Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.

BUG=

Review URL: https://codereview.webrtc.org/1350163005

Cr-Commit-Position: refs/heads/master@{#10005}

7 files changed, 29 insertions, 28 deletions

diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn
index 69cd6db558..3e478c1e76 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -28,5 +28,6 @@ source_set("pacing") {
   deps = [
     "../../system_wrappers",
     "../bitrate_controller",
+    "../rtp_rtcp",
   ]
 }
diff --git a/webrtc/modules/pacing/bitrate_prober.cc b/webrtc/modules/pacing/bitrate_prober.cc
index bedb892267..d09ad2df16 100644
--- a/webrtc/modules/pacing/bitrate_prober.cc
+++ b/webrtc/modules/pacing/bitrate_prober.cc
@@ -15,6 +15,7 @@
 #include <limits>
 #include <sstream>
 
+#include "webrtc/modules/pacing/include/paced_sender.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
@@ -29,8 +30,6 @@ int ComputeDeltaFromBitrate(size_t packet_size, int bitrate_bps) {
 }
 }  // namespace
 
-const size_t BitrateProber::kMinProbePacketSize = 200;
-
 BitrateProber::BitrateProber()
     : probing_state_(kDisabled),
       packet_size_last_send_(0),
@@ -90,7 +89,7 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
   // We will send the first probe packet immediately if no packet has been
   // sent before.
   int time_until_probe_ms = 0;
-  if (packet_size_last_send_ > kMinProbePacketSize &&
+  if (packet_size_last_send_ > PacedSender::kMinProbePacketSize &&
       probing_state_ == kProbing) {
     int next_delta_ms = ComputeDeltaFromBitrate(packet_size_last_send_,
                                                 probe_bitrates_.front());
diff --git a/webrtc/modules/pacing/bitrate_prober.h b/webrtc/modules/pacing/bitrate_prober.h
index c26b0d6275..b3f52afeb6 100644
--- a/webrtc/modules/pacing/bitrate_prober.h
+++ b/webrtc/modules/pacing/bitrate_prober.h
@@ -22,8 +22,6 @@ namespace webrtc {
 // on being protected by the caller.
 class BitrateProber {
  public:
-  static const size_t kMinProbePacketSize;
-
   BitrateProber();
 
   void SetEnabled(bool enable);
diff --git a/webrtc/modules/pacing/include/paced_sender.h b/webrtc/modules/pacing/include/paced_sender.h
index 645999d507..afb196fe45 100644
--- a/webrtc/modules/pacing/include/paced_sender.h
+++ b/webrtc/modules/pacing/include/paced_sender.h
@@ -17,6 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -30,16 +31,8 @@ struct Packet;
 class PacketQueue;
 }  // namespace paced_sender
 
-class PacedSender : public Module {
+class PacedSender : public Module, public RtpPacketSender {
  public:
-  enum Priority {
-    kHighPriority = 0,  // Pass through; will be sent immediately.
-    kNormalPriority = 2,  // Put in back of the line.
-    kLowPriority = 3,  // Put in back of the low priority line.
-  };
-  // Low priority packets are mixed with the normal priority packets
-  // while we are paused.
-
   class Callback {
    public:
     // Note: packets sent as a result of a callback should not pass by this
@@ -68,6 +61,8 @@ class PacedSender : public Module {
   // overshoots from the encoder.
   static const float kDefaultPaceMultiplier;
 
+  static const size_t kMinProbePacketSize = 200;
+
   PacedSender(Clock* clock,
               Callback* callback,
               int bitrate_kbps,
@@ -103,12 +98,12 @@ class PacedSender : public Module {
 
   // Returns true if we send the packet now, else it will add the packet
   // information to the queue and call TimeToSendPacket when it's time to send.
-  virtual bool SendPacket(Priority priority,
-                          uint32_t ssrc,
-                          uint16_t sequence_number,
-                          int64_t capture_time_ms,
-                          size_t bytes,
-                          bool retransmission);
+  bool SendPacket(RtpPacketSender::Priority priority,
+                  uint32_t ssrc,
+                  uint16_t sequence_number,
+                  int64_t capture_time_ms,
+                  size_t bytes,
+                  bool retransmission) override;
 
   // Returns the time since the oldest queued packet was enqueued.
   virtual int64_t QueueInMs() const;
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h
index d0bcf558f6..9d461d13a9 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/include/packet_router.h
@@ -19,15 +19,19 @@
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class RtpRtcp;
+namespace rtcp {
+class TransportFeedback;
+}  // namespace rtcp
 
 // PacketRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
-class PacketRouter : public PacedSender::Callback {
+class PacketRouter : public PacedSender::Callback,
+                     public TransportSequenceNumberAllocator {
  public:
   PacketRouter();
   virtual ~PacketRouter();
@@ -44,7 +48,7 @@ class PacketRouter : public PacedSender::Callback {
   size_t TimeToSendPadding(size_t bytes) override;
 
   void SetTransportWideSequenceNumber(uint16_t sequence_number);
-  uint16_t AllocateSequenceNumber();
+  uint16_t AllocateSequenceNumber() override;
 
   // Send transport feedback packet to send-side.
   virtual bool SendFeedback(rtcp::TransportFeedback* packet);
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index d5df480974..55c361b085 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -36,7 +36,7 @@ const int64_t kMaxIntervalTimeMs = 30;
 namespace webrtc {
 namespace paced_sender {
 struct Packet {
-  Packet(PacedSender::Priority priority,
+  Packet(RtpPacketSender::Priority priority,
          uint32_t ssrc,
          uint16_t seq_number,
          int64_t capture_time_ms,
@@ -53,7 +53,7 @@ struct Packet {
         retransmission(retransmission),
         enqueue_order(enqueue_order) {}
 
-  PacedSender::Priority priority;
+  RtpPacketSender::Priority priority;
   uint32_t ssrc;
   uint16_t sequence_number;
   int64_t capture_time_ms;
@@ -268,9 +268,12 @@ void PacedSender::UpdateBitrate(int bitrate_kbps,
   bitrate_bps_ = 1000 * bitrate_kbps;
 }
 
-bool PacedSender::SendPacket(Priority priority, uint32_t ssrc,
-    uint16_t sequence_number, int64_t capture_time_ms, size_t bytes,
-    bool retransmission) {
+bool PacedSender::SendPacket(RtpPacketSender::Priority priority,
+                             uint32_t ssrc,
+                             uint16_t sequence_number,
+                             int64_t capture_time_ms,
+                             size_t bytes,
+                             bool retransmission) {
   CriticalSectionScoped cs(critsect_.get());
 
   if (!enabled_) {
diff --git a/webrtc/modules/pacing/pacing.gypi b/webrtc/modules/pacing/pacing.gypi
index 29d9508648..faa97841c1 100644
--- a/webrtc/modules/pacing/pacing.gypi
+++ b/webrtc/modules/pacing/pacing.gypi
@@ -14,6 +14,7 @@
       'dependencies': [
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
+        '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       ],
       'sources': [
         'include/paced_sender.h',