mirror of
https://github.com/mollyim/webrtc.git
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3b3622fafc
Replaced with scalars ulpfec_payload_type and red_payload_type. In particular, ulpfec.red_rtx_payload_type, which duplicated info in rtx_associated_payload_types, is deleted. This is a followup to cl https://codereview.webrtc.org/3012963002. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3019453002 Cr-Commit-Position: refs/heads/master@{#19965}
132 lines
4.8 KiB
C++
132 lines
4.8 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_receive_stream.h"
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namespace webrtc {
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VideoReceiveStream::Decoder::Decoder() = default;
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VideoReceiveStream::Decoder::Decoder(const Decoder&) = default;
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VideoReceiveStream::Decoder::~Decoder() = default;
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std::string VideoReceiveStream::Decoder::ToString() const {
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std::stringstream ss;
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ss << "{decoder: " << (decoder ? "(VideoDecoder)" : "nullptr");
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ss << ", payload_type: " << payload_type;
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ss << ", payload_name: " << payload_name;
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ss << ", codec_params: {";
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for (const auto& it : codec_params)
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ss << it.first << ": " << it.second;
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ss << '}';
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ss << '}';
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return ss.str();
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}
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VideoReceiveStream::Stats::Stats() = default;
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VideoReceiveStream::Stats::~Stats() = default;
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std::string VideoReceiveStream::Stats::ToString(int64_t time_ms) const {
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std::stringstream ss;
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ss << "VideoReceiveStream stats: " << time_ms << ", {ssrc: " << ssrc << ", ";
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ss << "total_bps: " << total_bitrate_bps << ", ";
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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 << "network_fps: " << network_frame_rate << ", ";
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ss << "decode_fps: " << decode_frame_rate << ", ";
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ss << "render_fps: " << render_frame_rate << ", ";
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ss << "decode_ms: " << decode_ms << ", ";
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ss << "max_decode_ms: " << max_decode_ms << ", ";
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ss << "cur_delay_ms: " << current_delay_ms << ", ";
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ss << "targ_delay_ms: " << target_delay_ms << ", ";
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ss << "jb_delay_ms: " << jitter_buffer_ms << ", ";
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ss << "min_playout_delay_ms: " << min_playout_delay_ms << ", ";
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ss << "discarded: " << discarded_packets << ", ";
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ss << "sync_offset_ms: " << sync_offset_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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ss << '}';
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return ss.str();
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}
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VideoReceiveStream::Config::Config(const Config&) = default;
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VideoReceiveStream::Config::Config(Config&&) = default;
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VideoReceiveStream::Config::Config(Transport* rtcp_send_transport)
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: rtcp_send_transport(rtcp_send_transport) {}
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VideoReceiveStream::Config& VideoReceiveStream::Config::operator=(Config&&) =
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default;
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VideoReceiveStream::Config::Config::~Config() = default;
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std::string VideoReceiveStream::Config::ToString() const {
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std::stringstream ss;
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ss << "{decoders: [";
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for (size_t i = 0; i < decoders.size(); ++i) {
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ss << decoders[i].ToString();
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if (i != decoders.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << ", rtp: " << rtp.ToString();
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ss << ", renderer: " << (renderer ? "(renderer)" : "nullptr");
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ss << ", render_delay_ms: " << render_delay_ms;
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if (!sync_group.empty())
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ss << ", sync_group: " << sync_group;
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ss << ", pre_decode_callback: "
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<< (pre_decode_callback ? "(EncodedFrameObserver)" : "nullptr");
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ss << ", target_delay_ms: " << target_delay_ms;
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ss << '}';
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return ss.str();
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}
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VideoReceiveStream::Config::Rtp::Rtp() = default;
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VideoReceiveStream::Config::Rtp::Rtp(const Rtp&) = default;
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VideoReceiveStream::Config::Rtp::~Rtp() = default;
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std::string VideoReceiveStream::Config::Rtp::ToString() const {
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std::stringstream ss;
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ss << "{remote_ssrc: " << remote_ssrc;
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ss << ", local_ssrc: " << local_ssrc;
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ss << ", rtcp_mode: "
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<< (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
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: "RtcpMode::kReducedSize");
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ss << ", rtcp_xr: ";
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ss << "{receiver_reference_time_report: "
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<< (rtcp_xr.receiver_reference_time_report ? "on" : "off");
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ss << '}';
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ss << ", remb: " << (remb ? "on" : "off");
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ss << ", transport_cc: " << (transport_cc ? "on" : "off");
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ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
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ss << ", ulpfec_payload_type: " << ulpfec_payload_type;
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ss << ", red_type: " << red_payload_type;
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ss << ", rtx_ssrc: " << rtx_ssrc;
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ss << ", rtx_payload_types: {";
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for (auto& kv : rtx_associated_payload_types) {
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ss << kv.first << " (pt) -> " << kv.second << " (apt), ";
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}
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ss << '}';
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ss << ", extensions: [";
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for (size_t i = 0; i < extensions.size(); ++i) {
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ss << extensions[i].ToString();
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if (i != extensions.size() - 1)
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ss << ", ";
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}
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ss << ']';
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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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