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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 01:47:29 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 01:47:29 +0000
commit0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d (patch)
treea31f07c9bcca9d56ce61e9a1ffd30ef350d513aa /third_party/libwebrtc/call/video_send_stream.cc
parentInitial commit. (diff)
downloadfirefox-esr-0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d.tar.xz
firefox-esr-0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d.zip
Adding upstream version 115.8.0esr.upstream/115.8.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/video_send_stream.cc')
-rw-r--r--third_party/libwebrtc/call/video_send_stream.cc129
1 files changed, 129 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/video_send_stream.cc b/third_party/libwebrtc/call/video_send_stream.cc
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+++ b/third_party/libwebrtc/call/video_send_stream.cc
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+/*
+ * 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"
+#include "rtc_base/strings/string_format.h"
+
+namespace webrtc {
+
+namespace {
+
+const char* StreamTypeToString(VideoSendStream::StreamStats::StreamType type) {
+ switch (type) {
+ case VideoSendStream::StreamStats::StreamType::kMedia:
+ return "media";
+ case VideoSendStream::StreamStats::StreamType::kRtx:
+ return "rtx";
+ case VideoSendStream::StreamStats::StreamType::kFlexfec:
+ return "flexfec";
+ }
+ RTC_CHECK_NOTREACHED();
+}
+
+} // namespace
+
+VideoSendStream::StreamStats::StreamStats() = default;
+VideoSendStream::StreamStats::~StreamStats() = default;
+
+std::string VideoSendStream::StreamStats::ToString() const {
+ char buf[1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "type: " << StreamTypeToString(type);
+ if (referenced_media_ssrc.has_value())
+ ss << " (for: " << referenced_media_ssrc.value() << ")";
+ 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 << ", ";
+ if (report_block_data) {
+ ss << "cum_loss: " << report_block_data->report_block().packets_lost
+ << ", ";
+ ss << "max_ext_seq: "
+ << report_block_data->report_block().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[2048];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "VideoSendStream stats: " << time_ms << ", {";
+ ss << "input_fps: " << rtc::StringFormat("%.1f", 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_res: " << (bw_limited_resolution ? "true" : "false")
+ << ", ";
+ ss << "cpu_adapted_res: " << (cpu_limited_resolution ? "true" : "false")
+ << ", ";
+ ss << "bw_adapted_fps: " << (bw_limited_framerate ? "true" : "false") << ", ";
+ ss << "cpu_adapted_fps: " << (cpu_limited_framerate ? "true" : "false")
+ << ", ";
+ ss << "#cpu_adaptations: " << number_of_cpu_adapt_changes << ", ";
+ ss << "#quality_adaptations: " << number_of_quality_adapt_changes;
+ ss << '}';
+ for (const auto& substream : substreams) {
+ if (substream.second.type ==
+ VideoSendStream::StreamStats::StreamType::kMedia) {
+ 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)
+ : rtp(),
+ encoder_settings(VideoEncoder::Capabilities(rtp.lntf.enabled)),
+ 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_report_interval_ms: " << rtcp_report_interval_ms;
+ ss << ", send_transport: " << (send_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