00bb744764
Upstream commit: https://webrtc.googlesource.com/src/+/882b32d00f6330da948d8e578d259dec0078e7ed Reland "Use PayloadTypePicker for video PT assignment" This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95. Reason for revert: Revised codec matching to fix issue. Changes also back out some changes that should not have been included (using PayloadTypePicker for codec list merging). Original change's description: > Revert "Use PayloadTypePicker for video PT assignment" > > This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3. > > Reason for revert: Broke internal tests. > > Original change's description: > > Use PayloadTypePicker for video PT assignment > > > > This includes changes that change the order of codecs. > > It is preparatory to doing late assignment of video PTs. > > > > Bug: webrtc:360058654 > > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400 > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#43489} > > Bug: webrtc:360058654 > Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43490} Bug: webrtc:360058654 Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43554}
139 lines
4.2 KiB
Diff
139 lines
4.2 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 | 11 ++---------
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call/call_basic_stats.cc | 20 ++++++++++++++++++++
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call/call_basic_stats.h | 21 +++++++++++++++++++++
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5 files changed, 49 insertions(+), 22 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 9e4c4d1d96..7bfa30fae0 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 6816212164..eea71e449a 100644
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--- a/call/call.cc
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+++ b/call/call.cc
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@@ -514,19 +514,6 @@ 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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std::unique_ptr<Call> Call::Create(CallConfig config) {
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std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
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if (config.rtp_transport_controller_send_factory != nullptr) {
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diff --git a/call/call.h b/call/call.h
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index ae6eaf82d9..eb4dae0f95 100644
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--- a/call/call.h
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+++ b/call/call.h
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@@ -25,6 +25,7 @@
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#include "api/transport/bitrate_settings.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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@@ -50,15 +51,7 @@ namespace webrtc {
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class Call {
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public:
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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 std::unique_ptr<Call> Create(CallConfig 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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