Be less pessimistic about turning "default" receive streams into signaled streams.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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.
  */
 
@@ skipping 2796 matching lines @@
 
   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(2, 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_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) {
+  EXPECT_TRUE(SetupChannel());
+
+  // Spawn unsignaled stream with SSRC=1.
+  DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+  EXPECT_EQ(1, 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(1, streams.size());
+  int audio_receive_stream_id = streams.front()->id();
+  EXPECT_TRUE(AddRecvStream(1));
+  EXPECT_EQ(1, streams.size());
+  EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) {
+  EXPECT_TRUE(SetupChannel());
+
+  // Spawn unsignaled stream with SSRC=1.
+  DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
+  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
+                                                sizeof(kPcmuFrame)));
+
+  // Verify that the underlying stream object in Call *is* recreated when a
+  // stream with SSRC=1 is added, and which has changed stream parameters.
+  const auto& streams = call_.GetAudioReceiveStreams();
+  EXPECT_EQ(1, streams.size());
+  int audio_receive_stream_id = streams.front()->id();
+  cricket::StreamParams stream_params;
+  stream_params.ssrcs.push_back(1);
+  stream_params.sync_label = "sync_label";
+  EXPECT_TRUE(channel_->AddRecvStream(stream_params));
+  EXPECT_EQ(1, streams.size());
+  EXPECT_NE(audio_receive_stream_id, streams.front()->id());
+}
+
 // Test that we properly handle failures to add a receive stream.
 TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamFail) {
   EXPECT_TRUE(SetupChannel());
   voe_.set_fail_create_channel(true);
   EXPECT_FALSE(AddRecvStream(2));
 }
 
 // Test that we properly handle failures to add a send stream.
 TEST_F(WebRtcVoiceEngineTestFake, AddSendStreamFail) {
   EXPECT_TRUE(SetupChannel());
@@ skipping 928 matching lines @@
   // 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);
 }