Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 16 matching lines @@
   RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
   virtual ~RTPSenderAudio();
 
   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                int8_t payload_type,
                                uint32_t frequency,
                                size_t channels,
                                uint32_t rate,
                                RtpUtility::Payload** payload);
 
-  bool SendAudio(FrameType frame_type,
-                 int8_t payload_type,
-                 uint32_t capture_timestamp,
-                 const uint8_t* payload_data,
-                 size_t payload_size,
-                 const RTPFragmentationHeader* fragmentation);
+  int32_t SendAudio(FrameType frame_type,
+                    int8_t payload_type,
+                    uint32_t capture_timestamp,
+                    const uint8_t* payload_data,
+                    size_t payload_size,
+                    const RTPFragmentationHeader* fragmentation);
 
   // 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);
 
   // Store the audio level in dBov for
   // header-extension-for-audio-level-indication.
   // Valid range is [0,100]. Actual value is negative.
   int32_t SetAudioLevel(uint8_t level_dbov);
 
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   int AudioFrequency() const;
 
   // Set payload type for Redundant Audio Data RFC 2198
   int32_t SetRED(int8_t payload_type);
 
   // Get payload type for Redundant Audio Data RFC 2198
   int32_t RED(int8_t* payload_type) const;
 
  protected:
-  bool SendTelephoneEventPacket(
+  int32_t SendTelephoneEventPacket(
       bool ended,
       int8_t dtmf_payload_type,
       uint32_t dtmf_timestamp,
       uint16_t duration,
       bool marker_bit);  // set on first packet in talk burst
 
   bool MarkerBit(FrameType frame_type, int8_t payload_type);
 
  private:
   Clock* const clock_;
@@ skipping 26 matching lines @@
 
   // Audio level indication.
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
   OneTimeEvent first_packet_sent_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_