webrtc_m130/webrtc/config.cc
perkj fa10b557d9 Releand of Let ViEEncoder handle resolution changes.
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.

Original cl description:
Let ViEEncoder handle resolution changes.

This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
2016-10-03 06:45:33 +00:00

200 lines
6.2 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 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(nullptr),
min_transmit_bitrate_bps(0),
max_bitrate_bps(0),
number_of_streams(0) {}
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();
}
void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
VideoCodec* codec) const {
if (codec->codecType == kVideoCodecH264) {
FillVideoCodecH264(&codec->codecSpecific.H264);
} else if (codec->codecType == kVideoCodecVP8) {
FillVideoCodecVp8(&codec->codecSpecific.VP8);
} else if (codec->codecType == kVideoCodecVP9) {
FillVideoCodecVp9(&codec->codecSpecific.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