From: Michael Froman <mjfroman@mac.com>
Date: Mon, 4 Apr 2022 12:25:26 -0500
Subject: Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into
 seperate files

---
 call/BUILD.gn             |  6 ++++++
 call/call.cc              | 13 -------------
 call/call.h               | 13 ++-----------
 call/call_basic_stats.cc  | 20 ++++++++++++++++++++
 call/call_basic_stats.h   | 21 +++++++++++++++++++++
 video/video_send_stream.h |  1 -
 6 files changed, 49 insertions(+), 25 deletions(-)
 create mode 100644 call/call_basic_stats.cc
 create mode 100644 call/call_basic_stats.h

diff --git a/call/BUILD.gn b/call/BUILD.gn
index 0e52e8fb3f..26618aee80 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -33,6 +33,12 @@ rtc_library("call_interfaces") {
     "syncable.cc",
     "syncable.h",
   ]
+  if (build_with_mozilla) {
+    sources += [
+      "call_basic_stats.cc",
+      "call_basic_stats.h",
+    ]
+  }
 
   deps = [
     ":audio_sender_interface",
diff --git a/call/call.cc b/call/call.cc
index a63087f5c1..4c3f4b63fc 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -472,19 +472,6 @@ class Call final : public webrtc::Call,
 };
 }  // namespace internal
 
-std::string Call::Stats::ToString(int64_t time_ms) const {
-  char buf[1024];
-  rtc::SimpleStringBuilder ss(buf);
-  ss << "Call stats: " << time_ms << ", {";
-  ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
-  ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
-  ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
-  ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
-  ss << "rtt_ms: " << rtt_ms;
-  ss << '}';
-  return ss.str();
-}
-
 /* Mozilla: Avoid this since it could use GetRealTimeClock().
 Call* Call::Create(const Call::Config& config) {
   Clock* clock = Clock::GetRealTimeClock();
diff --git a/call/call.h b/call/call.h
index 366978392e..42daa95a6c 100644
--- a/call/call.h
+++ b/call/call.h
@@ -21,6 +21,7 @@
 #include "api/task_queue/task_queue_base.h"
 #include "call/audio_receive_stream.h"
 #include "call/audio_send_stream.h"
+#include "call/call_basic_stats.h"
 #include "call/call_config.h"
 #include "call/flexfec_receive_stream.h"
 #include "call/packet_receiver.h"
@@ -30,7 +31,6 @@
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
-#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -47,16 +47,7 @@ namespace webrtc {
 class Call {
  public:
   using Config = CallConfig;
-
-  struct Stats {
-    std::string ToString(int64_t time_ms) const;
-
-    int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
-    int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
-    int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
-    int64_t pacer_delay_ms = 0;
-    int64_t rtt_ms = -1;
-  };
+  using Stats = CallBasicStats;
 
   static Call* Create(const Call::Config& config);
   static Call* Create(const Call::Config& config,
diff --git a/call/call_basic_stats.cc b/call/call_basic_stats.cc
new file mode 100644
index 0000000000..74333a663b
--- /dev/null
+++ b/call/call_basic_stats.cc
@@ -0,0 +1,20 @@
+#include "call/call_basic_stats.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+std::string CallBasicStats::ToString(int64_t time_ms) const {
+  char buf[1024];
+  rtc::SimpleStringBuilder ss(buf);
+  ss << "Call stats: " << time_ms << ", {";
+  ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
+  ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
+  ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
+  ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
+  ss << "rtt_ms: " << rtt_ms;
+  ss << '}';
+  return ss.str();
+}
+
+}  // namespace webrtc
diff --git a/call/call_basic_stats.h b/call/call_basic_stats.h
new file mode 100644
index 0000000000..98febe9405
--- /dev/null
+++ b/call/call_basic_stats.h
@@ -0,0 +1,21 @@
+#ifndef CALL_CALL_BASIC_STATS_H_
+#define CALL_CALL_BASIC_STATS_H_
+
+#include <string>
+
+namespace webrtc {
+
+// named to avoid conflicts with video/call_stats.h
+struct CallBasicStats {
+  std::string ToString(int64_t time_ms) const;
+
+  int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
+  int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
+  int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
+  int64_t pacer_delay_ms = 0;
+  int64_t rtt_ms = -1;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_CALL_BASIC_STATS_H_
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index a7ce112b21..404873fd39 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -37,7 +37,6 @@ namespace test {
 class VideoSendStreamPeer;
 }  // namespace test
 
-class CallStats;
 class IvfFileWriter;
 class RateLimiter;
 class RtpRtcp;
-- 
2.34.1