зеркало из https://github.com/mozilla/gecko-dev.git
164 строки
4.8 KiB
Diff
164 строки
4.8 KiB
Diff
From: Michael Froman <mjfroman@mac.com>
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Date: Mon, 4 Apr 2022 12:25:26 -0500
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Subject: Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into
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seperate files
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---
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call/BUILD.gn | 6 ++++++
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call/call.cc | 13 -------------
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call/call.h | 13 ++-----------
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call/call_basic_stats.cc | 20 ++++++++++++++++++++
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call/call_basic_stats.h | 21 +++++++++++++++++++++
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video/video_send_stream.h | 1 -
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6 files changed, 49 insertions(+), 25 deletions(-)
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create mode 100644 call/call_basic_stats.cc
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create mode 100644 call/call_basic_stats.h
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diff --git a/call/BUILD.gn b/call/BUILD.gn
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index 0e52e8fb3f..26618aee80 100644
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--- a/call/BUILD.gn
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+++ b/call/BUILD.gn
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@@ -33,6 +33,12 @@ rtc_library("call_interfaces") {
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"syncable.cc",
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"syncable.h",
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]
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+ if (build_with_mozilla) {
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+ sources += [
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+ "call_basic_stats.cc",
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+ "call_basic_stats.h",
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+ ]
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+ }
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deps = [
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":audio_sender_interface",
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diff --git a/call/call.cc b/call/call.cc
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index a63087f5c1..4c3f4b63fc 100644
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--- a/call/call.cc
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+++ b/call/call.cc
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@@ -472,19 +472,6 @@ class Call final : public webrtc::Call,
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};
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} // namespace internal
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-std::string Call::Stats::ToString(int64_t time_ms) const {
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- char buf[1024];
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- rtc::SimpleStringBuilder ss(buf);
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- ss << "Call stats: " << time_ms << ", {";
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- ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
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- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
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- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
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- ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
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- ss << "rtt_ms: " << rtt_ms;
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- ss << '}';
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- return ss.str();
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-}
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-
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/* Mozilla: Avoid this since it could use GetRealTimeClock().
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Call* Call::Create(const Call::Config& config) {
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Clock* clock = Clock::GetRealTimeClock();
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diff --git a/call/call.h b/call/call.h
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index 366978392e..42daa95a6c 100644
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--- a/call/call.h
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+++ b/call/call.h
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@@ -21,6 +21,7 @@
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#include "api/task_queue/task_queue_base.h"
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#include "call/audio_receive_stream.h"
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#include "call/audio_send_stream.h"
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+#include "call/call_basic_stats.h"
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#include "call/call_config.h"
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#include "call/flexfec_receive_stream.h"
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#include "call/packet_receiver.h"
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@@ -30,7 +31,6 @@
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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-#include "rtc_base/ref_count.h"
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namespace webrtc {
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@@ -47,16 +47,7 @@ namespace webrtc {
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class Call {
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public:
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using Config = CallConfig;
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-
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- struct Stats {
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- std::string ToString(int64_t time_ms) const;
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-
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- int send_bandwidth_bps = 0; // Estimated available send bandwidth.
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- int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
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- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
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- int64_t pacer_delay_ms = 0;
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- int64_t rtt_ms = -1;
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- };
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+ using Stats = CallBasicStats;
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static Call* Create(const Call::Config& config);
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static Call* Create(const Call::Config& config,
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diff --git a/call/call_basic_stats.cc b/call/call_basic_stats.cc
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new file mode 100644
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index 0000000000..74333a663b
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--- /dev/null
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+++ b/call/call_basic_stats.cc
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@@ -0,0 +1,20 @@
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+#include "call/call_basic_stats.h"
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+
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+#include "rtc_base/strings/string_builder.h"
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+
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+namespace webrtc {
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+
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+std::string CallBasicStats::ToString(int64_t time_ms) const {
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+ char buf[1024];
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+ rtc::SimpleStringBuilder ss(buf);
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+ ss << "Call stats: " << time_ms << ", {";
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+ ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
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+ ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
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+ ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
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+ ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
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+ ss << "rtt_ms: " << rtt_ms;
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+ ss << '}';
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+ return ss.str();
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+}
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+
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+} // namespace webrtc
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diff --git a/call/call_basic_stats.h b/call/call_basic_stats.h
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new file mode 100644
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index 0000000000..98febe9405
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--- /dev/null
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+++ b/call/call_basic_stats.h
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@@ -0,0 +1,21 @@
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+#ifndef CALL_CALL_BASIC_STATS_H_
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+#define CALL_CALL_BASIC_STATS_H_
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+
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+#include <string>
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+
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+namespace webrtc {
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+
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+// named to avoid conflicts with video/call_stats.h
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+struct CallBasicStats {
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+ std::string ToString(int64_t time_ms) const;
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+
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+ int send_bandwidth_bps = 0; // Estimated available send bandwidth.
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+ int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
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+ int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
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+ int64_t pacer_delay_ms = 0;
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+ int64_t rtt_ms = -1;
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+};
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+
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+} // namespace webrtc
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+
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+#endif // CALL_CALL_BASIC_STATS_H_
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diff --git a/video/video_send_stream.h b/video/video_send_stream.h
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index a7ce112b21..404873fd39 100644
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--- a/video/video_send_stream.h
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+++ b/video/video_send_stream.h
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@@ -37,7 +37,6 @@ namespace test {
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class VideoSendStreamPeer;
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} // namespace test
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-class CallStats;
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class IvfFileWriter;
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class RateLimiter;
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class RtpRtcp;
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--
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2.34.1
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