Allow applications to control audio send bitrate through 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 196 matching lines @@
     parameters.codecs.push_back(codec);
     parameters.max_bandwidth_bps = max_bitrate;
     EXPECT_EQ(expected_result, channel_->SetSendParameters(parameters));
 
     int channel_num = voe_.GetLastChannel();
     webrtc::CodecInst temp_codec;
     EXPECT_FALSE(voe_.GetSendCodec(channel_num, temp_codec));
     EXPECT_EQ(expected_bitrate, temp_codec.rate);
   }
 
+  // Sets the per-stream maximum bitrate limit for the specified SSRC.
+  bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) {
+    webrtc::RtpParameters parameters = channel_->GetRtpParameters(ssrc);
+    EXPECT_EQ(1UL, parameters.encodings.size());
+
+    parameters.encodings[0].max_bitrate_bps = bitrate;
+    return channel_->SetRtpParameters(ssrc, parameters);
+  }
+
+  bool SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
+    cricket::AudioSendParameters send_parameters;
+    send_parameters.codecs.push_back(codec);
+    send_parameters.max_bandwidth_bps = bitrate;
+    return channel_->SetSendParameters(send_parameters);
+  }
+
+  int GetCodecBitrate(int32_t ssrc) {
+    cricket::WebRtcVoiceMediaChannel* media_channel =
+        static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+    int channel = media_channel->GetSendChannelId(ssrc);
+    EXPECT_NE(-1, channel);
+    webrtc::CodecInst codec;
+    EXPECT_FALSE(voe_.GetSendCodec(channel, codec));
+    return codec.rate;
+  }
+
+  void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec,
+                              int global_max,
+                              int stream_max,
+                              bool expected_result,
+                              int expected_codec_bitrate) {
+    // Clear the bitrate limit from the previous test case.
+    EXPECT_TRUE(SetMaxBitrateForStream(kSsrc1, -1));
+
+    // Attempt to set the requested bitrate limits.
+    EXPECT_TRUE(SetGlobalMaxBitrate(codec, global_max));
+    EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrc1, stream_max));
+
+    // Verify that reading back the parameters gives results
+    // consistent with the Set() result.
+    webrtc::RtpParameters resulting_parameters =
+        channel_->GetRtpParameters(kSsrc1);
+    EXPECT_EQ(1UL, resulting_parameters.encodings.size());
+    EXPECT_EQ(expected_result ? stream_max : -1,
+              resulting_parameters.encodings[0].max_bitrate_bps);
+
+    // Verify that the codec settings have the expected bitrate.
+    EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrc1));
+  }
+
   void TestSetSendRtpHeaderExtensions(const std::string& ext) {
     EXPECT_TRUE(SetupSendStream());
 
     // Ensure extensions are off by default.
     EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
 
     // Ensure unknown extensions won't cause an error.
     send_parameters_.extensions.push_back(cricket::RtpHeaderExtension(
         "urn:ietf:params:unknownextention", 1));
     EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
@@ skipping 538 matching lines @@
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
   EXPECT_EQ(64000, codec.rate);
 
   send_parameters_.max_bandwidth_bps = 128;
   EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
   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(SetupSendStream());
+
+  // opus, default bitrate == 64000.
+  SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 64000);
+  SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000);
+  SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000);
+  SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000);
+
+  // CBR codecs allow both maximums to exceed the bitrate.
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000);
+
+  // CBR codecs don't allow per stream maximums to be too low.
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000);
+}
+
+// Test that an attempt to set RtpParameters for a stream that does not exist
+// fails.
+TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
+  EXPECT_TRUE(SetupChannel());
+  webrtc::RtpParameters nonexistent_parameters =
+      channel_->GetRtpParameters(kSsrc1);
+  EXPECT_EQ(0, nonexistent_parameters.encodings.size());
+
+  nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, nonexistent_parameters));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake,
+       CannotSetRtpParametersWithIncorrectNumberOfEncodings) {
+  // This test verifies that setting RtpParameters succeeds only if
+  // the structure contains exactly one encoding.
+  // TODO(skvlad): Update this test when we start supporting setting parameters
+  // for each encoding individually.
+
+  EXPECT_TRUE(SetupSendStream());
+  // Setting RtpParameters with no encoding is expected to fail.
+  webrtc::RtpParameters parameters;
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
+  // Setting RtpParameters with exactly one encoding should succeed.
+  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_TRUE(channel_->SetRtpParameters(kSsrc1, parameters));
+  // Two or more encodings should result in failure.
+  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpParameters(kSsrc1, parameters));
+}
+
+// Test that SetRtpParameters configures the correct encoding channel for each
+// SSRC.
+TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
+  SetupForMultiSendStream();
+  // Create send streams.
+  for (uint32_t ssrc : kSsrcs4) {
+    EXPECT_TRUE(
+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
+  }
+  // Configure one stream to be limited by the stream config, another to be
+  // limited by the global max, and the third one with no per-stream limit
+  // (still subject to the global limit).
+  EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, 64000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 48000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 96000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1));
+
+  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+
+  // Remove the global cap; the streams should switch to their respective
+  // maximums (or remain unchanged if there was no other limit on them.)
+  EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, -1));
+  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(96000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+}
+
 // Test that we apply codecs properly.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
   parameters.codecs.push_back(kRedCodec);
   parameters.codecs[0].id = 96;
   parameters.codecs[0].bitrate = 48000;
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
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 TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
   cricket::WebRtcVoiceEngine engine(nullptr);
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config()));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }