6f8d686d35
Currently there are two structs that are identical and track extension details: webrtc::RtpExtension cricket::RtpHeaderExtension The use of the structs is mixed in the code to track the extensions being supported. This results in duplicate definition of the URI constants and there is code to convert between the two structs. Clean up to use a single RtpHeader throughout the codebase. The actual location of RtpHeader may change in future (perhaps to be located in api/). Additionally, this CL renames some of the constants to clarify Uri and Id use. BUG= webrtc:5895 Review-Url: https://codereview.webrtc.org/1984983002 Cr-Commit-Position: refs/heads/master@{#12924}
134 lines
4.0 KiB
C++
134 lines
4.0 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/config.h"
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#include <sstream>
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#include <string>
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namespace webrtc {
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std::string FecConfig::ToString() const {
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std::stringstream ss;
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ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
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ss << ", red_payload_type: " << red_payload_type;
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ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
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ss << '}';
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return ss.str();
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}
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std::string RtpExtension::ToString() const {
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std::stringstream ss;
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ss << "{uri: " << uri;
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ss << ", id: " << id;
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ss << '}';
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return ss.str();
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}
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const char* RtpExtension::kAudioLevelUri =
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"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
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const int RtpExtension::kAudioLevelDefaultId = 1;
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const char* RtpExtension::kTimestampOffsetUri =
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"urn:ietf:params:rtp-hdrext:toffset";
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const int RtpExtension::kTimestampOffsetDefaultId = 2;
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const char* RtpExtension::kAbsSendTimeUri =
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
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const int RtpExtension::kAbsSendTimeDefaultId = 3;
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const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation";
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const int RtpExtension::kVideoRotationDefaultId = 4;
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const char* RtpExtension::kTransportSequenceNumberUri =
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
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const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
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bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
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return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
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uri == webrtc::RtpExtension::kAudioLevelUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
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}
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bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
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return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
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uri == webrtc::RtpExtension::kAbsSendTimeUri ||
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uri == webrtc::RtpExtension::kVideoRotationUri ||
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uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
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}
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VideoStream::VideoStream()
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: width(0),
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height(0),
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max_framerate(-1),
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min_bitrate_bps(-1),
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target_bitrate_bps(-1),
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max_bitrate_bps(-1),
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max_qp(-1) {}
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VideoStream::~VideoStream() = default;
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std::string VideoStream::ToString() const {
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std::stringstream ss;
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ss << "{width: " << width;
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ss << ", height: " << height;
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ss << ", max_framerate: " << max_framerate;
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ss << ", min_bitrate_bps:" << min_bitrate_bps;
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ss << ", target_bitrate_bps:" << target_bitrate_bps;
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ss << ", max_bitrate_bps:" << max_bitrate_bps;
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ss << ", max_qp: " << max_qp;
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ss << ", temporal_layer_thresholds_bps: [";
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for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
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ss << temporal_layer_thresholds_bps[i];
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if (i != temporal_layer_thresholds_bps.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << '}';
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return ss.str();
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}
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VideoEncoderConfig::VideoEncoderConfig()
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: content_type(ContentType::kRealtimeVideo),
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encoder_specific_settings(NULL),
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min_transmit_bitrate_bps(0) {
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}
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VideoEncoderConfig::~VideoEncoderConfig() = default;
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std::string VideoEncoderConfig::ToString() const {
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std::stringstream ss;
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ss << "{streams: [";
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for (size_t i = 0; i < streams.size(); ++i) {
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ss << streams[i].ToString();
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if (i != streams.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << ", content_type: ";
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switch (content_type) {
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case ContentType::kRealtimeVideo:
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ss << "kRealtimeVideo";
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break;
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case ContentType::kScreen:
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ss << "kScreenshare";
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break;
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}
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ss << ", encoder_specific_settings: ";
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ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
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ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
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ss << '}';
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return ss.str();
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}
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} // namespace webrtc
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