blob: a1161553cbc35cd2f8c11e9b9ea9a3e18eda445d [file] [log] [blame]
/*
* Copyright (c) 2017 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 "call/video_send_stream.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
VideoSendStream::StreamStats::StreamStats() = default;
VideoSendStream::StreamStats::~StreamStats() = default;
std::string VideoSendStream::StreamStats::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "width: " << width << ", ";
ss << "height: " << height << ", ";
ss << "key: " << frame_counts.key_frames << ", ";
ss << "delta: " << frame_counts.delta_frames << ", ";
ss << "total_bps: " << total_bitrate_bps << ", ";
ss << "retransmit_bps: " << retransmit_bitrate_bps << ", ";
ss << "avg_delay_ms: " << avg_delay_ms << ", ";
ss << "max_delay_ms: " << max_delay_ms << ", ";
ss << "cum_loss: " << rtcp_stats.packets_lost << ", ";
ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", ";
ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", ";
ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", ";
ss << "pli: " << rtcp_packet_type_counts.pli_packets;
return ss.str();
}
VideoSendStream::Stats::Stats() = default;
VideoSendStream::Stats::~Stats() = default;
std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "VideoSendStream stats: " << time_ms << ", {";
ss << "input_fps: " << input_frame_rate << ", ";
ss << "encode_fps: " << encode_frame_rate << ", ";
ss << "encode_ms: " << avg_encode_time_ms << ", ";
ss << "encode_usage_perc: " << encode_usage_percent << ", ";
ss << "target_bps: " << target_media_bitrate_bps << ", ";
ss << "media_bps: " << media_bitrate_bps << ", ";
ss << "suspended: " << (suspended ? "true" : "false") << ", ";
ss << "bw_adapted: " << (bw_limited_resolution ? "true" : "false");
ss << '}';
for (const auto& substream : substreams) {
if (!substream.second.is_rtx && !substream.second.is_flexfec) {
ss << " {ssrc: " << substream.first << ", ";
ss << substream.second.ToString();
ss << '}';
}
}
return ss.str();
}
VideoSendStream::Config::Config(const Config&) = default;
VideoSendStream::Config::Config(Config&&) = default;
VideoSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default;
VideoSendStream::Config::Config::~Config() = default;
std::string VideoSendStream::Config::ToString() const {
char buf[2 * 1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{encoder_settings: { experiment_cpu_load_estimator: "
<< (encoder_settings.experiment_cpu_load_estimator ? "on" : "off") << "}}";
ss << ", rtp: " << rtp.ToString();
ss << ", rtcp: " << rtcp.ToString();
ss << ", pre_encode_callback: "
<< (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr");
ss << ", render_delay_ms: " << render_delay_ms;
ss << ", target_delay_ms: " << target_delay_ms;
ss << ", suspend_below_min_bitrate: "
<< (suspend_below_min_bitrate ? "on" : "off");
ss << '}';
return ss.str();
}
} // namespace webrtc