Refactoring PayloadRouter.
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}diff --git a/call/rtp_config.cc b/call/rtp_config.cc
index 71322f9..1445c25 100644
--- a/call/rtp_config.cc
+++ b/call/rtp_config.cc
@@ -9,6 +9,7 @@
*/
#include "call/rtp_config.h"
+
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
@@ -36,4 +37,89 @@
red_payload_type == other.red_payload_type &&
red_rtx_payload_type == other.red_rtx_payload_type;
}
+
+RtpConfig::RtpConfig() = default;
+RtpConfig::RtpConfig(const RtpConfig&) = default;
+RtpConfig::~RtpConfig() = default;
+
+RtpConfig::Flexfec::Flexfec() = default;
+RtpConfig::Flexfec::Flexfec(const Flexfec&) = default;
+RtpConfig::Flexfec::~Flexfec() = default;
+
+std::string RtpConfig::ToString() const {
+ char buf[2 * 1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "{ssrcs: [";
+ for (size_t i = 0; i < ssrcs.size(); ++i) {
+ ss << ssrcs[i];
+ if (i != ssrcs.size() - 1)
+ ss << ", ";
+ }
+ ss << ']';
+ ss << ", rtcp_mode: "
+ << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
+ : "RtcpMode::kReducedSize");
+ ss << ", max_packet_size: " << max_packet_size;
+ ss << ", extensions: [";
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ ss << extensions[i].ToString();
+ if (i != extensions.size() - 1)
+ ss << ", ";
+ }
+ ss << ']';
+
+ ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
+ ss << ", ulpfec: " << ulpfec.ToString();
+ ss << ", payload_name: " << payload_name;
+ ss << ", payload_type: " << payload_type;
+
+ ss << ", flexfec: {payload_type: " << flexfec.payload_type;
+ ss << ", ssrc: " << flexfec.ssrc;
+ ss << ", protected_media_ssrcs: [";
+ for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) {
+ ss << flexfec.protected_media_ssrcs[i];
+ if (i != flexfec.protected_media_ssrcs.size() - 1)
+ ss << ", ";
+ }
+ ss << "]}";
+
+ ss << ", rtx: " << rtx.ToString();
+ ss << ", c_name: " << c_name;
+ ss << '}';
+ return ss.str();
+}
+
+RtpConfig::Rtx::Rtx() = default;
+RtpConfig::Rtx::Rtx(const Rtx&) = default;
+RtpConfig::Rtx::~Rtx() = default;
+
+std::string RtpConfig::Rtx::ToString() const {
+ char buf[1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "{ssrcs: [";
+ for (size_t i = 0; i < ssrcs.size(); ++i) {
+ ss << ssrcs[i];
+ if (i != ssrcs.size() - 1)
+ ss << ", ";
+ }
+ ss << ']';
+
+ ss << ", payload_type: " << payload_type;
+ ss << '}';
+ return ss.str();
+}
+
+RtcpConfig::RtcpConfig() = default;
+RtcpConfig::RtcpConfig(const RtcpConfig&) = default;
+RtcpConfig::~RtcpConfig() = default;
+
+std::string RtcpConfig::ToString() const {
+ char buf[1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "{video_report_interval_ms: " << video_report_interval_ms;
+ ss << ", audio_report_interval_ms: " << audio_report_interval_ms;
+ ss << '}';
+ return ss.str();
+}
+
} // namespace webrtc
