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123 lines
4.5 KiB
123 lines
4.5 KiB
/*
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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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namespace {
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const char* StreamTypeToString(VideoSendStream::StreamStats::StreamType type) {
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switch (type) {
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case VideoSendStream::StreamStats::StreamType::kMedia:
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return "media";
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case VideoSendStream::StreamStats::StreamType::kRtx:
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return "rtx";
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case VideoSendStream::StreamStats::StreamType::kFlexfec:
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return "flexfec";
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}
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}
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} // namespace
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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 << "type: " << StreamTypeToString(type);
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if (referenced_media_ssrc.has_value())
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ss << " (for: " << referenced_media_ssrc.value() << ")";
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ss << ", ";
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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.type ==
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VideoSendStream::StreamStats::StreamType::kMedia) {
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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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: rtp(),
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encoder_settings(VideoEncoder::Capabilities(rtp.lntf.enabled)),
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send_transport(send_transport) {}
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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 << ", 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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