| /* |
| * 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" |
| |
| namespace webrtc { |
| |
| VideoSendStream::StreamStats::StreamStats() = default; |
| VideoSendStream::StreamStats::~StreamStats() = default; |
| |
| std::string VideoSendStream::StreamStats::ToString() const { |
| std::stringstream ss; |
| 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 { |
| std::stringstream ss; |
| 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 << "preferred_media_bitrate_bps: " << preferred_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 { |
| std::stringstream ss; |
| ss << "{encoder_settings: " << encoder_settings.ToString(); |
| ss << ", rtp: " << rtp.ToString(); |
| ss << ", rtcp: " << rtcp.ToString(); |
| ss << ", pre_encode_callback: " |
| << (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr"); |
| ss << ", post_encode_callback: " |
| << (post_encode_callback ? "(EncodedFrameObserver)" : "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(); |
| } |
| |
| std::string VideoSendStream::Config::EncoderSettings::ToString() const { |
| std::stringstream ss; |
| ss << "{payload_name: " << payload_name; |
| ss << ", payload_type: " << payload_type; |
| ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr"); |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| VideoSendStream::Config::Rtp::Rtp() = default; |
| VideoSendStream::Config::Rtp::Rtp(const Rtp&) = default; |
| VideoSendStream::Config::Rtp::~Rtp() = default; |
| |
| VideoSendStream::Config::Rtp::Flexfec::Flexfec() = default; |
| VideoSendStream::Config::Rtp::Flexfec::Flexfec(const Flexfec&) = default; |
| VideoSendStream::Config::Rtp::Flexfec::~Flexfec() = default; |
| |
| std::string VideoSendStream::Config::Rtp::ToString() const { |
| std::stringstream ss; |
| ss << "{ssrcs: ["; |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| ss << ssrcs[i]; |
| if (i != ssrcs.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| ss << ", rtcp_mode: " |
| << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound" |
| : "RtcpMode::kReducedSize"); |
| ss << ", max_packet_size: " << max_packet_size; |
| ss << ", extensions: ["; |
| for (size_t i = 0; i < extensions.size(); ++i) { |
| ss << extensions[i].ToString(); |
| if (i != extensions.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| |
| ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}'; |
| ss << ", ulpfec: " << ulpfec.ToString(); |
| |
| ss << ", flexfec: {payload_type: " << flexfec.payload_type; |
| ss << ", ssrc: " << flexfec.ssrc; |
| ss << ", protected_media_ssrcs: ["; |
| for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) { |
| ss << flexfec.protected_media_ssrcs[i]; |
| if (i != flexfec.protected_media_ssrcs.size() - 1) |
| ss << ", "; |
| } |
| ss << "]}"; |
| |
| ss << ", rtx: " << rtx.ToString(); |
| ss << ", c_name: " << c_name; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| VideoSendStream::Config::Rtp::Rtx::Rtx() = default; |
| VideoSendStream::Config::Rtp::Rtx::Rtx(const Rtx&) = default; |
| VideoSendStream::Config::Rtp::Rtx::~Rtx() = default; |
| |
| std::string VideoSendStream::Config::Rtp::Rtx::ToString() const { |
| std::stringstream ss; |
| ss << "{ssrcs: ["; |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| ss << ssrcs[i]; |
| if (i != ssrcs.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| |
| ss << ", payload_type: " << payload_type; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| VideoSendStream::Config::Rtcp::Rtcp() = default; |
| VideoSendStream::Config::Rtcp::Rtcp(const Rtcp&) = default; |
| VideoSendStream::Config::Rtcp::~Rtcp() = default; |
| |
| std::string VideoSendStream::Config::Rtcp::ToString() const { |
| std::stringstream ss; |
| ss << "{video_report_interval_ms: " << video_report_interval_ms; |
| ss << ", audio_report_interval_ms: " << audio_report_interval_ms; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| } // namespace webrtc |