webrtc_m130/webrtc/config.cc
ilnik 27c46e2872 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 13:20:05 +00:00

212 lines
6.6 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 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;
}
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