Move video send/receive stream headers to webrtc/call.

webrtc/call/video_send_stream.cc

Index: webrtc/call/video_send_stream.cc
diff --git a/webrtc/call/video_send_stream.cc b/webrtc/call/video_send_stream.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f6ea13f710934ad5407e53fba293226ef425d88a
--- /dev/null
+++ b/webrtc/call/video_send_stream.cc
@@ -0,0 +1,162 @@
+/*
+ *  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 "webrtc/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 << ", 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();
+}
+
+}  // namespace webrtc