Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 1e23d381cd1ca7298c3ed9d21da840910f418d6f..610b7dbbbe1ea020ff2dc7e94d5eafb0907b0659 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -200,6 +200,43 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_EQ(expected_bitrate, temp_codec.rate);
   }
 
+  void TestSendStreamBitrate(const cricket::AudioCodec& codec,
+                             int max_bitrate_global,
+                             int max_bitrate_ssrc,
+                             bool expected_result,
+                             int expected_bitrate) {
+    // clear the bitrate limit from the previous test case
+    webrtc::RTCRtpParameters rtp_parameters =
+        channel_->GetRtpParameters(kSsrc1);
+    EXPECT_EQ(1UL, rtp_parameters.encodings.size());
+    rtp_parameters.encodings[0].max_bitrate_bps = -1;
+    EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, rtp_parameters));
+
+    // Attempt to set the requested bitrate limits
+    cricket::AudioSendParameters send_parameters;
+    send_parameters.codecs.push_back(codec);
+    send_parameters.max_bandwidth_bps = max_bitrate_global;
+    EXPECT_TRUE(channel_->SetSendParameters(send_parameters));
+
+    rtp_parameters.encodings[0].max_bitrate_bps = max_bitrate_ssrc;
+    EXPECT_EQ(expected_result,
+              channel_->SetRtpParameters(kSsrc1, rtp_parameters));
+
+    // Verify that reading back the parameters gives results
+    // consistent with the Set() result
+    webrtc::RTCRtpParameters resulting_parameters =
+        channel_->GetRtpParameters(kSsrc1);
+    EXPECT_EQ(1UL, resulting_parameters.encodings.size());
+    EXPECT_EQ(expected_result ? max_bitrate_ssrc : -1,
+              resulting_parameters.encodings[0].max_bitrate_bps);
+
+    // Verify that the codec settings have the expected bitrate
+    int channel_num = voe_.GetLastChannel();
+    webrtc::CodecInst temp_codec;
+    EXPECT_FALSE(voe_.GetSendCodec(channel_num, temp_codec));
+    EXPECT_EQ(expected_bitrate, temp_codec.rate);
+  }
+
   void TestSetSendRtpHeaderExtensions(const std::string& ext) {
     EXPECT_TRUE(SetupEngineWithSendStream());
 
@@ -769,6 +806,53 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) {
   EXPECT_EQ(64000, codec.rate);
 }
 
+// Test that the per-stream bitrate limit and the global
+// bitrate limit both apply
+TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
+  EXPECT_TRUE(SetupEngineWithSendStream());
+
+  // opus, default bitrate == 64000.
+  TestSendStreamBitrate(kOpusCodec, 0, 0, true, 64000);
+  TestSendStreamBitrate(kOpusCodec, 48000, 0, true, 48000);
+  TestSendStreamBitrate(kOpusCodec, 48000, 64000, true, 48000);
+  TestSendStreamBitrate(kOpusCodec, 64000, 48000, true, 48000);
+
+  // CBR codecs allow both maximums to exceed the bitrate
+  TestSendStreamBitrate(kPcmuCodec, 0, 0, true, 64000);
+  TestSendStreamBitrate(kPcmuCodec, 64001, 0, true, 64000);
+  TestSendStreamBitrate(kPcmuCodec, 0, 64001, true, 64000);
+  TestSendStreamBitrate(kPcmuCodec, 64001, 64001, true, 64000);
+
+  // CBR codecs don't allow per stream maximums to be too low
+  TestSendStreamBitrate(kPcmuCodec, 0, 63999, false, 64000);
+  TestSendStreamBitrate(kPcmuCodec, 64001, 63999, false, 64000);
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
+  EXPECT_TRUE(SetupEngine());
+  webrtc::RTCRtpParameters nonexistent_parameters =
+      channel_->GetRtpParameters(kSsrc1);
+  EXPECT_EQ(0, nonexistent_parameters.encodings.size());
+
+  nonexistent_parameters.encodings.push_back(
+      webrtc::RTCRtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, nonexistent_parameters));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake,
+       CannotSetRtpParametersWithIncorrectNumberOfEncodings) {

[Taylor Brandstetter, 2016/03/12 01:57:06]

If you add a TODO comment for the other "IncorrectNumberOfEncodings" test, add
it for this one too.

[skvlad, 2016/03/15 21:18:18]

Done.

+  EXPECT_TRUE(SetupEngineWithSendStream());
+  // no encodings - should fail
+  webrtc::RTCRtpParameters parameters;
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
+  // one encoding - expected to succeed
+  parameters.encodings.push_back(webrtc::RTCRtpEncodingParameters());
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  // two encodings - too many
+  parameters.encodings.push_back(webrtc::RTCRtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
+}
+
 // Test that we apply codecs properly.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
   EXPECT_TRUE(SetupEngineWithSendStream());