Set max bitrate for audio send stream based on RtpParameters.

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 1194 matching lines @@
   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));
 
   // ... 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_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
+  webrtc::test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-SendSideBwe/Enabled/");
+  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 =
+      rtc::Optional<int>(kMaxBitrateBps);
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
+
+  const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
+  EXPECT_EQ(max_bitrate, kMaxBitrateBps);
+}
+
 // Test that GetRtpReceiveParameters returns the currently configured codecs.
 TEST_F(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);
@@ skipping 1393 matching lines @@
   parameters.codecs.push_back(kOpusCodec);
   SetSendParameters(parameters);
   const int initial_num = call_.GetNumCreatedSendStreams();
   EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
 
   // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
   constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
   constexpr int kOpusMaxPtimeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
   constexpr int kMinOverheadBps =
       kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs;
-  constexpr int kMaxOverheadBps = kOverheadPerPacket * 8 * 1000 / 10;
 
   constexpr int kOpusMinBitrateBps = 6000;
   EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadBps,
             GetSendStreamConfig(kSsrcX).min_bitrate_bps);
   constexpr int kOpusBitrateFbBps = 32000;
-  EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadBps,
+  EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadBps,

[minyue-webrtc, 2017/03/27 19:35:25]

changed the behavior intentionally.

             GetSendStreamConfig(kSsrcX).max_bitrate_bps);
 
   parameters.options.audio_network_adaptor = rtc::Optional<bool>(true);
   parameters.options.audio_network_adaptor_config =
       rtc::Optional<std::string>("1234");
   SetSendParameters(parameters);
 
   constexpr int kMinOverheadWithAnaBps =
       kOverheadPerPacket * 8 * 1000 / kOpusMaxPtimeMs;
-  constexpr int kMaxOverheadWithAnaBps = kOverheadPerPacket * 8 * 1000 / 20;
 
   EXPECT_EQ(kOpusMinBitrateBps + kMinOverheadWithAnaBps,
             GetSendStreamConfig(kSsrcX).min_bitrate_bps);
 
-  EXPECT_EQ(kOpusBitrateFbBps + kMaxOverheadWithAnaBps,
+  EXPECT_EQ(kOpusBitrateFbBps + kMinOverheadWithAnaBps,
             GetSendStreamConfig(kSsrcX).max_bitrate_bps);
 }
 
+// This test is similar to
+// WebRtcVoiceEngineTestFake.SetRtpSendParameterUpdatesMaxBitrate but with an
+// additional field trial.
+TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest,
+       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 =
+      rtc::Optional<int>(kMaxBitrateBps);
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
+
+  const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  constexpr int kMinOverhead = 3333;
+#else
+  constexpr int kMinOverhead = 6666;
+#endif
+  EXPECT_EQ(max_bitrate, kMaxBitrateBps + kMinOverhead);
+}
+
 // Test that we can set the outgoing SSRC properly.
 // SSRC is set in SetupSendStream() by calling AddSendStream.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
   // Setup. We need send codec to be set to get all stats.
   EXPECT_TRUE(SetupSendStream());
@@ skipping 1173 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);
 }