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Side by Side Diff: webrtc/video/video_quality_test.cc

Issue 2705093002: Injectable audio encoders: WebRtcVoiceEngine and company (Closed)
Patch Set: Channel::GetSendCodec asks both its acm and its codec manager. Created 3 years, 8 months ago
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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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