Store/restore RTP state for audio streams with same SSRC within a call

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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 <list>
 #include <map>
 #include <memory>
 #include <utility>
 
+#include "webrtc/api/test/mock_audio_mixer.h"
 #include "webrtc/base/ptr_util.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
@@ skipping 94 matching lines @@
       call->DestroyAudioReceiveStream(s);
     }
     streams.clear();
   }
 }
 
 TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
       new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
   CallHelper call(decoder_factory);
+  ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp;
 
   constexpr int kRecvChannelId = 101;
 
   // Set up the mock to create a channel proxy which we know of, so that we can
   // add our expectations to it.
   test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
   EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
       .WillRepeatedly(testing::Invoke([&](int channel_id) {
         test::MockVoEChannelProxy* channel_proxy =
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
         EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
             .WillRepeatedly(testing::Invoke(
                 [](const std::map<int, SdpAudioFormat>& codecs) {
                   EXPECT_THAT(codecs, testing::IsEmpty());
                 }));
+        EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_))
+            .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {
           EXPECT_FALSE(recv_channel_proxy);
           recv_channel_proxy = channel_proxy;
         }
         return channel_proxy;
       }));
 
   AudioReceiveStream::Config recv_config;
@@ skipping 15 matching lines @@
   call->DestroyAudioSendStream(send_stream);
 
   EXPECT_CALL(*recv_channel_proxy, DisassociateSendChannel()).Times(1);
   call->DestroyAudioReceiveStream(recv_stream);
 }
 
 TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
       new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
   CallHelper call(decoder_factory);
+  ::testing::NiceMock<MockRtpRtcp> mock_rtp_rtcp;
 
   constexpr int kRecvChannelId = 101;
 
   // Set up the mock to create a channel proxy which we know of, so that we can
   // add our expectations to it.
   test::MockVoEChannelProxy* recv_channel_proxy = nullptr;
   EXPECT_CALL(*call.voice_engine(), ChannelProxyFactory(testing::_))
       .WillRepeatedly(testing::Invoke([&](int channel_id) {
         test::MockVoEChannelProxy* channel_proxy =
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
         EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
             .WillRepeatedly(testing::Invoke(
                 [](const std::map<int, SdpAudioFormat>& codecs) {
                   EXPECT_THAT(codecs, testing::IsEmpty());
                 }));
+        EXPECT_CALL(*channel_proxy, GetRtpRtcp(testing::_, testing::_))
+            .WillRepeatedly(testing::SetArgPointee<0>(&mock_rtp_rtcp));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {
           EXPECT_FALSE(recv_channel_proxy);
           recv_channel_proxy = channel_proxy;
           // We need to set this expectation here since the channel proxy is
           // created as a side effect of CreateAudioReceiveStream().
           EXPECT_CALL(*recv_channel_proxy,
                       AssociateSendChannel(testing::_)).Times(1);
         }
@@ skipping 193 matching lines @@
   bitrate_config.start_bitrate_bps = 2;
   bitrate_config.max_bitrate_bps = 3;
 
   EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)).Times(1);
   call->SetBitrateConfig(bitrate_config);
 
   bitrate_config.start_bitrate_bps = -1;
   call->SetBitrateConfig(bitrate_config);
 }
 
+TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
+  constexpr uint32_t kSSRC = 12345;
+  testing::NiceMock<test::MockAudioDeviceModule> mock_adm;
+  // Reply with a 10ms timer every time TimeUntilNextProcess is called to
+  // avoid entering a tight loop on the process thread.
+  EXPECT_CALL(mock_adm, TimeUntilNextProcess())
+       .WillRepeatedly(testing::Return(10));
+  rtc::scoped_refptr<test::MockAudioMixer> mock_mixer(
+      new rtc::RefCountedObject<test::MockAudioMixer>);
+
+  // There's similar functionality in cricket::VoEWrapper but it's not reachable
+  // from here. Since we're working on removing VoE interfaces, I doubt it's
+  // worth making VoEWrapper more easily available.
+  struct ScopedVoiceEngine {
+    ScopedVoiceEngine()
+        : voe(VoiceEngine::Create()),
+          base(VoEBase::GetInterface(voe)) {}
+    ~ScopedVoiceEngine() {
+      base->Release();
+      EXPECT_TRUE(VoiceEngine::Delete(voe));
+    }
+
+    VoiceEngine* voe;
+    VoEBase* base;
+  };
+  ScopedVoiceEngine voice_engine;
+
+  voice_engine.base->Init(&mock_adm);
+  AudioState::Config audio_state_config;
+  audio_state_config.voice_engine = voice_engine.voe;
+  audio_state_config.audio_mixer = mock_mixer;
+  auto audio_state = AudioState::Create(audio_state_config);
+  RtcEventLogNullImpl event_log;
+  Call::Config call_config(&event_log);
+  call_config.audio_state = audio_state;
+  std::unique_ptr<Call> call(Call::Create(call_config));
+
+  auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
+    AudioSendStream::Config config(nullptr);
+    config.rtp.ssrc = ssrc;
+    config.voe_channel_id = voice_engine.base->CreateChannel();
+    AudioSendStream* stream = call->CreateAudioSendStream(config);
+    VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine.voe);
+    auto channel_proxy = voe_impl->GetChannelProxy(config.voe_channel_id);
+    RtpRtcp* rtp_rtcp = nullptr;
+    RtpReceiver* rtp_receiver = nullptr;  // Unused but required for call.
+    channel_proxy->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
+    const RtpState rtp_state = rtp_rtcp->GetRtpState();
+    call->DestroyAudioSendStream(stream);
+    voice_engine.base->DeleteChannel(config.voe_channel_id);
+    return rtp_state;
+  };
+
+  const RtpState rtp_state1 = create_stream_and_get_rtp_state(kSSRC);
+  const RtpState rtp_state2 = create_stream_and_get_rtp_state(kSSRC);
+
+  EXPECT_EQ(rtp_state1.sequence_number, rtp_state2.sequence_number);
+  EXPECT_EQ(rtp_state1.start_timestamp, rtp_state2.start_timestamp);
+  EXPECT_EQ(rtp_state1.timestamp, rtp_state2.timestamp);
+  EXPECT_EQ(rtp_state1.capture_time_ms, rtp_state2.capture_time_ms);
+  EXPECT_EQ(rtp_state1.last_timestamp_time_ms,
+            rtp_state2.last_timestamp_time_ms);
+  EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
+}
+
 }  // namespace webrtc