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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 #include "webrtc/video/video_quality_test.h" | 10 #include "webrtc/video/video_quality_test.h" |
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1721 // Add extension to enable audio send side BWE, and allow audio bit rate | 1721 // Add extension to enable audio send side BWE, and allow audio bit rate |
1722 // adaptation. | 1722 // adaptation. |
1723 audio_send_config_.rtp.extensions.clear(); | 1723 audio_send_config_.rtp.extensions.clear(); |
1724 if (params_.call.send_side_bwe) { | 1724 if (params_.call.send_side_bwe) { |
1725 audio_send_config_.rtp.extensions.push_back( | 1725 audio_send_config_.rtp.extensions.push_back( |
1726 webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, | 1726 webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, |
1727 test::kTransportSequenceNumberExtensionId)); | 1727 test::kTransportSequenceNumberExtensionId)); |
1728 audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps; | 1728 audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps; |
1729 audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps; | 1729 audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps; |
1730 } | 1730 } |
1731 audio_send_config_.send_codec_spec.codec_inst = | 1731 audio_send_config_.send_codec_spec = |
1732 CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000}; | 1732 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
1733 audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx; | 1733 {kAudioSendPayloadType, |
| 1734 {"OPUS", 48000, 2, |
| 1735 {{"usedtx", (params_.audio.dtx ? "1" : "0")}, |
| 1736 {"stereo", "1"}}}}); |
| 1737 |
1734 audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_); | 1738 audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_); |
1735 | 1739 |
1736 AudioReceiveStream::Config audio_config; | 1740 AudioReceiveStream::Config audio_config; |
1737 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; | 1741 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; |
1738 audio_config.rtcp_send_transport = transport; | 1742 audio_config.rtcp_send_transport = transport; |
1739 audio_config.voe_channel_id = receive_channel_id; | 1743 audio_config.voe_channel_id = receive_channel_id; |
1740 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; | 1744 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; |
1741 audio_config.rtp.transport_cc = params_.call.send_side_bwe; | 1745 audio_config.rtp.transport_cc = params_.call.send_side_bwe; |
1742 audio_config.rtp.extensions = audio_send_config_.rtp.extensions; | 1746 audio_config.rtp.extensions = audio_send_config_.rtp.extensions; |
1743 audio_config.decoder_factory = decoder_factory_; | 1747 audio_config.decoder_factory = decoder_factory_; |
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1902 if (!params_.video.encoded_frame_base_path.empty()) { | 1906 if (!params_.video.encoded_frame_base_path.empty()) { |
1903 std::ostringstream str; | 1907 std::ostringstream str; |
1904 str << receive_logs_++; | 1908 str << receive_logs_++; |
1905 std::string path = | 1909 std::string path = |
1906 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; | 1910 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; |
1907 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), | 1911 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), |
1908 10000000); | 1912 10000000); |
1909 } | 1913 } |
1910 } | 1914 } |
1911 } // namespace webrtc | 1915 } // namespace webrtc |
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