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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 | 12 ++----------
call/call_basic_stats.cc | 20 ++++++++++++++++++++
call/call_basic_stats.h | 21 +++++++++++++++++++++
video/video_send_stream.h | 1 -
6 files changed, 49 insertions(+), 24 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 fa733a67b9..626ed95066 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 d2ac705274..63dc370f1a 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -460,19 +460,6 @@ 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().
std::unique_ptr<Call> Call::Create(const CallConfig& config) {
Clock* clock =
diff --git a/call/call.h b/call/call.h
index 6f8e4cd6d7..b36872f5b5 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 {
@@ -46,15 +46,7 @@ namespace webrtc {
class Call {
public:
- 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 std::unique_ptr<Call> Create(const CallConfig& config);
static std::unique_ptr<Call> Create(
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 05970d619e..4afafcf8e4 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -36,7 +36,6 @@ namespace test {
class VideoSendStreamPeer;
} // namespace test
-class CallStats;
class IvfFileWriter;
class RateLimiter;
class RtpRtcp;
|
