370f93a34a
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4 Now that I am sure that WebRTC code is not calling the obsolete versions, I will just remove the NOT_REACHED and call the new version from the old ones, so as not to trip up downstream projects. Original change's description: > Inform VideoEncoder of negotiated capabilities > > After this CL lands, an announcement will be made to > discuss-webrtc about the deprecation of one version > of InitEncode(). > > Bug: webrtc:10720 > Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Elad Alon <eladalon@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28224} TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org Bug: webrtc:10720 Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28236}
108 lines
4.3 KiB
C++
108 lines
4.3 KiB
C++
/*
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* Copyright (c) 2017 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 "call/video_send_stream.h"
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#include <utility>
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#include "api/crypto/frame_encryptor_interface.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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VideoSendStream::StreamStats::StreamStats() = default;
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VideoSendStream::StreamStats::~StreamStats() = default;
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std::string VideoSendStream::StreamStats::ToString() const {
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char buf[1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "width: " << width << ", ";
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ss << "height: " << height << ", ";
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ss << "key: " << frame_counts.key_frames << ", ";
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ss << "delta: " << frame_counts.delta_frames << ", ";
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ss << "total_bps: " << total_bitrate_bps << ", ";
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ss << "retransmit_bps: " << retransmit_bitrate_bps << ", ";
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ss << "avg_delay_ms: " << avg_delay_ms << ", ";
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ss << "max_delay_ms: " << max_delay_ms << ", ";
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ss << "cum_loss: " << rtcp_stats.packets_lost << ", ";
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ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", ";
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ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", ";
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ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", ";
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ss << "pli: " << rtcp_packet_type_counts.pli_packets;
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return ss.str();
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}
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VideoSendStream::Stats::Stats() = default;
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VideoSendStream::Stats::~Stats() = default;
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std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
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char buf[2048];
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rtc::SimpleStringBuilder ss(buf);
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ss << "VideoSendStream stats: " << time_ms << ", {";
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ss << "input_fps: " << input_frame_rate << ", ";
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ss << "encode_fps: " << encode_frame_rate << ", ";
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ss << "encode_ms: " << avg_encode_time_ms << ", ";
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ss << "encode_usage_perc: " << encode_usage_percent << ", ";
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ss << "target_bps: " << target_media_bitrate_bps << ", ";
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ss << "media_bps: " << media_bitrate_bps << ", ";
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ss << "suspended: " << (suspended ? "true" : "false") << ", ";
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ss << "bw_adapted_res: " << (bw_limited_resolution ? "true" : "false")
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<< ", ";
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ss << "cpu_adapted_res: " << (cpu_limited_resolution ? "true" : "false")
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<< ", ";
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ss << "bw_adapted_fps: " << (bw_limited_framerate ? "true" : "false") << ", ";
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ss << "cpu_adapted_fps: " << (cpu_limited_framerate ? "true" : "false")
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<< ", ";
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ss << "#cpu_adaptations: " << number_of_cpu_adapt_changes << ", ";
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ss << "#quality_adaptations: " << number_of_quality_adapt_changes;
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ss << '}';
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for (const auto& substream : substreams) {
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if (!substream.second.is_rtx && !substream.second.is_flexfec) {
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ss << " {ssrc: " << substream.first << ", ";
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ss << substream.second.ToString();
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ss << '}';
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}
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}
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return ss.str();
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}
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VideoSendStream::Config::Config(const Config&) = default;
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VideoSendStream::Config::Config(Config&&) = default;
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VideoSendStream::Config::Config(Transport* send_transport,
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MediaTransportInterface* media_transport)
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: rtp(),
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encoder_settings(VideoEncoder::Capabilities(rtp.lntf.enabled)),
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send_transport(send_transport),
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media_transport(media_transport) {}
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VideoSendStream::Config::Config(Transport* send_transport)
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: Config(send_transport, nullptr) {}
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VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default;
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VideoSendStream::Config::Config::~Config() = default;
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std::string VideoSendStream::Config::ToString() const {
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char buf[2 * 1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "{encoder_settings: { experiment_cpu_load_estimator: "
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<< (encoder_settings.experiment_cpu_load_estimator ? "on" : "off") << "}}";
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ss << ", rtp: " << rtp.ToString();
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ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
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ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
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ss << ", media_transport: " << (media_transport ? "(Transport)" : "nullptr");
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ss << ", render_delay_ms: " << render_delay_ms;
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ss << ", target_delay_ms: " << target_delay_ms;
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ss << ", suspend_below_min_bitrate: "
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<< (suspend_below_min_bitrate ? "on" : "off");
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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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