Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 58 matching lines @@
   virtual size_t RtpHeaderLength() const = 0;
   // Returns the next sequence number to use for a packet and allocates
   // 'packets_to_send' number of sequence numbers. It's important all allocated
   // sequence numbers are used in sequence to avoid perceived packet loss.
   virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
   virtual uint16_t ActualSendBitrateKbit() const = 0;
 
-  virtual bool SendToNetwork(uint8_t* data_buffer,
-                             size_t payload_length,
-                             size_t rtp_header_length,
-                             int64_t capture_time_ms,
-                             StorageType storage,
-                             RtpPacketSender::Priority priority) = 0;
+  virtual int32_t SendToNetwork(uint8_t* data_buffer,
+                                size_t payload_length,
+                                size_t rtp_header_length,
+                                int64_t capture_time_ms,
+                                StorageType storage,
+                                RtpPacketSender::Priority priority) = 0;
 
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    VideoRotation rotation) const = 0;
   virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0;
   virtual bool ActivateCVORtpHeaderExtension() = 0;
 };
 
 class RTPSender : public RTPSenderInterface {
@@ skipping 52 matching lines @@
   uint32_t GenerateNewSSRC();
   void SetSSRC(uint32_t ssrc);
 
   uint16_t SequenceNumber() const override;
   void SetSequenceNumber(uint16_t seq);
 
   void SetCsrcs(const std::vector<uint32_t>& csrcs);
 
   void SetMaxPayloadLength(size_t max_payload_length);
 
-  bool SendOutgoingData(FrameType frame_type,
-                        int8_t payload_type,
-                        uint32_t timestamp,
-                        int64_t capture_time_ms,
-                        const uint8_t* payload_data,
-                        size_t payload_size,
-                        const RTPFragmentationHeader* fragmentation,
-                        const RTPVideoHeader* rtp_header,
-                        uint32_t* transport_frame_id_out);
+  int32_t SendOutgoingData(FrameType frame_type,
+                           int8_t payload_type,
+                           uint32_t timestamp,
+                           int64_t capture_time_ms,
+                           const uint8_t* payload_data,
+                           size_t payload_size,
+                           const RTPFragmentationHeader* fragmentation,
+                           const RTPVideoHeader* rtp_header);
 
   // RTP header extension
   int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
   int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
   void SetVideoRotation(VideoRotation rotation);
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
@@ skipping 94 matching lines @@
                          int64_t capture_time_ms) override;
 
   size_t RtpHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
-  bool SendToNetwork(uint8_t* data_buffer,
-                     size_t payload_length,
-                     size_t rtp_header_length,
-                     int64_t capture_time_ms,
-                     StorageType storage,
-                     RtpPacketSender::Priority priority) override;
+  int32_t SendToNetwork(uint8_t* data_buffer,
+                        size_t payload_length,
+                        size_t rtp_header_length,
+                        int64_t capture_time_ms,
+                        StorageType storage,
+                        RtpPacketSender::Priority priority) override;
 
   // Audio.
 
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   // Set audio packet size, used to determine when it's time to send a DTMF
   // packet in silence (CNG).
   int32_t SetAudioPacketSize(uint16_t packet_size_samples);
 
@@ skipping 201 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_