Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.

webrtc/video/payload_router.h

Index: webrtc/video/payload_router.h
diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h
index 81ec0dd4b1cb1342f44c1e6bc4588efea8dee056..c2f4b0442cb226bb77667330614e1a1251492fd1 100644
--- a/webrtc/video/payload_router.h
+++ b/webrtc/video/payload_router.h
@@ -17,6 +17,7 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
+#include "webrtc/video_encoder.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
@@ -27,10 +28,11 @@
 
 // PayloadRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
-class PayloadRouter {
+class PayloadRouter : public EncodedImageCallback {
  public:
   // Rtp modules are assumed to be sorted in simulcast index order.
-  explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules);
+  explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
+                         int payload_type);
   ~PayloadRouter();
 
   static size_t DefaultMaxPayloadLength();
@@ -41,16 +43,11 @@
   void set_active(bool active);
   bool active();
 
-  // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
-  // Returns true if the packet was routed / sent, false otherwise.
-  bool RoutePayload(FrameType frame_type,
-                    int8_t payload_type,
-                    uint32_t time_stamp,
-                    int64_t capture_time_ms,
-                    const uint8_t* payload_data,
-                    size_t payload_size,
-                    const RTPFragmentationHeader* fragmentation,
-                    const RTPVideoHeader* rtp_video_hdr);
+  // Implements EncodedImageCallback.
+  // Returns 0 if the packet was routed / sent, -1 otherwise.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override;
 
   // Configures current target bitrate per module. 'stream_bitrates' is assumed
   // to be in the same order as 'SetSendingRtpModules'.
@@ -69,6 +66,7 @@
 
   // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
   const std::vector<RtpRtcp*> rtp_modules_;
+  const int payload_type_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };