rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/config.cc

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2016-06-21 20:13:05 +00:00
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/config.h"
#include <sstream>
#include <string>
namespace webrtc {
std::string NackConfig::ToString() const {
std::stringstream ss;
ss << "{rtp_history_ms: " << rtp_history_ms;
ss << '}';
return ss.str();
}
std::string FecConfig::ToString() const {
std::stringstream ss;
ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
ss << ", red_payload_type: " << red_payload_type;
ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
ss << '}';
return ss.str();
}
std::string RtpExtension::ToString() const {
std::stringstream ss;
ss << "{uri: " << uri;
ss << ", id: " << id;
ss << '}';
return ss.str();
}
const char* RtpExtension::kAudioLevelUri =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
const int RtpExtension::kAudioLevelDefaultId = 1;
const char* RtpExtension::kTimestampOffsetUri =
"urn:ietf:params:rtp-hdrext:toffset";
const int RtpExtension::kTimestampOffsetDefaultId = 2;
const char* RtpExtension::kAbsSendTimeUri =
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
const int RtpExtension::kAbsSendTimeDefaultId = 3;
const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation";
const int RtpExtension::kVideoRotationDefaultId = 4;
const char* RtpExtension::kTransportSequenceNumberUri =
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
// This extension allows applications to adaptively limit the playout delay
// on frames as per the current needs. For example, a gaming application
// has very different needs on end-to-end delay compared to a video-conference
// application.
const char* RtpExtension::kPlayoutDelayUri =
"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
const int RtpExtension::kPlayoutDelayDefaultId = 6;
bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
uri == webrtc::RtpExtension::kAudioLevelUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
}
bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
uri == webrtc::RtpExtension::kAbsSendTimeUri ||
uri == webrtc::RtpExtension::kVideoRotationUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kPlayoutDelayUri;
}
VideoStream::VideoStream()
: width(0),
height(0),
max_framerate(-1),
min_bitrate_bps(-1),
target_bitrate_bps(-1),
max_bitrate_bps(-1),
max_qp(-1) {}
VideoStream::~VideoStream() = default;
std::string VideoStream::ToString() const {
std::stringstream ss;
ss << "{width: " << width;
ss << ", height: " << height;
ss << ", max_framerate: " << max_framerate;
ss << ", min_bitrate_bps:" << min_bitrate_bps;
ss << ", target_bitrate_bps:" << target_bitrate_bps;
ss << ", max_bitrate_bps:" << max_bitrate_bps;
ss << ", max_qp: " << max_qp;
ss << ", temporal_layer_thresholds_bps: [";
for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
ss << temporal_layer_thresholds_bps[i];
if (i != temporal_layer_thresholds_bps.size() - 1)
ss << ", ";
}
ss << ']';
ss << '}';
return ss.str();
}
VideoEncoderConfig::VideoEncoderConfig()
: content_type(ContentType::kRealtimeVideo),
encoder_specific_settings(NULL),
min_transmit_bitrate_bps(0),
expect_encode_from_texture(false) {}
VideoEncoderConfig::~VideoEncoderConfig() = default;
std::string VideoEncoderConfig::ToString() const {
std::stringstream ss;
ss << "{streams: [";
for (size_t i = 0; i < streams.size(); ++i) {
ss << streams[i].ToString();
if (i != streams.size() - 1)
ss << ", ";
}
ss << ']';
ss << ", content_type: ";
switch (content_type) {
case ContentType::kRealtimeVideo:
ss << "kRealtimeVideo";
break;
case ContentType::kScreen:
ss << "kScreenshare";
break;
}
ss << ", encoder_specific_settings: ";
ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
ss << '}';
return ss.str();
}
} // namespace webrtc