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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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1701 // Add extension to enable audio send side BWE, and allow audio bit rate | 1701 // Add extension to enable audio send side BWE, and allow audio bit rate |
1702 // adaptation. | 1702 // adaptation. |
1703 audio_send_config_.rtp.extensions.clear(); | 1703 audio_send_config_.rtp.extensions.clear(); |
1704 if (params_.call.send_side_bwe) { | 1704 if (params_.call.send_side_bwe) { |
1705 audio_send_config_.rtp.extensions.push_back( | 1705 audio_send_config_.rtp.extensions.push_back( |
1706 webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, | 1706 webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, |
1707 test::kTransportSequenceNumberExtensionId)); | 1707 test::kTransportSequenceNumberExtensionId)); |
1708 audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps; | 1708 audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps; |
1709 audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps; | 1709 audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps; |
1710 } | 1710 } |
1711 audio_send_config_.send_codec_spec.codec_inst = | 1711 // TODO(ossu): "Add stereo here?" |
kwiberg-webrtc
2017/03/22 00:00:45
Probably. Looks like it was stereo before.
ossu
2017/04/04 15:36:37
Agreed.
| |
1712 CodecInst{120, "OPUS", 48000, 960, 2, 64000}; | 1712 audio_send_config_.send_codec_spec = |
1713 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( | |
1714 {120, {"OPUS", 48000, 2}}); | |
1713 | 1715 |
1714 audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_); | 1716 audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_); |
1715 | 1717 |
1716 AudioReceiveStream::Config audio_config; | 1718 AudioReceiveStream::Config audio_config; |
1717 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; | 1719 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; |
1718 audio_config.rtcp_send_transport = transport; | 1720 audio_config.rtcp_send_transport = transport; |
1719 audio_config.voe_channel_id = receive_channel_id; | 1721 audio_config.voe_channel_id = receive_channel_id; |
1720 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; | 1722 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; |
1721 audio_config.rtp.transport_cc = params_.call.send_side_bwe; | 1723 audio_config.rtp.transport_cc = params_.call.send_side_bwe; |
1722 audio_config.rtp.extensions = audio_send_config_.rtp.extensions; | 1724 audio_config.rtp.extensions = audio_send_config_.rtp.extensions; |
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1881 if (!params_.video.encoded_frame_base_path.empty()) { | 1883 if (!params_.video.encoded_frame_base_path.empty()) { |
1882 std::ostringstream str; | 1884 std::ostringstream str; |
1883 str << receive_logs_++; | 1885 str << receive_logs_++; |
1884 std::string path = | 1886 std::string path = |
1885 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; | 1887 params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf"; |
1886 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), | 1888 stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path), |
1887 10000000); | 1889 10000000); |
1888 } | 1890 } |
1889 } | 1891 } |
1890 } // namespace webrtc | 1892 } // namespace webrtc |
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