`RtpPacketInfo`: new ctor + deprecated ctors clean-up

New ctor added without optional and media specific fields.

Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}

api/rtp_packet_info.h

diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h
index bc9839f..f9980a1c 100644
--- a/api/rtp_packet_info.h
+++ b/api/rtp_packet_info.h
@@ -34,6 +34,13 @@
   RtpPacketInfo(uint32_t ssrc,
                 std::vector<uint32_t> csrcs,
                 uint32_t rtp_timestamp,
+                Timestamp receive_time);
+
+  // TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects
+  // are updated.
+  RtpPacketInfo(uint32_t ssrc,
+                std::vector<uint32_t> csrcs,
+                uint32_t rtp_timestamp,
                 absl::optional<uint8_t> audio_level,
                 absl::optional<AbsoluteCaptureTime> absolute_capture_time,
                 Timestamp receive_time);
@@ -64,31 +71,36 @@
   uint32_t rtp_timestamp() const { return rtp_timestamp_; }
   void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
 
-  absl::optional<uint8_t> audio_level() const { return audio_level_; }
-  void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
-
-  const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
-    return absolute_capture_time_;
-  }
-  void set_absolute_capture_time(
-      const absl::optional<AbsoluteCaptureTime>& value) {
-    absolute_capture_time_ = value;
-  }
-
-  const absl::optional<int64_t>& local_capture_clock_offset() const {
-    return local_capture_clock_offset_;
-  }
-
-  void set_local_capture_clock_offset(const absl::optional<int64_t>& value) {
-    local_capture_clock_offset_ = value;
-  }
-
   Timestamp receive_time() const { return receive_time_; }
   void set_receive_time(Timestamp value) { receive_time_ = value; }
   // TODO(bugs.webrtc.org/12722): Deprecated, remove once downstream projects
   // are updated.
   int64_t receive_time_ms() const { return receive_time_.ms(); }
 
+  absl::optional<uint8_t> audio_level() const { return audio_level_; }
+  RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) {
+    audio_level_ = value;
+    return *this;
+  }
+
+  const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
+    return absolute_capture_time_;
+  }
+  RtpPacketInfo& set_absolute_capture_time(
+      const absl::optional<AbsoluteCaptureTime>& value) {
+    absolute_capture_time_ = value;
+    return *this;
+  }
+
+  const absl::optional<int64_t>& local_capture_clock_offset() const {
+    return local_capture_clock_offset_;
+  }
+  RtpPacketInfo& set_local_capture_clock_offset(
+      const absl::optional<int64_t>& value) {
+    local_capture_clock_offset_ = value;
+    return *this;
+  }
+
  private:
   // Fields from the RTP header:
   // https://tools.ietf.org/html/rfc3550#section-5.1
@@ -96,6 +108,9 @@
   std::vector<uint32_t> csrcs_;
   uint32_t rtp_timestamp_;
 
+  // Local `webrtc::Clock`-based timestamp of when the packet was received.
+  Timestamp receive_time_;
+
   // Fields from the Audio Level header extension:
   // https://tools.ietf.org/html/rfc6464#section-3
   absl::optional<uint8_t> audio_level_;
@@ -112,9 +127,6 @@
   // Clock offset against capturer's clock. Should be derived from the estimated
   // capture clock offset defined in the Absolute Capture Time header extension.
   absl::optional<int64_t> local_capture_clock_offset_;
-
-  // Local `webrtc::Clock`-based timestamp of when the packet was received.
-  Timestamp receive_time_;
 };
 
 bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);