034f1916b5
This reverts commit d0298f4b161fe2afd4c6b334da31218c115a7eeb. Reason for revert: This change makes the text output for three simulcast layers dangerously close to 1024 characters. Will reland with a larger value. Original change's description: > Add more adaptation text logs for VideoSendStream. > > Tested: Manual tests in app. > Bug: None > Change-Id: I2739a23d37c05cbe1ba9be5c788d1c647265a895 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133186 > Reviewed-by: Niels Moller <nisse@webrtc.org> > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27644} TBR=brandtr@webrtc.org,nisse@webrtc.org Change-Id: Ic6c0c863f6e4aba12feb6c6938db2930396c32f4 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133204 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27647}
97 lines
3.8 KiB
C++
97 lines
3.8 KiB
C++
/*
|
|
* 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 <utility>
|
|
|
|
#include "api/crypto/frame_encryptor_interface.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,
|
|
MediaTransportInterface* media_transport)
|
|
: send_transport(send_transport), media_transport(media_transport) {}
|
|
VideoSendStream::Config::Config(Transport* send_transport)
|
|
: Config(send_transport, nullptr) {}
|
|
|
|
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_report_interval_ms: " << rtcp_report_interval_ms;
|
|
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
|
|
ss << ", media_transport: " << (media_transport ? "(Transport)" : "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
|