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.h b/call/rtp_config.h
index 86d32ac..96fe15f 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -12,8 +12,17 @@
#define CALL_RTP_CONFIG_H_
#include <string>
+#include <vector>
+
+#include "api/rtp_headers.h"
+#include "api/rtpparameters.h"
namespace webrtc {
+// Currently only VP8/VP9 specific.
+struct RtpPayloadState {
+ int16_t picture_id = -1;
+ uint8_t tl0_pic_idx = 0;
+};
// Settings for NACK, see RFC 4585 for details.
struct NackConfig {
NackConfig() : rtp_history_ms(0) {}
@@ -44,5 +53,92 @@
// RTX payload type for RED payload.
int red_rtx_payload_type;
};
+
+static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
+struct RtpConfig {
+ RtpConfig();
+ RtpConfig(const RtpConfig&);
+ ~RtpConfig();
+ std::string ToString() const;
+
+ std::vector<uint32_t> ssrcs;
+
+ // The value to send in the MID RTP header extension if the extension is
+ // included in the list of extensions.
+ std::string mid;
+
+ // See RtcpMode for description.
+ RtcpMode rtcp_mode = RtcpMode::kCompound;
+
+ // Max RTP packet size delivered to send transport from VideoEngine.
+ size_t max_packet_size = kDefaultMaxPacketSize;
+
+ // RTP header extensions to use for this send stream.
+ std::vector<RtpExtension> extensions;
+
+ // TODO(nisse): For now, these are fixed, but we'd like to support
+ // changing codec without recreating the VideoSendStream. Then these
+ // fields must be removed, and association between payload type and codec
+ // must move above the per-stream level. Ownership could be with
+ // RtpTransportControllerSend, with a reference from PayloadRouter, where
+ // the latter would be responsible for mapping the codec type of encoded
+ // images to the right payload type.
+ std::string payload_name;
+ int payload_type = -1;
+
+ // See NackConfig for description.
+ NackConfig nack;
+
+ // See UlpfecConfig for description.
+ UlpfecConfig ulpfec;
+
+ struct Flexfec {
+ Flexfec();
+ Flexfec(const Flexfec&);
+ ~Flexfec();
+ // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
+ int payload_type = -1;
+
+ // SSRC of FlexFEC stream.
+ uint32_t ssrc = 0;
+
+ // Vector containing a single element, corresponding to the SSRC of the
+ // media stream being protected by this FlexFEC stream.
+ // The vector MUST have size 1.
+ //
+ // TODO(brandtr): Update comment above when we support
+ // multistream protection.
+ std::vector<uint32_t> protected_media_ssrcs;
+ } flexfec;
+
+ // Settings for RTP retransmission payload format, see RFC 4588 for
+ // details.
+ struct Rtx {
+ Rtx();
+ Rtx(const Rtx&);
+ ~Rtx();
+ std::string ToString() const;
+ // SSRCs to use for the RTX streams.
+ std::vector<uint32_t> ssrcs;
+
+ // Payload type to use for the RTX stream.
+ int payload_type = -1;
+ } rtx;
+
+ // RTCP CNAME, see RFC 3550.
+ std::string c_name;
+};
+
+struct RtcpConfig {
+ RtcpConfig();
+ RtcpConfig(const RtcpConfig&);
+ ~RtcpConfig();
+ std::string ToString() const;
+
+ // Time interval between RTCP report for video
+ int64_t video_report_interval_ms = 1000;
+ // Time interval between RTCP report for audio
+ int64_t audio_report_interval_ms = 5000;
+};
} // namespace webrtc
#endif // CALL_RTP_CONFIG_H_