptr);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).muted());
|
|
SetAudioSend(kSsrcX, false, nullptr);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).muted());
|
|
}
|
|
|
|
// Test that SetSendParameters() does not alter a stream's send state.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Turn on sending.
|
|
SetSend(true);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Changing RTP header extensions will recreate the AudioSendStream.
|
|
send_parameters_.extensions.push_back(
|
|
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Turn off sending.
|
|
SetSend(false);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Changing RTP header extensions will recreate the AudioSendStream.
|
|
send_parameters_.extensions.clear();
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
}
|
|
|
|
// Test that we can create a channel and start playing out on it.
|
|
TEST_P(WebRtcVoiceEngineTestFake, Playout) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
channel_->SetPlayout(true);
|
|
EXPECT_TRUE(GetRecvStream(kSsrcX).started());
|
|
channel_->SetPlayout(false);
|
|
EXPECT_FALSE(GetRecvStream(kSsrcX).started());
|
|
}
|
|
|
|
// Test that we can add and remove send streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
|
|
SetupForMultiSendStream();
|
|
|
|
// Set the global state for sending.
|
|
SetSend(true);
|
|
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
SetAudioSend(ssrc, true, &fake_source_);
|
|
// Verify that we are in a sending state for all the created streams.
|
|
EXPECT_TRUE(GetSendStream(ssrc).IsSending());
|
|
}
|
|
EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size());
|
|
|
|
// Delete the send streams.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(channel_->RemoveSendStream(ssrc));
|
|
EXPECT_FALSE(call_.GetAudioSendStream(ssrc));
|
|
EXPECT_FALSE(channel_->RemoveSendStream(ssrc));
|
|
}
|
|
EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
|
|
}
|
|
|
|
// Test SetSendCodecs correctly configure the codecs in all send streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
|
SetupForMultiSendStream();
|
|
|
|
// Create send streams.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
}
|
|
|
|
cricket::AudioSendParameters parameters;
|
|
// Set ISAC(16K) and CN(16K). VAD should be activated.
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kCn16000Codec);
|
|
parameters.codecs[1].id = 97;
|
|
SetSendParameters(parameters);
|
|
|
|
// Verify ISAC and VAD are corrected configured on all send channels.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
|
|
const auto& send_codec_spec =
|
|
*call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
|
|
EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(1u, send_codec_spec.format.num_channels);
|
|
EXPECT_EQ(97, send_codec_spec.cng_payload_type);
|
|
}
|
|
|
|
// Change to PCMU(8K) and CN(16K).
|
|
parameters.codecs[0] = kPcmuCodec;
|
|
SetSendParameters(parameters);
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
|
|
const auto& send_codec_spec =
|
|
*call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
|
|
EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
|
|
EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
|
|
}
|
|
}
|
|
|
|
// Test we can SetSend on all send streams correctly.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
|
|
SetupForMultiSendStream();
|
|
|
|
// Create the send channels and they should be a "not sending" date.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
SetAudioSend(ssrc, true, &fake_source_);
|
|
EXPECT_FALSE(GetSendStream(ssrc).IsSending());
|
|
}
|
|
|
|
// Set the global state for starting sending.
|
|
SetSend(true);
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
// Verify that we are in a sending state for all the send streams.
|
|
EXPECT_TRUE(GetSendStream(ssrc).IsSending());
|
|
}
|
|
|
|
// Set the global state for stopping sending.
|
|
SetSend(false);
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
// Verify that we are in a stop state for all the send streams.
|
|
EXPECT_FALSE(GetSendStream(ssrc).IsSending());
|
|
}
|
|
}
|
|
|
|
// Test we can set the correct statistics on all send streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
|
|
SetupForMultiSendStream();
|
|
|
|
// Create send streams.
|
|
for (uint32_t ssrc : kSsrcs4) {
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
}
|
|
|
|
// Create a receive stream to check that none of the send streams end up in
|
|
// the receive stream stats.
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
|
|
// We need send codec to be set to get all stats.
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
SetAudioSendStreamStats();
|
|
|
|
// Check stats for the added streams.
|
|
{
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
cricket::VoiceMediaInfo info;
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
|
|
// We have added 4 send streams. We should see empty stats for all.
|
|
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
|
|
for (const auto& sender : info.senders) {
|
|
VerifyVoiceSenderInfo(sender, false);
|
|
}
|
|
VerifyVoiceSendRecvCodecs(info);
|
|
|
|
// We have added one receive stream. We should see empty stats.
|
|
EXPECT_EQ(info.receivers.size(), 1u);
|
|
EXPECT_EQ(info.receivers[0].ssrc(), 0u);
|
|
}
|
|
|
|
// Remove the kSsrcY stream. No receiver stats.
|
|
{
|
|
cricket::VoiceMediaInfo info;
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
|
|
EXPECT_EQ(0u, info.receivers.size());
|
|
}
|
|
|
|
// Deliver a new packet - a default receive stream should be created and we
|
|
// should see stats again.
|
|
{
|
|
cricket::VoiceMediaInfo info;
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
SetAudioReceiveStreamStats();
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
|
|
EXPECT_EQ(1u, info.receivers.size());
|
|
VerifyVoiceReceiverInfo(info.receivers[0]);
|
|
VerifyVoiceSendRecvCodecs(info);
|
|
}
|
|
}
|
|
|
|
// Test that we can add and remove receive streams, and do proper send/playout.
|
|
// We can receive on multiple streams while sending one stream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
|
|
// Start playout without a receive stream.
|
|
SetSendParameters(send_parameters_);
|
|
channel_->SetPlayout(true);
|
|
|
|
// Adding another stream should enable playout on the new stream only.
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
SetSend(true);
|
|
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Make sure only the new stream is played out.
|
|
EXPECT_TRUE(GetRecvStream(kSsrcY).started());
|
|
|
|
// Adding yet another stream should have stream 2 and 3 enabled for playout.
|
|
EXPECT_TRUE(AddRecvStream(kSsrcZ));
|
|
EXPECT_TRUE(GetRecvStream(kSsrcY).started());
|
|
EXPECT_TRUE(GetRecvStream(kSsrcZ).started());
|
|
|
|
// Stop sending.
|
|
SetSend(false);
|
|
EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
|
|
|
|
// Stop playout.
|
|
channel_->SetPlayout(false);
|
|
EXPECT_FALSE(GetRecvStream(kSsrcY).started());
|
|
EXPECT_FALSE(GetRecvStream(kSsrcZ).started());
|
|
|
|
// Restart playout and make sure recv streams are played out.
|
|
channel_->SetPlayout(true);
|
|
EXPECT_TRUE(GetRecvStream(kSsrcY).started());
|
|
EXPECT_TRUE(GetRecvStream(kSsrcZ).started());
|
|
|
|
// Now remove the recv streams.
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcZ));
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
send_parameters_.options.audio_network_adaptor = true;
|
|
send_parameters_.options.audio_network_adaptor_config = {"1234"};
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
send_parameters_.options.audio_network_adaptor = true;
|
|
send_parameters_.options.audio_network_adaptor_config = {"1234"};
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
cricket::AudioOptions options;
|
|
options.audio_network_adaptor = false;
|
|
SetAudioSend(kSsrcX, true, nullptr, &options);
|
|
EXPECT_EQ(absl::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
send_parameters_.options.audio_network_adaptor = true;
|
|
send_parameters_.options.audio_network_adaptor_config = {"1234"};
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
const int initial_num = call_.GetNumCreatedSendStreams();
|
|
cricket::AudioOptions options;
|
|
options.audio_network_adaptor = absl::nullopt;
|
|
// Unvalued `options.audio_network_adaptor` should not reset audio network
|
|
// adaptor.
|
|
SetAudioSend(kSsrcX, true, nullptr, &options);
|
|
// AudioSendStream not expected to be recreated.
|
|
EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
|
|
EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
|
|
GetAudioNetworkAdaptorConfig(kSsrcX));
|
|
}
|
|
|
|
// Test that we can set the outgoing SSRC properly.
|
|
// SSRC is set in SetupSendStream() by calling AddSendStream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrc) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetStats) {
|
|
// Setup. We need send codec to be set to get all stats.
|
|
EXPECT_TRUE(SetupSendStream());
|
|
// SetupSendStream adds a send stream with kSsrcX, so the receive
|
|
// stream has to use a different SSRC.
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
|
|
SetAudioSendStreamStats();
|
|
|
|
// Check stats for the added streams.
|
|
{
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
cricket::VoiceMediaInfo info;
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
|
|
// We have added one send stream. We should see the stats we've set.
|
|
EXPECT_EQ(1u, info.senders.size());
|
|
VerifyVoiceSenderInfo(info.senders[0], false);
|
|
// We have added one receive stream. We should see empty stats.
|
|
EXPECT_EQ(info.receivers.size(), 1u);
|
|
EXPECT_EQ(info.receivers[0].ssrc(), 0u);
|
|
}
|
|
|
|
// Start sending - this affects some reported stats.
|
|
{
|
|
cricket::VoiceMediaInfo info;
|
|
SetSend(true);
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
VerifyVoiceSenderInfo(info.senders[0], true);
|
|
VerifyVoiceSendRecvCodecs(info);
|
|
}
|
|
|
|
// Remove the kSsrcY stream. No receiver stats.
|
|
{
|
|
cricket::VoiceMediaInfo info;
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
EXPECT_EQ(1u, info.senders.size());
|
|
EXPECT_EQ(0u, info.receivers.size());
|
|
}
|
|
|
|
// Deliver a new packet - a default receive stream should be created and we
|
|
// should see stats again.
|
|
{
|
|
cricket::VoiceMediaInfo info;
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
SetAudioReceiveStreamStats();
|
|
EXPECT_CALL(*adm_, GetPlayoutUnderrunCount()).WillOnce(Return(0));
|
|
EXPECT_EQ(true,
|
|
channel_->GetStats(&info, /*get_and_clear_legacy_stats=*/true));
|
|
EXPECT_EQ(1u, info.senders.size());
|
|
EXPECT_EQ(1u, info.receivers.size());
|
|
VerifyVoiceReceiverInfo(info.receivers[0]);
|
|
VerifyVoiceSendRecvCodecs(info);
|
|
}
|
|
}
|
|
|
|
// Test that we can set the outgoing SSRC properly with multiple streams.
|
|
// SSRC is set in SetupSendStream() by calling AddSendStream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
|
}
|
|
|
|
// Test that the local SSRC is the same on sending and receiving channels if the
|
|
// receive channel is created before the send channel.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
|
|
EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
|
|
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
|
}
|
|
|
|
// Test that we can properly receive packets.
|
|
TEST_P(WebRtcVoiceEngineTestFake, Recv) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_TRUE(AddRecvStream(1));
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
EXPECT_TRUE(
|
|
GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
}
|
|
|
|
// Test that we can properly receive packets on multiple streams.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
const uint32_t ssrc1 = 1;
|
|
const uint32_t ssrc2 = 2;
|
|
const uint32_t ssrc3 = 3;
|
|
EXPECT_TRUE(AddRecvStream(ssrc1));
|
|
EXPECT_TRUE(AddRecvStream(ssrc2));
|
|
EXPECT_TRUE(AddRecvStream(ssrc3));
|
|
// Create packets with the right SSRCs.
|
|
unsigned char packets[4][sizeof(kPcmuFrame)];
|
|
for (size_t i = 0; i < arraysize(packets); ++i) {
|
|
memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
|
|
rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
|
|
}
|
|
|
|
const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1);
|
|
const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2);
|
|
const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3);
|
|
|
|
EXPECT_EQ(s1.received_packets(), 0);
|
|
EXPECT_EQ(s2.received_packets(), 0);
|
|
EXPECT_EQ(s3.received_packets(), 0);
|
|
|
|
DeliverPacket(packets[0], sizeof(packets[0]));
|
|
EXPECT_EQ(s1.received_packets(), 0);
|
|
EXPECT_EQ(s2.received_packets(), 0);
|
|
EXPECT_EQ(s3.received_packets(), 0);
|
|
|
|
DeliverPacket(packets[1], sizeof(packets[1]));
|
|
EXPECT_EQ(s1.received_packets(), 1);
|
|
EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1])));
|
|
EXPECT_EQ(s2.received_packets(), 0);
|
|
EXPECT_EQ(s3.received_packets(), 0);
|
|
|
|
DeliverPacket(packets[2], sizeof(packets[2]));
|
|
EXPECT_EQ(s1.received_packets(), 1);
|
|
EXPECT_EQ(s2.received_packets(), 1);
|
|
EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2])));
|
|
EXPECT_EQ(s3.received_packets(), 0);
|
|
|
|
DeliverPacket(packets[3], sizeof(packets[3]));
|
|
EXPECT_EQ(s1.received_packets(), 1);
|
|
EXPECT_EQ(s2.received_packets(), 1);
|
|
EXPECT_EQ(s3.received_packets(), 1);
|
|
EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3])));
|
|
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(ssrc3));
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2));
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1));
|
|
}
|
|
|
|
// Test that receiving on an unsignaled stream works (a stream is created).
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaled) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
|
|
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_TRUE(
|
|
GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
}
|
|
|
|
// Tests that when we add a stream without SSRCs, but contains a stream_id
|
|
// that it is stored and its stream id is later used when the first packet
|
|
// arrives to properly create a receive stream with a sync label.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
|
|
const char kSyncLabel[] = "sync_label";
|
|
EXPECT_TRUE(SetupChannel());
|
|
cricket::StreamParams unsignaled_stream;
|
|
unsignaled_stream.set_stream_ids({kSyncLabel});
|
|
ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
|
|
// The stream shouldn't have been created at this point because it doesn't
|
|
// have any SSRCs.
|
|
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
|
|
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_TRUE(
|
|
GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group);
|
|
|
|
// Remset the unsignaled stream to clear the cached parameters. If a new
|
|
// default unsignaled receive stream is created it will not have a sync group.
|
|
channel_->ResetUnsignaledRecvStream();
|
|
channel_->RemoveRecvStream(kSsrc1);
|
|
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_TRUE(
|
|
GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
ResetUnsignaledRecvStreamDeletesAllDefaultStreams) {
|
|
ASSERT_TRUE(SetupChannel());
|
|
// No receive streams to start with.
|
|
ASSERT_TRUE(call_.GetAudioReceiveStreams().empty());
|
|
|
|
// Deliver a couple packets with unsignaled SSRCs.
|
|
unsigned char packet[sizeof(kPcmuFrame)];
|
|
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
|
|
rtc::SetBE32(&packet[8], 0x1234);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
rtc::SetBE32(&packet[8], 0x5678);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
|
|
// Verify that the receive streams were created.
|
|
const auto& receivers1 = call_.GetAudioReceiveStreams();
|
|
ASSERT_EQ(receivers1.size(), 2u);
|
|
|
|
// Should remove all default streams.
|
|
channel_->ResetUnsignaledRecvStream();
|
|
const auto& receivers2 = call_.GetAudioReceiveStreams();
|
|
EXPECT_EQ(0u, receivers2.size());
|
|
}
|
|
|
|
// Test that receiving N unsignaled stream works (streams will be created), and
|
|
// that packets are forwarded to them all.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
unsigned char packet[sizeof(kPcmuFrame)];
|
|
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
// Note that SSRC = 0 is not supported.
|
|
for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
|
|
rtc::SetBE32(&packet[8], ssrc);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
|
|
// Verify we have one new stream for each loop iteration.
|
|
EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_EQ(1, GetRecvStream(ssrc).received_packets());
|
|
EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
|
|
}
|
|
|
|
// Sending on the same SSRCs again should not create new streams.
|
|
for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
|
|
rtc::SetBE32(&packet[8], ssrc);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
|
|
EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_EQ(2, GetRecvStream(ssrc).received_packets());
|
|
EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
|
|
}
|
|
|
|
// Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced.
|
|
constexpr uint32_t kAnotherSsrc = 667;
|
|
rtc::SetBE32(&packet[8], kAnotherSsrc);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
|
|
const auto& streams = call_.GetAudioReceiveStreams();
|
|
EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size());
|
|
size_t i = 0;
|
|
for (uint32_t ssrc = 2; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc, ++i) {
|
|
EXPECT_EQ(ssrc, streams[i]->GetConfig().rtp.remote_ssrc);
|
|
EXPECT_EQ(2, streams[i]->received_packets());
|
|
}
|
|
EXPECT_EQ(kAnotherSsrc, streams[i]->GetConfig().rtp.remote_ssrc);
|
|
EXPECT_EQ(1, streams[i]->received_packets());
|
|
// Sanity check that we've checked all streams.
|
|
EXPECT_EQ(kMaxUnsignaledRecvStreams, (i + 1));
|
|
}
|
|
|
|
// Test that a default channel is created even after a signaled stream has been
|
|
// added, and that this stream will get any packets for unknown SSRCs.
|
|
TEST_P(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
unsigned char packet[sizeof(kPcmuFrame)];
|
|
memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
|
|
|
|
// Add a known stream, send packet and verify we got it.
|
|
const uint32_t signaled_ssrc = 1;
|
|
rtc::SetBE32(&packet[8], signaled_ssrc);
|
|
EXPECT_TRUE(AddRecvStream(signaled_ssrc));
|
|
DeliverPacket(packet, sizeof(packet));
|
|
EXPECT_TRUE(
|
|
GetRecvStream(signaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
|
|
// Note that the first unknown SSRC cannot be 0, because we only support
|
|
// creating receive streams for SSRC!=0.
|
|
const uint32_t unsignaled_ssrc = 7011;
|
|
rtc::SetBE32(&packet[8], unsignaled_ssrc);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
EXPECT_TRUE(
|
|
GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
|
|
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
|
|
DeliverPacket(packet, sizeof(packet));
|
|
EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
|
|
|
|
rtc::SetBE32(&packet[8], signaled_ssrc);
|
|
DeliverPacket(packet, sizeof(packet));
|
|
EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
|
|
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
}
|
|
|
|
// Two tests to verify that adding a receive stream with the same SSRC as a
|
|
// previously added unsignaled stream will only recreate underlying stream
|
|
// objects if the stream parameters have changed.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
|
|
// Spawn unsignaled stream with SSRC=1.
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_TRUE(
|
|
GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
|
|
// Verify that the underlying stream object in Call is not recreated when a
|
|
// stream with SSRC=1 is added.
|
|
const auto& streams = call_.GetAudioReceiveStreams();
|
|
EXPECT_EQ(1u, streams.size());
|
|
int audio_receive_stream_id = streams.front()->id();
|
|
EXPECT_TRUE(AddRecvStream(1));
|
|
EXPECT_EQ(1u, streams.size());
|
|
EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Updates) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
|
|
// Spawn unsignaled stream with SSRC=1.
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_TRUE(
|
|
GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
|
|
|
|
// Verify that the underlying stream object in Call gets updated when a
|
|
// stream with SSRC=1 is added, and which has changed stream parameters.
|
|
const auto& streams = call_.GetAudioReceiveStreams();
|
|
EXPECT_EQ(1u, streams.size());
|
|
// The sync_group id should be empty.
|
|
EXPECT_TRUE(streams.front()->GetConfig().sync_group.empty());
|
|
|
|
const std::string new_stream_id("stream_id");
|
|
int audio_receive_stream_id = streams.front()->id();
|
|
cricket::StreamParams stream_params;
|
|
stream_params.ssrcs.push_back(1);
|
|
stream_params.set_stream_ids({new_stream_id});
|
|
|
|
EXPECT_TRUE(channel_->AddRecvStream(stream_params));
|
|
EXPECT_EQ(1u, streams.size());
|
|
// The audio receive stream should not have been recreated.
|
|
EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
|
|
|
|
// The sync_group id should now match with the new stream params.
|
|
EXPECT_EQ(new_stream_id, streams.front()->GetConfig().sync_group);
|
|
}
|
|
|
|
// Test that AddRecvStream creates new stream.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStream) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
EXPECT_TRUE(AddRecvStream(1));
|
|
}
|
|
|
|
// Test that after adding a recv stream, we do not decode more codecs than
|
|
// those previously passed into SetRecvCodecs.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs.push_back(kIsacCodec);
|
|
parameters.codecs.push_back(kPcmuCodec);
|
|
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
|
|
(ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
|
|
{{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
|
|
}
|
|
|
|
// Test that we properly clean up any streams that were added, even if
|
|
// not explicitly removed.
|
|
TEST_P(WebRtcVoiceEngineTestFake, StreamCleanup) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_TRUE(AddRecvStream(1));
|
|
EXPECT_TRUE(AddRecvStream(2));
|
|
|
|
EXPECT_EQ(1u, call_.GetAudioSendStreams().size());
|
|
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
delete channel_;
|
|
channel_ = NULL;
|
|
EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
|
|
EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamSuccessWithZeroSsrc) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(AddRecvStream(0));
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithSameSsrc) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_TRUE(AddRecvStream(1));
|
|
EXPECT_FALSE(AddRecvStream(1));
|
|
}
|
|
|
|
// Test the InsertDtmf on default send stream as caller.
|
|
TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) {
|
|
TestInsertDtmf(0, true, kTelephoneEventCodec1);
|
|
}
|
|
|
|
// Test the InsertDtmf on default send stream as callee
|
|
TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) {
|
|
TestInsertDtmf(0, false, kTelephoneEventCodec2);
|
|
}
|
|
|
|
// Test the InsertDtmf on specified send stream as caller.
|
|
TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) {
|
|
TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2);
|
|
}
|
|
|
|
// Test the InsertDtmf on specified send stream as callee.
|
|
TEST_P(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
|
|
TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1);
|
|
}
|
|
|
|
// Test propagation of extmap allow mixed setting.
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCaller) {
|
|
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
|
|
}
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCaller) {
|
|
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
|
|
}
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCallee) {
|
|
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
|
|
}
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCallee) {
|
|
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetAudioOptions) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_CALL(*adm_, BuiltInAECIsAvailable())
|
|
.Times(8)
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, BuiltInAGCIsAvailable())
|
|
.Times(4)
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, BuiltInNSIsAvailable())
|
|
.Times(2)
|
|
.WillRepeatedly(Return(false));
|
|
|
|
EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
|
|
EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
|
|
|
|
// Nothing set in AudioOptions, so everything should be as default.
|
|
send_parameters_.options = cricket::AudioOptions();
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_TRUE(IsHighPassFilterEnabled());
|
|
}
|
|
EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
|
|
EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
|
|
|
|
// Turn echo cancellation off
|
|
send_parameters_.options.echo_cancellation = false;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/false);
|
|
}
|
|
|
|
// Turn echo cancellation back on, with settings, and make sure
|
|
// nothing else changed.
|
|
send_parameters_.options.echo_cancellation = true;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
}
|
|
|
|
// Turn off echo cancellation and delay agnostic aec.
|
|
send_parameters_.options.echo_cancellation = false;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/false);
|
|
}
|
|
|
|
// Restore AEC to be on to work with the following tests.
|
|
send_parameters_.options.echo_cancellation = true;
|
|
SetSendParameters(send_parameters_);
|
|
|
|
// Turn off AGC
|
|
send_parameters_.options.auto_gain_control = false;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(apm_config_.gain_controller1.enabled);
|
|
}
|
|
|
|
// Turn AGC back on
|
|
send_parameters_.options.auto_gain_control = true;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_TRUE(apm_config_.gain_controller1.enabled);
|
|
}
|
|
|
|
// Turn off other options.
|
|
send_parameters_.options.noise_suppression = false;
|
|
send_parameters_.options.highpass_filter = false;
|
|
send_parameters_.options.stereo_swapping = true;
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(IsHighPassFilterEnabled());
|
|
EXPECT_TRUE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_FALSE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
}
|
|
|
|
// Set options again to ensure it has no impact.
|
|
SetSendParameters(send_parameters_);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_TRUE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_FALSE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
}
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, InitRecordingOnSend) {
|
|
EXPECT_CALL(*adm_, RecordingIsInitialized()).WillOnce(Return(false));
|
|
EXPECT_CALL(*adm_, Recording()).WillOnce(Return(false));
|
|
EXPECT_CALL(*adm_, InitRecording()).Times(1);
|
|
|
|
std::unique_ptr<cricket::VoiceMediaChannel> channel(
|
|
engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
|
|
cricket::AudioOptions(),
|
|
webrtc::CryptoOptions()));
|
|
|
|
channel->SetSend(true);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SkipInitRecordingOnSend) {
|
|
EXPECT_CALL(*adm_, RecordingIsInitialized()).Times(0);
|
|
EXPECT_CALL(*adm_, Recording()).Times(0);
|
|
EXPECT_CALL(*adm_, InitRecording()).Times(0);
|
|
|
|
cricket::AudioOptions options;
|
|
options.init_recording_on_send = false;
|
|
|
|
std::unique_ptr<cricket::VoiceMediaChannel> channel(
|
|
engine_->CreateMediaChannel(&call_, cricket::MediaConfig(), options,
|
|
webrtc::CryptoOptions()));
|
|
|
|
channel->SetSend(true);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_CALL(*adm_, BuiltInAECIsAvailable())
|
|
.Times(use_null_apm_ ? 4 : 8)
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, BuiltInAGCIsAvailable())
|
|
.Times(use_null_apm_ ? 7 : 8)
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, BuiltInNSIsAvailable())
|
|
.Times(use_null_apm_ ? 5 : 8)
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, RecordingIsInitialized())
|
|
.Times(2)
|
|
.WillRepeatedly(Return(false));
|
|
|
|
EXPECT_CALL(*adm_, Recording()).Times(2).WillRepeatedly(Return(false));
|
|
EXPECT_CALL(*adm_, InitRecording()).Times(2).WillRepeatedly(Return(0));
|
|
|
|
std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
|
|
static_cast<cricket::WebRtcVoiceMediaChannel*>(
|
|
engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
|
|
cricket::AudioOptions(),
|
|
webrtc::CryptoOptions())));
|
|
std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel2(
|
|
static_cast<cricket::WebRtcVoiceMediaChannel*>(
|
|
engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
|
|
cricket::AudioOptions(),
|
|
webrtc::CryptoOptions())));
|
|
|
|
// Have to add a stream to make SetSend work.
|
|
cricket::StreamParams stream1;
|
|
stream1.ssrcs.push_back(1);
|
|
channel1->AddSendStream(stream1);
|
|
cricket::StreamParams stream2;
|
|
stream2.ssrcs.push_back(2);
|
|
channel2->AddSendStream(stream2);
|
|
|
|
// AEC and AGC and NS
|
|
cricket::AudioSendParameters parameters_options_all = send_parameters_;
|
|
parameters_options_all.options.echo_cancellation = true;
|
|
parameters_options_all.options.auto_gain_control = true;
|
|
parameters_options_all.options.noise_suppression = true;
|
|
EXPECT_TRUE(channel1->SetSendParameters(parameters_options_all));
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
VerifyGainControlEnabledCorrectly();
|
|
EXPECT_TRUE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
EXPECT_EQ(parameters_options_all.options, channel1->options());
|
|
EXPECT_TRUE(channel2->SetSendParameters(parameters_options_all));
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
VerifyGainControlEnabledCorrectly();
|
|
EXPECT_EQ(parameters_options_all.options, channel2->options());
|
|
}
|
|
|
|
// unset NS
|
|
cricket::AudioSendParameters parameters_options_no_ns = send_parameters_;
|
|
parameters_options_no_ns.options.noise_suppression = false;
|
|
EXPECT_TRUE(channel1->SetSendParameters(parameters_options_no_ns));
|
|
cricket::AudioOptions expected_options = parameters_options_all.options;
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
VerifyGainControlEnabledCorrectly();
|
|
expected_options.echo_cancellation = true;
|
|
expected_options.auto_gain_control = true;
|
|
expected_options.noise_suppression = false;
|
|
EXPECT_EQ(expected_options, channel1->options());
|
|
}
|
|
|
|
// unset AGC
|
|
cricket::AudioSendParameters parameters_options_no_agc = send_parameters_;
|
|
parameters_options_no_agc.options.auto_gain_control = false;
|
|
EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc));
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_TRUE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
expected_options.echo_cancellation = true;
|
|
expected_options.auto_gain_control = false;
|
|
expected_options.noise_suppression = true;
|
|
EXPECT_EQ(expected_options, channel2->options());
|
|
}
|
|
|
|
EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all));
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
VerifyGainControlEnabledCorrectly();
|
|
EXPECT_TRUE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
}
|
|
|
|
channel1->SetSend(true);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
VerifyGainControlEnabledCorrectly();
|
|
EXPECT_FALSE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
}
|
|
|
|
channel2->SetSend(true);
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_TRUE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
}
|
|
|
|
// Make sure settings take effect while we are sending.
|
|
cricket::AudioSendParameters parameters_options_no_agc_nor_ns =
|
|
send_parameters_;
|
|
parameters_options_no_agc_nor_ns.options.auto_gain_control = false;
|
|
parameters_options_no_agc_nor_ns.options.noise_suppression = false;
|
|
EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc_nor_ns));
|
|
if (!use_null_apm_) {
|
|
VerifyEchoCancellationSettings(/*enabled=*/true);
|
|
EXPECT_FALSE(apm_config_.gain_controller1.enabled);
|
|
EXPECT_FALSE(apm_config_.noise_suppression.enabled);
|
|
EXPECT_EQ(apm_config_.noise_suppression.level, kDefaultNsLevel);
|
|
expected_options.echo_cancellation = true;
|
|
expected_options.auto_gain_control = false;
|
|
expected_options.noise_suppression = false;
|
|
EXPECT_EQ(expected_options, channel2->options());
|
|
}
|
|
}
|
|
|
|
// This test verifies DSCP settings are properly applied on voice media channel.
|
|
TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::FakeNetworkInterface network_interface;
|
|
cricket::MediaConfig config;
|
|
std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel;
|
|
webrtc::RtpParameters parameters;
|
|
|
|
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
|
|
engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
|
|
webrtc::CryptoOptions())));
|
|
channel->SetInterface(&network_interface);
|
|
// Default value when DSCP is disabled should be DSCP_DEFAULT.
|
|
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
|
|
channel->SetInterface(nullptr);
|
|
|
|
config.enable_dscp = true;
|
|
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
|
|
engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
|
|
webrtc::CryptoOptions())));
|
|
channel->SetInterface(&network_interface);
|
|
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
|
|
|
|
// Create a send stream to configure
|
|
EXPECT_TRUE(
|
|
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
|
|
parameters = channel->GetRtpSendParameters(kSsrcZ);
|
|
ASSERT_FALSE(parameters.encodings.empty());
|
|
|
|
// Various priorities map to various dscp values.
|
|
parameters.encodings[0].network_priority = webrtc::Priority::kHigh;
|
|
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
|
|
EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp());
|
|
parameters.encodings[0].network_priority = webrtc::Priority::kVeryLow;
|
|
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
|
|
EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp());
|
|
|
|
// Packets should also self-identify their dscp in PacketOptions.
|
|
const uint8_t kData[10] = {0};
|
|
EXPECT_TRUE(channel->SendRtcp(kData, sizeof(kData)));
|
|
EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
|
|
channel->SetInterface(nullptr);
|
|
|
|
// Verify that setting the option to false resets the
|
|
// DiffServCodePoint.
|
|
config.enable_dscp = false;
|
|
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
|
|
engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
|
|
webrtc::CryptoOptions())));
|
|
channel->SetInterface(&network_interface);
|
|
// Default value when DSCP is disabled should be DSCP_DEFAULT.
|
|
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
|
|
|
|
channel->SetInterface(nullptr);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolume) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5));
|
|
cricket::StreamParams stream;
|
|
stream.ssrcs.push_back(kSsrcY);
|
|
EXPECT_TRUE(channel_->AddRecvStream(stream));
|
|
EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain());
|
|
EXPECT_TRUE(channel_->SetOutputVolume(kSsrcY, 3));
|
|
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
|
|
// Spawn an unsignaled stream by sending a packet - gain should be 1.
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain());
|
|
|
|
// Should remember the volume "2" which will be set on new unsignaled streams,
|
|
// and also set the gain to 2 on existing unsignaled streams.
|
|
EXPECT_TRUE(channel_->SetDefaultOutputVolume(2));
|
|
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain());
|
|
|
|
// Spawn an unsignaled stream by sending a packet - gain should be 2.
|
|
unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
|
|
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
|
|
rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
|
|
DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
|
|
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
|
|
|
|
// Setting gain for all unsignaled streams.
|
|
EXPECT_TRUE(channel_->SetDefaultOutputVolume(3));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain());
|
|
}
|
|
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcX).gain());
|
|
|
|
// Setting gain on an individual stream affects only that.
|
|
EXPECT_TRUE(channel_->SetOutputVolume(kSsrcX, 4));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain());
|
|
}
|
|
EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, BaseMinimumPlayoutDelayMs) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
EXPECT_FALSE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200));
|
|
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
|
|
|
cricket::StreamParams stream;
|
|
stream.ssrcs.push_back(kSsrcY);
|
|
EXPECT_TRUE(channel_->AddRecvStream(stream));
|
|
EXPECT_EQ(0, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
|
|
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300));
|
|
EXPECT_EQ(300, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake,
|
|
BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
|
|
// Here base minimum delay is abbreviated to delay in comments for shortness.
|
|
EXPECT_TRUE(SetupChannel());
|
|
|
|
// Spawn an unsignaled stream by sending a packet - delay should be 0.
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_EQ(0, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
|
// Check that it doesn't provide default values for unknown ssrc.
|
|
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
|
|
|
// Check that default value for unsignaled streams is 0.
|
|
EXPECT_EQ(0, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
|
|
|
// Should remember the delay 100 which will be set on new unsignaled streams,
|
|
// and also set the delay to 100 on existing unsignaled streams.
|
|
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100));
|
|
EXPECT_EQ(100, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
|
// Check that it doesn't provide default values for unknown ssrc.
|
|
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
|
|
|
// Spawn an unsignaled stream by sending a packet - delay should be 100.
|
|
unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
|
|
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
|
|
rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
|
|
DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
|
|
EXPECT_EQ(100, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
|
|
|
// Setting delay with SSRC=0 should affect all unsignaled streams.
|
|
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
|
}
|
|
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
|
|
|
// Setting delay on an individual stream affects only that.
|
|
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
|
|
}
|
|
EXPECT_EQ(400, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
|
|
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
|
|
// Check that it doesn't provide default values for unknown ssrc.
|
|
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) {
|
|
const uint32_t kAudioSsrc = 123;
|
|
const std::string kStreamId = "AvSyncLabel";
|
|
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc);
|
|
sp.set_stream_ids({kStreamId});
|
|
// Creating two channels to make sure that sync label is set properly for both
|
|
// the default voice channel and following ones.
|
|
EXPECT_TRUE(channel_->AddRecvStream(sp));
|
|
sp.ssrcs[0] += 1;
|
|
EXPECT_TRUE(channel_->AddRecvStream(sp));
|
|
|
|
ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
EXPECT_EQ(kStreamId,
|
|
call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group)
|
|
<< "SyncGroup should be set based on stream id";
|
|
EXPECT_EQ(kStreamId,
|
|
call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group)
|
|
<< "SyncGroup should be set based on stream id";
|
|
}
|
|
|
|
// TODO(solenberg): Remove, once recv streams are configured through Call.
|
|
// (This is then covered by TestSetRecvRtpHeaderExtensions.)
|
|
TEST_P(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
|
|
// Test that setting the header extensions results in the expected state
|
|
// changes on an associated Call.
|
|
std::vector<uint32_t> ssrcs;
|
|
ssrcs.push_back(223);
|
|
ssrcs.push_back(224);
|
|
|
|
EXPECT_TRUE(SetupSendStream());
|
|
SetSendParameters(send_parameters_);
|
|
for (uint32_t ssrc : ssrcs) {
|
|
EXPECT_TRUE(
|
|
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)));
|
|
}
|
|
|
|
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
for (uint32_t ssrc : ssrcs) {
|
|
const auto* s = call_.GetAudioReceiveStream(ssrc);
|
|
EXPECT_NE(nullptr, s);
|
|
EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
|
|
}
|
|
|
|
// Set up receive extensions.
|
|
const std::vector<webrtc::RtpExtension> header_extensions =
|
|
GetDefaultEnabledRtpHeaderExtensions(*engine_);
|
|
cricket::AudioRecvParameters recv_parameters;
|
|
recv_parameters.extensions = header_extensions;
|
|
channel_->SetRecvParameters(recv_parameters);
|
|
EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
|
|
for (uint32_t ssrc : ssrcs) {
|
|
const auto* s = call_.GetAudioReceiveStream(ssrc);
|
|
EXPECT_NE(nullptr, s);
|
|
const auto& s_exts = s->GetConfig().rtp.extensions;
|
|
EXPECT_EQ(header_extensions.size(), s_exts.size());
|
|
for (const auto& e_ext : header_extensions) {
|
|
for (const auto& s_ext : s_exts) {
|
|
if (e_ext.id == s_ext.id) {
|
|
EXPECT_EQ(e_ext.uri, s_ext.uri);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Disable receive extensions.
|
|
channel_->SetRecvParameters(cricket::AudioRecvParameters());
|
|
for (uint32_t ssrc : ssrcs) {
|
|
const auto* s = call_.GetAudioReceiveStream(ssrc);
|
|
EXPECT_NE(nullptr, s);
|
|
EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
|
|
}
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
|
|
// Test that packets are forwarded to the Call when configured accordingly.
|
|
const uint32_t kAudioSsrc = 1;
|
|
rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
static const unsigned char kRtcp[] = {
|
|
0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00,
|
|
0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
|
|
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
|
|
rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp));
|
|
|
|
EXPECT_TRUE(SetupSendStream());
|
|
cricket::WebRtcVoiceMediaChannel* media_channel =
|
|
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
|
|
SetSendParameters(send_parameters_);
|
|
EXPECT_TRUE(media_channel->AddRecvStream(
|
|
cricket::StreamParams::CreateLegacy(kAudioSsrc)));
|
|
|
|
EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
|
|
const cricket::FakeAudioReceiveStream* s =
|
|
call_.GetAudioReceiveStream(kAudioSsrc);
|
|
EXPECT_EQ(0, s->received_packets());
|
|
channel_->OnPacketReceived(kPcmuPacket, /* packet_time_us */ -1);
|
|
rtc::Thread::Current()->ProcessMessages(0);
|
|
|
|
EXPECT_EQ(1, s->received_packets());
|
|
}
|
|
|
|
// All receive channels should be associated with the first send channel,
|
|
// since they do not send RTCP SR.
|
|
TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
|
|
EXPECT_TRUE(SetupSendStream());
|
|
EXPECT_TRUE(AddRecvStream(kSsrcY));
|
|
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
|
|
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
|
|
EXPECT_TRUE(AddRecvStream(kSsrcW));
|
|
EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
|
|
EXPECT_TRUE(SetupRecvStream());
|
|
EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
|
|
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
|
EXPECT_TRUE(AddRecvStream(kSsrcZ));
|
|
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
|
|
EXPECT_TRUE(
|
|
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcW)));
|
|
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
|
|
EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSink) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink());
|
|
|
|
// Setting the sink before a recv stream exists should do nothing.
|
|
channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1));
|
|
EXPECT_TRUE(AddRecvStream(kSsrcX));
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
|
|
|
|
// Now try actually setting the sink.
|
|
channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2));
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
|
|
|
|
// Now try resetting it.
|
|
channel_->SetRawAudioSink(kSsrcX, nullptr);
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
|
|
}
|
|
|
|
TEST_P(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_3(new FakeAudioSink());
|
|
std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink());
|
|
|
|
// Should be able to set a default sink even when no stream exists.
|
|
channel_->SetDefaultRawAudioSink(std::move(fake_sink_1));
|
|
|
|
// Spawn an unsignaled stream by sending a packet - it should be assigned the
|
|
// default sink.
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
|
|
|
// Try resetting the default sink.
|
|
channel_->SetDefaultRawAudioSink(nullptr);
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
|
|
|
|
// Try setting the default sink while the default stream exists.
|
|
channel_->SetDefaultRawAudioSink(std::move(fake_sink_2));
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
|
|
|
// If we remove and add a default stream, it should get the same sink.
|
|
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1));
|
|
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
|
|
|
// Spawn another unsignaled stream - it should be assigned the default sink
|
|
// and the previous unsignaled stream should lose it.
|
|
unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
|
|
memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
|
|
rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
|
|
DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
|
|
}
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
|
|
|
|
// Reset the default sink - the second unsignaled stream should lose it.
|
|
channel_->SetDefaultRawAudioSink(nullptr);
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
|
|
}
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
|
|
|
|
// Try setting the default sink while two streams exists.
|
|
channel_->SetDefaultRawAudioSink(std::move(fake_sink_3));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
|
|
}
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
|
|
|
|
// Try setting the sink for the first unsignaled stream using its known SSRC.
|
|
channel_->SetRawAudioSink(kSsrc1, std::move(fake_sink_4));
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
|
|
}
|
|
EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
|
|
if (kMaxUnsignaledRecvStreams > 1) {
|
|
EXPECT_NE(GetRecvStream(kSsrc1).sink(), GetRecvStream(kSsrcX).sink());
|
|
}
|
|
}
|
|
|
|
// Test that, just like the video channel, the voice channel communicates the
|
|
// network state to the call.
|
|
TEST_P(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
|
|
EXPECT_TRUE(SetupChannel());
|
|
|
|
EXPECT_EQ(webrtc::kNetworkUp,
|
|
call_.GetNetworkState(webrtc::MediaType::AUDIO));
|
|
EXPECT_EQ(webrtc::kNetworkUp,
|
|
call_.GetNetworkState(webrtc::MediaType::VIDEO));
|
|
|
|
channel_->OnReadyToSend(false);
|
|
EXPECT_EQ(webrtc::kNetworkDown,
|
|
call_.GetNetworkState(webrtc::MediaType::AUDIO));
|
|
EXPECT_EQ(webrtc::kNetworkUp,
|
|
call_.GetNetworkState(webrtc::MediaType::VIDEO));
|
|
|
|
channel_->OnReadyToSend(true);
|
|
EXPECT_EQ(webrtc::kNetworkUp,
|
|
call_.GetNetworkState(webrtc::MediaType::AUDIO));
|
|
EXPECT_EQ(webrtc::kNetworkUp,
|
|
call_.GetNetworkState(webrtc::MediaType::VIDEO));
|
|
}
|
|
|
|
// Test that playout is still started after changing parameters
|
|
TEST_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
|
|
SetupRecvStream();
|
|
channel_->SetPlayout(true);
|
|
EXPECT_TRUE(GetRecvStream(kSsrcX).started());
|
|
|
|
// Changing RTP header extensions will recreate the
|
|
// AudioReceiveStreamInterface.
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.extensions.push_back(
|
|
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
|
|
channel_->SetRecvParameters(parameters);
|
|
|
|
EXPECT_TRUE(GetRecvStream(kSsrcX).started());
|
|
}
|
|
|
|
// Tests when GetSources is called with non-existing ssrc, it will return an
|
|
// empty list of RtpSource without crashing.
|
|
TEST_P(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) {
|
|
// Setup an recv stream with `kSsrcX`.
|
|
SetupRecvStream();
|
|
cricket::WebRtcVoiceMediaChannel* media_channel =
|
|
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
|
|
// Call GetSources with `kSsrcY` which doesn't exist.
|
|
std::vector<webrtc::RtpSource> sources = media_channel->GetSources(kSsrcY);
|
|
EXPECT_EQ(0u, sources.size());
|
|
}
|
|
|
|
// Tests that the library initializes and shuts down properly.
|
|
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
|
|
rtc::AutoThread main_thread;
|
|
for (bool use_null_apm : {false, true}) {
|
|
// If the VoiceEngine wants to gather available codecs early, that's fine
|
|
// but we never want it to create a decoder at this stage.
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
|
|
webrtc::test::MockAudioDeviceModule::CreateNice();
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(),
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
|
|
nullptr, field_trials);
|
|
engine.Init();
|
|
webrtc::RtcEventLogNull event_log;
|
|
webrtc::Call::Config call_config(&event_log);
|
|
call_config.trials = &field_trials;
|
|
call_config.task_queue_factory = task_queue_factory.get();
|
|
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
|
|
cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
|
|
call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
|
|
webrtc::CryptoOptions());
|
|
EXPECT_TRUE(channel != nullptr);
|
|
delete channel;
|
|
}
|
|
}
|
|
|
|
// Tests that reference counting on the external ADM is correct.
|
|
TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
|
|
rtc::AutoThread main_thread;
|
|
for (bool use_null_apm : {false, true}) {
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
auto adm = rtc::make_ref_counted<
|
|
::testing::NiceMock<webrtc::test::MockAudioDeviceModule>>();
|
|
{
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(),
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
|
|
nullptr, field_trials);
|
|
engine.Init();
|
|
webrtc::RtcEventLogNull event_log;
|
|
webrtc::Call::Config call_config(&event_log);
|
|
call_config.trials = &field_trials;
|
|
call_config.task_queue_factory = task_queue_factory.get();
|
|
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
|
|
cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
|
|
call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
|
|
webrtc::CryptoOptions());
|
|
EXPECT_TRUE(channel != nullptr);
|
|
delete channel;
|
|
}
|
|
// The engine/channel should have dropped their references.
|
|
EXPECT_EQ(adm.release()->Release(),
|
|
rtc::RefCountReleaseStatus::kDroppedLastRef);
|
|
}
|
|
}
|
|
|
|
// Verify the payload id of common audio codecs, including CN, ISAC, and G722.
|
|
TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
|
|
for (bool use_null_apm : {false, true}) {
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
// TODO(ossu): Why are the payload types of codecs with non-static payload
|
|
// type assignments checked here? It shouldn't really matter.
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
|
|
webrtc::test::MockAudioDeviceModule::CreateNice();
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(),
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
|
|
nullptr, field_trials);
|
|
engine.Init();
|
|
for (const cricket::AudioCodec& codec : engine.send_codecs()) {
|
|
auto is_codec = [&codec](const char* name, int clockrate = 0) {
|
|
return absl::EqualsIgnoreCase(codec.name, name) &&
|
|
(clockrate == 0 || codec.clockrate == clockrate);
|
|
};
|
|
if (is_codec("CN", 16000)) {
|
|
EXPECT_EQ(105, codec.id);
|
|
} else if (is_codec("CN", 32000)) {
|
|
EXPECT_EQ(106, codec.id);
|
|
} else if (is_codec("ISAC", 16000)) {
|
|
EXPECT_EQ(103, codec.id);
|
|
} else if (is_codec("ISAC", 32000)) {
|
|
EXPECT_EQ(104, codec.id);
|
|
} else if (is_codec("G722", 8000)) {
|
|
EXPECT_EQ(9, codec.id);
|
|
} else if (is_codec("telephone-event", 8000)) {
|
|
EXPECT_EQ(126, codec.id);
|
|
// TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
|
|
// Remove these checks once both send and receive side assigns payload
|
|
// types dynamically.
|
|
} else if (is_codec("telephone-event", 16000)) {
|
|
EXPECT_EQ(113, codec.id);
|
|
} else if (is_codec("telephone-event", 32000)) {
|
|
EXPECT_EQ(112, codec.id);
|
|
} else if (is_codec("telephone-event", 48000)) {
|
|
EXPECT_EQ(110, codec.id);
|
|
} else if (is_codec("opus")) {
|
|
EXPECT_EQ(111, codec.id);
|
|
ASSERT_TRUE(codec.params.find("minptime") != codec.params.end());
|
|
EXPECT_EQ("10", codec.params.find("minptime")->second);
|
|
ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end());
|
|
EXPECT_EQ("1", codec.params.find("useinbandfec")->second);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Tests that VoE supports at least 32 channels
|
|
TEST(WebRtcVoiceEngineTest, Has32Channels) {
|
|
rtc::AutoThread main_thread;
|
|
for (bool use_null_apm : {false, true}) {
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
|
|
webrtc::test::MockAudioDeviceModule::CreateNice();
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(),
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
|
|
nullptr, field_trials);
|
|
engine.Init();
|
|
webrtc::RtcEventLogNull event_log;
|
|
webrtc::Call::Config call_config(&event_log);
|
|
call_config.trials = &field_trials;
|
|
call_config.task_queue_factory = task_queue_factory.get();
|
|
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
|
|
|
|
cricket::VoiceMediaChannel* channels[32];
|
|
size_t num_channels = 0;
|
|
while (num_channels < arraysize(channels)) {
|
|
cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
|
|
call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
|
|
webrtc::CryptoOptions());
|
|
if (!channel)
|
|
break;
|
|
channels[num_channels++] = channel;
|
|
}
|
|
|
|
size_t expected = arraysize(channels);
|
|
EXPECT_EQ(expected, num_channels);
|
|
|
|
while (num_channels > 0) {
|
|
delete channels[--num_channels];
|
|
}
|
|
}
|
|
}
|
|
|
|
// Test that we set our preferred codecs properly.
|
|
TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
|
|
rtc::AutoThread main_thread;
|
|
for (bool use_null_apm : {false, true}) {
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
// TODO(ossu): I'm not sure of the intent of this test. It's either:
|
|
// - Check that our builtin codecs are usable by Channel.
|
|
// - The codecs provided by the engine is usable by Channel.
|
|
// It does not check that the codecs in the RecvParameters are actually
|
|
// what we sent in - though it's probably reasonable to expect so, if
|
|
// SetRecvParameters returns true.
|
|
// I think it will become clear once audio decoder injection is completed.
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
|
|
webrtc::test::MockAudioDeviceModule::CreateNice();
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(),
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
|
|
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr,
|
|
field_trials);
|
|
engine.Init();
|
|
webrtc::RtcEventLogNull event_log;
|
|
webrtc::Call::Config call_config(&event_log);
|
|
call_config.trials = &field_trials;
|
|
call_config.task_queue_factory = task_queue_factory.get();
|
|
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
|
|
cricket::WebRtcVoiceMediaChannel channel(
|
|
&engine, cricket::MediaConfig(), cricket::AudioOptions(),
|
|
webrtc::CryptoOptions(), call.get());
|
|
cricket::AudioRecvParameters parameters;
|
|
parameters.codecs = engine.recv_codecs();
|
|
EXPECT_TRUE(channel.SetRecvParameters(parameters));
|
|
}
|
|
}
|
|
|
|
TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
|
|
for (bool use_null_apm : {false, true}) {
|
|
std::vector<webrtc::AudioCodecSpec> specs;
|
|
webrtc::AudioCodecSpec spec1{{"codec1", 48000, 2, {{"param1", "value1"}}},
|
|
{48000, 2, 16000, 10000, 20000}};
|
|
spec1.info.allow_comfort_noise = false;
|
|
spec1.info.supports_network_adaption = true;
|
|
specs.push_back(spec1);
|
|
webrtc::AudioCodecSpec spec2{{"codec2", 32000, 1}, {32000, 1, 32000}};
|
|
spec2.info.allow_comfort_noise = false;
|
|
specs.push_back(spec2);
|
|
specs.push_back(webrtc::AudioCodecSpec{
|
|
{"codec3", 16000, 1, {{"param1", "value1b"}, {"param2", "value2"}}},
|
|
{16000, 1, 13300}});
|
|
specs.push_back(
|
|
webrtc::AudioCodecSpec{{"codec4", 8000, 1}, {8000, 1, 64000}});
|
|
specs.push_back(
|
|
webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}});
|
|
|
|
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
|
|
webrtc::CreateDefaultTaskQueueFactory();
|
|
rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory =
|
|
webrtc::MockAudioEncoderFactory::CreateUnusedFactory();
|
|
rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
|
|
rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
|
|
EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders())
|
|
.WillOnce(Return(specs));
|
|
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
|
|
webrtc::test::MockAudioDeviceModule::CreateNice();
|
|
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
|
|
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
|
|
webrtc::FieldTrialBasedConfig field_trials;
|
|
cricket::WebRtcVoiceEngine engine(
|
|
task_queue_factory.get(), adm.get(), unused_encoder_factory,
|
|
mock_decoder_factory, nullptr, apm, nullptr, field_trials);
|
|
engine.Init();
|
|
auto codecs = engine.recv_codecs();
|
|
EXPECT_EQ(11u, codecs.size());
|
|
|
|
// Rather than just ASSERTing that there are enough codecs, ensure that we
|
|
// can check the actual values safely, to provide better test results.
|
|
auto get_codec = [&codecs](size_t index) -> const cricket::AudioCodec& {
|
|
static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0);
|
|
if (codecs.size() > index)
|
|
return codecs[index];
|
|
return missing_codec;
|
|
};
|
|
|
|
// Ensure the general codecs are generated first and in order.
|
|
for (size_t i = 0; i != specs.size(); ++i) {
|
|
EXPECT_EQ(specs[i].format.name, get_codec(i).name);
|
|
EXPECT_EQ(specs[i].format.clockrate_hz, get_codec(i).clockrate);
|
|
EXPECT_EQ(specs[i].format.num_channels, get_codec(i).channels);
|
|
EXPECT_EQ(specs[i].format.parameters, get_codec(i).params);
|
|
}
|
|
|
|
// Find the index of a codec, or -1 if not found, so that we can easily
|
|
// check supplementary codecs are ordered after the general codecs.
|
|
auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int {
|
|
for (size_t i = 0; i != codecs.size(); ++i) {
|
|
const cricket::AudioCodec& codec = codecs[i];
|
|
if (absl::EqualsIgnoreCase(codec.name, format.name) &&
|
|
codec.clockrate == format.clockrate_hz &&
|
|
codec.channels == format.num_channels) {
|
|
return rtc::checked_cast<int>(i);
|
|
}
|
|
}
|
|
return -1;
|
|
};
|
|
|
|
// Ensure all supplementary codecs are generated last. Their internal
|
|
// ordering is not important. Without this cast, the comparison turned
|
|
// unsigned and, thus, failed for -1.
|
|
const int num_specs = static_cast<int>(specs.size());
|
|
EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
|
|
EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
|
|
EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
|
|
EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
|
|
EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
|
|
EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
|
|
EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
|
|
}
|
|
}
|