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https://github.com/mollyim/webrtc.git
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55718120e6
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201. Issue 43201 didn't do the job properly. 1. The audio rtcp report interval is not properly hooked up. 2. We don't need to propagate audio rtcp interval into video send stream or vice versa. 3. We don't need to propagate rtcp report interval to any receiving streams. Bug: webrtc:8789 Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f Reviewed-on: https://webrtc-review.googlesource.com/c/110105 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#25610}
115 lines
3.3 KiB
C++
115 lines
3.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/rtp_config.h"
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#include <cstdint>
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#include "api/array_view.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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std::string NackConfig::ToString() const {
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char buf[1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "{rtp_history_ms: " << rtp_history_ms;
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ss << '}';
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return ss.str();
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}
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std::string UlpfecConfig::ToString() const {
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char buf[1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
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ss << ", red_payload_type: " << red_payload_type;
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ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
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ss << '}';
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return ss.str();
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}
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bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
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return ulpfec_payload_type == other.ulpfec_payload_type &&
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red_payload_type == other.red_payload_type &&
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red_rtx_payload_type == other.red_rtx_payload_type;
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}
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RtpConfig::RtpConfig() = default;
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RtpConfig::RtpConfig(const RtpConfig&) = default;
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RtpConfig::~RtpConfig() = default;
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RtpConfig::Flexfec::Flexfec() = default;
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RtpConfig::Flexfec::Flexfec(const Flexfec&) = default;
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RtpConfig::Flexfec::~Flexfec() = default;
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std::string RtpConfig::ToString() const {
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char buf[2 * 1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "{ssrcs: [";
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for (size_t i = 0; i < ssrcs.size(); ++i) {
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ss << ssrcs[i];
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if (i != ssrcs.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << ", rtcp_mode: "
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<< (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
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: "RtcpMode::kReducedSize");
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ss << ", max_packet_size: " << max_packet_size;
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ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false");
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ss << ", extensions: [";
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for (size_t i = 0; i < extensions.size(); ++i) {
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ss << extensions[i].ToString();
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if (i != extensions.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
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ss << ", ulpfec: " << ulpfec.ToString();
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ss << ", payload_name: " << payload_name;
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ss << ", payload_type: " << payload_type;
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ss << ", flexfec: {payload_type: " << flexfec.payload_type;
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ss << ", ssrc: " << flexfec.ssrc;
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ss << ", protected_media_ssrcs: [";
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for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) {
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ss << flexfec.protected_media_ssrcs[i];
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if (i != flexfec.protected_media_ssrcs.size() - 1)
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ss << ", ";
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}
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ss << "]}";
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ss << ", rtx: " << rtx.ToString();
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ss << ", c_name: " << c_name;
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ss << '}';
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return ss.str();
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}
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RtpConfig::Rtx::Rtx() = default;
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RtpConfig::Rtx::Rtx(const Rtx&) = default;
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RtpConfig::Rtx::~Rtx() = default;
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std::string RtpConfig::Rtx::ToString() const {
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char buf[1024];
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rtc::SimpleStringBuilder ss(buf);
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ss << "{ssrcs: [";
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for (size_t i = 0; i < ssrcs.size(); ++i) {
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ss << ssrcs[i];
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if (i != ssrcs.size() - 1)
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ss << ", ";
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}
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ss << ']';
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ss << ", payload_type: " << payload_type;
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ss << '}';
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return ss.str();
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}
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} // namespace webrtc
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