Remove voe_auto_test and add new tests to cover the missing cases.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3007383002
Cr-Commit-Position: refs/heads/master@{#19865}
diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
new file mode 100644
index 0000000..4283b73
--- /dev/null
+++ b/audio/audio_send_stream_tests.cc
@@ -0,0 +1,238 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/call_test.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+class AudioSendTest : public SendTest {
+ public:
+ AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
+
+ size_t GetNumVideoStreams() const override {
+ return 0;
+ }
+ size_t GetNumAudioStreams() const override {
+ return 1;
+ }
+ size_t GetNumFlexfecStreams() const override {
+ return 0;
+ }
+};
+} // namespace
+
+using AudioSendStreamCallTest = CallTest;
+
+TEST_F(AudioSendStreamCallTest, SupportsCName) {
+ static std::string kCName = "PjqatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
+ class CNameObserver : public AudioSendTest {
+ public:
+ CNameObserver() = default;
+
+ private:
+ Action OnSendRtcp(const uint8_t* packet, size_t length) override {
+ RtcpPacketParser parser;
+ EXPECT_TRUE(parser.Parse(packet, length));
+ if (parser.sdes()->num_packets() > 0) {
+ EXPECT_EQ(1u, parser.sdes()->chunks().size());
+ EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
+
+ observation_complete_.Set();
+ }
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.c_name = kCName;
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) {
+ class NoExtensionsObserver : public AudioSendTest {
+ public:
+ NoExtensionsObserver() = default;
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+ EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
+ EXPECT_FALSE(header.extension.hasTransportSequenceNumber);
+ EXPECT_FALSE(header.extension.hasAudioLevel);
+ EXPECT_FALSE(header.extension.hasVideoRotation);
+ EXPECT_FALSE(header.extension.hasVideoContentType);
+ observation_complete_.Set();
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
+ class AudioLevelObserver : public AudioSendTest {
+ public:
+ AudioLevelObserver() : AudioSendTest() {
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, test::kAudioLevelExtensionId));
+ }
+
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_TRUE(header.extension.hasAudioLevel);
+ if (header.extension.audioLevel != 0) {
+ // Wait for at least one packet with a non-zero level.
+ observation_complete_.Set();
+ } else {
+ LOG(LS_WARNING) << "Got a packet with zero audioLevel - waiting"
+ " for another packet...";
+ }
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAudioLevelUri, test::kAudioLevelExtensionId));
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SupportsTransportWideSequenceNumbers) {
+ static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+ class TransportWideSequenceNumberObserver : public AudioSendTest {
+ public:
+ TransportWideSequenceNumberObserver() : AudioSendTest() {
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransportSequenceNumber, kExtensionId));
+ }
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+ EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+ EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
+
+ observation_complete_.Set();
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+ }
+ } test;
+
+ RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SendDtmf) {
+ static const uint8_t kDtmfPayloadType = 120;
+ static const int kDtmfPayloadFrequency = 8000;
+ static const int kDtmfEventFirst = 12;
+ static const int kDtmfEventLast = 31;
+ static const int kDtmfDuration = 50;
+ class DtmfObserver : public AudioSendTest {
+ public:
+ DtmfObserver() = default;
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ if (header.payloadType == kDtmfPayloadType) {
+ EXPECT_EQ(12u, header.headerLength);
+ EXPECT_EQ(16u, length);
+ const int event = packet[12];
+ if (event != expected_dtmf_event_) {
+ ++expected_dtmf_event_;
+ EXPECT_EQ(event, expected_dtmf_event_);
+ if (expected_dtmf_event_ == kDtmfEventLast) {
+ observation_complete_.Set();
+ }
+ }
+ }
+
+ return SEND_PACKET;
+ }
+
+ void OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStream*>& receive_streams) override {
+ // Need to start stream here, else DTMF events are dropped.
+ send_stream->Start();
+ for (int event = kDtmfEventFirst; event <= kDtmfEventLast; ++event) {
+ send_stream->SendTelephoneEvent(kDtmfPayloadType, kDtmfPayloadFrequency,
+ event, kDtmfDuration);
+ }
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for DTMF stream.";
+ }
+
+ int expected_dtmf_event_ = kDtmfEventFirst;
+ } test;
+
+ RunBaseTest(&test);
+}
+
+} // namespace test
+} // namespace webrtc