Replace voe_auto_test

webrtc/audio/audio_send_stream_tests.cc

+/*
+ *  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 "webrtc/test/call_test.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/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_) {
+          if (expected_dtmf_event_ == kDtmfEventLast) {
+            observation_complete_.Set();
+          }
+        } else {

[ossu, 2017/09/15 12:20:21]

So, now we only advance the expected dtmf event if we're no longer getting the
one we previously expected. This vibes with your explanation of the RFC (I
didn't know it could do that).

However, now if we get kDtmfEventLast only once, we'll never trigger
observation_complete_. Perhaps invert the check:
if(event != expected_dtmf_event_) {
/* advance and EXPECT_EQ */
}
if (event == kDtmfEventLast) observation_complete_.Set();

[the sun, 2017/09/15 13:08:26]

Yes. Good catch!

+          ++expected_dtmf_event_;
+          EXPECT_EQ(event, expected_dtmf_event_);
+        }
+      }
+
+      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