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Issue 2705603002: Fixes a bug where a video stream can get stuck in the suspended state. (Closed)
Patch Set: . Created 3 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); 208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
209 video_send_config_.rtp.extensions.push_back(RtpExtension( 209 video_send_config_.rtp.extensions.push_back(RtpExtension(
210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); 210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
211 } 211 }
212 212
213 if (num_audio_streams > 0) { 213 if (num_audio_streams > 0) {
214 audio_send_config_ = AudioSendStream::Config(send_transport); 214 audio_send_config_ = AudioSendStream::Config(send_transport);
215 audio_send_config_.voe_channel_id = voe_send_.channel_id; 215 audio_send_config_.voe_channel_id = voe_send_.channel_id;
216 audio_send_config_.rtp.ssrc = kAudioSendSsrc; 216 audio_send_config_.rtp.ssrc = kAudioSendSsrc;
217 audio_send_config_.send_codec_spec.codec_inst = 217 audio_send_config_.send_codec_spec.codec_inst =
218 CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; 218 CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
219 } 219 }
220 220
221 // TODO(brandtr): Update this when we support multistream protection. 221 // TODO(brandtr): Update this when we support multistream protection.
222 if (num_flexfec_streams > 0) { 222 if (num_flexfec_streams > 0) {
223 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; 223 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
224 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; 224 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
225 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; 225 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
226 } 226 }
227 } 227 }
228 228
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495 495
496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
497 } 497 }
498 498
499 bool EndToEndTest::ShouldCreateReceivers() const { 499 bool EndToEndTest::ShouldCreateReceivers() const {
500 return true; 500 return true;
501 } 501 }
502 502
503 } // namespace test 503 } // namespace test
504 } // namespace webrtc 504 } // namespace webrtc
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