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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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| 226 audio_send_config.voe_channel_id = send_channel_id; | 226 audio_send_config.voe_channel_id = send_channel_id; |
| 227 audio_send_config.rtp.ssrc = kAudioSendSsrc; | 227 audio_send_config.rtp.ssrc = kAudioSendSsrc; |
| 228 AudioSendStream* audio_send_stream = | 228 AudioSendStream* audio_send_stream = |
| 229 sender_call_->CreateAudioSendStream(audio_send_config); | 229 sender_call_->CreateAudioSendStream(audio_send_config); |
| 230 | 230 |
| 231 CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; | 231 CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| 232 EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac)); | 232 EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac)); |
| 233 | 233 |
| 234 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 234 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| 235 if (fec == FecMode::kOn) { | 235 if (fec == FecMode::kOn) { |
| 236 video_send_config_.rtp.fec.red_payload_type = kRedPayloadType; | 236 video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType; |
| 237 video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; | 237 video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; |
| 238 video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; | 238 video_receive_configs_[0].rtp.ulpfec.red_payload_type = kRedPayloadType; |
| 239 video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; | 239 video_receive_configs_[0].rtp.ulpfec.ulpfec_payload_type = |
| 240 kUlpfecPayloadType; |
| 240 } | 241 } |
| 241 video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000; | 242 video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000; |
| 242 video_receive_configs_[0].renderer = &observer; | 243 video_receive_configs_[0].renderer = &observer; |
| 243 video_receive_configs_[0].sync_group = kSyncGroup; | 244 video_receive_configs_[0].sync_group = kSyncGroup; |
| 244 | 245 |
| 245 AudioReceiveStream::Config audio_recv_config; | 246 AudioReceiveStream::Config audio_recv_config; |
| 246 audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; | 247 audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; |
| 247 audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; | 248 audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; |
| 248 audio_recv_config.voe_channel_id = recv_channel_id; | 249 audio_recv_config.voe_channel_id = recv_channel_id; |
| 249 audio_recv_config.sync_group = kSyncGroup; | 250 audio_recv_config.sync_group = kSyncGroup; |
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| 726 uint32_t last_set_bitrate_; | 727 uint32_t last_set_bitrate_; |
| 727 VideoSendStream* send_stream_; | 728 VideoSendStream* send_stream_; |
| 728 test::FrameGeneratorCapturer* frame_generator_; | 729 test::FrameGeneratorCapturer* frame_generator_; |
| 729 VideoEncoderConfig encoder_config_; | 730 VideoEncoderConfig encoder_config_; |
| 730 } test; | 731 } test; |
| 731 | 732 |
| 732 RunBaseTest(&test); | 733 RunBaseTest(&test); |
| 733 } | 734 } |
| 734 | 735 |
| 735 } // namespace webrtc | 736 } // namespace webrtc |
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