commit b5d918324ccf8df76c1482070cd5570c8a7578a2
author Johannes Kron <kron@webrtc.org> Tue May 21 13:19:22 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Wed May 22 08:53:08 2019 +0000
tree a377c21fe29ebbcea74a907fcfc41a0443c5e37c
parent afb8d5cdaeadce0cd4f10f0c5f6f748c9d8d3cad

Add RTP timestamp to contributing sources

RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.

Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}

api/rtp_receiver_interface.h

diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index 3934215..f79bf8f 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -24,6 +24,7 @@
 #include "api/proxy.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
+#include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 
 namespace webrtc {
@@ -36,13 +37,23 @@
 class RtpSource {
  public:
   RtpSource() = delete;
-  RtpSource(int64_t timestamp_ms,
-            uint32_t source_id,
-            RtpSourceType source_type);
+
   RtpSource(int64_t timestamp_ms,
             uint32_t source_id,
             RtpSourceType source_type,
-            uint8_t audio_level);
+            absl::optional<uint8_t> audio_level,
+            uint32_t rtp_timestamp);
+
+  // DEPRECATED: Will be removed after 2019-07-31.
+  RTC_DEPRECATED RtpSource(int64_t timestamp_ms,
+                           uint32_t source_id,
+                           RtpSourceType source_type);
+  // DEPRECATED: Will be removed after 2019-07-31.
+  RTC_DEPRECATED RtpSource(int64_t timestamp_ms,
+                           uint32_t source_id,
+                           RtpSourceType source_type,
+                           uint8_t audio_level);
+
   RtpSource(const RtpSource&);
   RtpSource& operator=(const RtpSource&);
   ~RtpSource();
@@ -64,9 +75,12 @@
     audio_level_ = level;
   }
 
+  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
+
   bool operator==(const RtpSource& o) const {
     return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
-           source_type_ == o.source_type() && audio_level_ == o.audio_level_;
+           source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
+           rtp_timestamp_ == o.rtp_timestamp();
   }
 
  private:
@@ -74,6 +88,7 @@
   uint32_t source_id_;
   RtpSourceType source_type_;
   absl::optional<uint8_t> audio_level_;
+  uint32_t rtp_timestamp_;
 };
 
 class RtpReceiverObserverInterface {