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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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205 std::vector<AudioReceiveStream::Config>* receive_configs) { | 205 std::vector<AudioReceiveStream::Config>* receive_configs) { |
206 if (num_audio_streams_ == 0) | 206 if (num_audio_streams_ == 0) |
207 return; | 207 return; |
208 | 208 |
209 EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_) | 209 EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_) |
210 << "Audio BWE not supported with toffset."; | 210 << "Audio BWE not supported with toffset."; |
211 | 211 |
212 send_config->rtp.ssrc = audio_ssrcs_[0]; | 212 send_config->rtp.ssrc = audio_ssrcs_[0]; |
213 send_config->rtp.extensions.clear(); | 213 send_config->rtp.extensions.clear(); |
214 | 214 |
215 send_config->min_bitrate_kbps = 6; | 215 send_config->min_bitrate_bps = 6000; |
216 send_config->max_bitrate_kbps = 60; | 216 send_config->max_bitrate_bps = 60000; |
217 | 217 |
218 bool transport_cc = false; | 218 bool transport_cc = false; |
219 if (extension_type_ == RtpExtension::kAbsSendTimeUri) { | 219 if (extension_type_ == RtpExtension::kAbsSendTimeUri) { |
220 transport_cc = false; | 220 transport_cc = false; |
221 send_config->rtp.extensions.push_back( | 221 send_config->rtp.extensions.push_back( |
222 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); | 222 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); |
223 } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { | 223 } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { |
224 transport_cc = true; | 224 transport_cc = true; |
225 send_config->rtp.extensions.push_back(RtpExtension( | 225 send_config->rtp.extensions.push_back(RtpExtension( |
226 extension_type_.c_str(), kTransportSequenceNumberExtensionId)); | 226 extension_type_.c_str(), kTransportSequenceNumberExtensionId)); |
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620 true); | 620 true); |
621 RunBaseTest(&test); | 621 RunBaseTest(&test); |
622 } | 622 } |
623 | 623 |
624 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) { | 624 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) { |
625 RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps, | 625 RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps, |
626 RtpExtension::kTransportSequenceNumberUri, false, false); | 626 RtpExtension::kTransportSequenceNumberUri, false, false); |
627 RunBaseTest(&test); | 627 RunBaseTest(&test); |
628 } | 628 } |
629 } // namespace webrtc | 629 } // namespace webrtc |
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