Delete class SSRCDatabase, and its global ssrc registry.

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 14 matching lines @@
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 
 namespace webrtc {
 
 class OverheadObserver;
 class RateLimiter;
 class RtcEventLog;
 class RtpPacketToSend;
 class RTPSenderAudio;
 class RTPSenderVideo;
 
@@ skipping 45 matching lines @@
 
   void SetSendingMediaStatus(bool enabled);
   bool SendingMedia() const;
 
   void GetDataCounters(StreamDataCounters* rtp_stats,
                        StreamDataCounters* rtx_stats) const;
 
   uint32_t TimestampOffset() const;
   void SetTimestampOffset(uint32_t timestamp);
 
-  uint32_t GenerateNewSSRC();
   void SetSSRC(uint32_t ssrc);
 
   uint16_t SequenceNumber() const;
   void SetSequenceNumber(uint16_t seq);
 
   void SetCsrcs(const std::vector<uint32_t>& csrcs);
 
   void SetMaxRtpPacketSize(size_t max_packet_size);
 
   bool SendOutgoingData(FrameType frame_type,
@@ skipping 186 matching lines @@
   RateStatistics total_bitrate_sent_ GUARDED_BY(statistics_crit_);
   RateStatistics nack_bitrate_sent_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
   RtcEventLog* const event_log_;
   SendPacketObserver* const send_packet_observer_;
   BitrateStatisticsObserver* const bitrate_callback_;
 
   // RTP variables
   uint32_t timestamp_offset_ GUARDED_BY(send_critsect_);
-  SSRCDatabase* const ssrc_db_;
   uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
   bool sequence_number_forced_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
-  bool ssrc_forced_ GUARDED_BY(send_critsect_);
-  uint32_t ssrc_ GUARDED_BY(send_critsect_);
+  // Must be explicitly set by the application, use of rtc:Optional
+  // only to keep track of correct use.
+  rtc::Optional<uint32_t> ssrc_ GUARDED_BY(send_critsect_);
   uint32_t last_rtp_timestamp_ GUARDED_BY(send_critsect_);
   int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
   int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
   bool media_has_been_sent_ GUARDED_BY(send_critsect_);
   bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
   std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
   int rtx_ GUARDED_BY(send_critsect_);
-  uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
+  rtc::Optional<uint32_t> ssrc_rtx_ GUARDED_BY(send_critsect_);
   // Mapping rtx_payload_type_map_[associated] = rtx.
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
   size_t rtp_overhead_bytes_per_packet_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
   OverheadObserver* overhead_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_