Files
platform-external-webrtc/webrtc/config.cc
ilnik 04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00

222 lines
7.0 KiB
C++

/*
* 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>
#include "webrtc/base/checks.h"
namespace webrtc {
std::string NackConfig::ToString() const {
std::stringstream ss;
ss << "{rtp_history_ms: " << rtp_history_ms;
ss << '}';
return ss.str();
}
std::string UlpfecConfig::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();
}
bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
return ulpfec_payload_type == other.ulpfec_payload_type &&
red_payload_type == other.red_payload_type &&
red_rtx_payload_type == other.red_rtx_payload_type;
}
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;
const char* RtpExtension::kVideoContentTypeUri =
"http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
const int RtpExtension::kVideoContentTypeDefaultId = 7;
const char* RtpExtension::kVideoTimingUri =
"http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
const int RtpExtension::kVideoTimingDefaultId = 8;
const int RtpExtension::kMinId = 1;
const int RtpExtension::kMaxId = 14;
bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
return 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 ||
uri == webrtc::RtpExtension::kVideoContentTypeUri ||
uri == webrtc::RtpExtension::kVideoTimingUri;
}
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(nullptr),
min_transmit_bitrate_bps(0),
max_bitrate_bps(0),
number_of_streams(0) {}
VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
VideoEncoderConfig::~VideoEncoderConfig() = default;
std::string VideoEncoderConfig::ToString() const {
std::stringstream 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();
}
VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
VideoCodec* codec) const {
if (codec->codecType == kVideoCodecH264) {
FillVideoCodecH264(codec->H264());
} else if (codec->codecType == kVideoCodecVP8) {
FillVideoCodecVp8(codec->VP8());
} else if (codec->codecType == kVideoCodecVP9) {
FillVideoCodecVp9(codec->VP9());
} else {
RTC_NOTREACHED() << "Encoder specifics set/used for unknown codec type.";
}
}
void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecH264(
VideoCodecH264* h264_settings) const {
RTC_NOTREACHED();
}
void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp8(
VideoCodecVP8* vp8_settings) const {
RTC_NOTREACHED();
}
void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp9(
VideoCodecVP9* vp9_settings) const {
RTC_NOTREACHED();
}
VideoEncoderConfig::H264EncoderSpecificSettings::H264EncoderSpecificSettings(
const VideoCodecH264& specifics)
: specifics_(specifics) {}
void VideoEncoderConfig::H264EncoderSpecificSettings::FillVideoCodecH264(
VideoCodecH264* h264_settings) const {
*h264_settings = specifics_;
}
VideoEncoderConfig::Vp8EncoderSpecificSettings::Vp8EncoderSpecificSettings(
const VideoCodecVP8& specifics)
: specifics_(specifics) {}
void VideoEncoderConfig::Vp8EncoderSpecificSettings::FillVideoCodecVp8(
VideoCodecVP8* vp8_settings) const {
*vp8_settings = specifics_;
}
VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
const VideoCodecVP9& specifics)
: specifics_(specifics) {}
void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
VideoCodecVP9* vp9_settings) const {
*vp9_settings = specifics_;
}
} // namespace webrtc