Added API for getting the send-side estimated bandwidth.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/372006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1591 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/interface/rtp_rtcp.h

@@ -566,6 +566,12 @@ public:
                              WebRtc_UWord32* fecRate,
                              WebRtc_UWord32* nackRate) const = 0;
 
+    /*
+     *  Get the send-side estimate of the available bandwidth.
+     */
+    virtual int EstimatedBandwidth(
+        WebRtc_UWord32* available_bandwidth) const = 0;
+
     /*
     *   Used by the codec module to deliver a video or audio frame for packetization
     *

src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -1,3 +1,13 @@
+/*
+ *  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.
+ */
+
 #include "../testing/gmock/include/gmock/gmock.h"
 
 #include "modules/interface/module.h"
@@ -132,6 +142,8 @@ class MockRtpRtcp : public RtpRtcp {
       bool());
   MOCK_CONST_METHOD4(BitrateSent,
       void(WebRtc_UWord32* totalRate, WebRtc_UWord32* videoRate, WebRtc_UWord32* fecRate, WebRtc_UWord32* nackRate));
+  MOCK_CONST_METHOD1(EstimatedBandwidth,
+      int(WebRtc_UWord32* available_bandwidth));
   MOCK_METHOD7(SendOutgoingData,
       WebRtc_Word32(const FrameType frameType, const WebRtc_Word8 payloadType, const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpVideoHdr));
   MOCK_METHOD1(RegisterIncomingRTCPCallback,

src/modules/rtp_rtcp/source/bandwidth_management.cc

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -182,6 +182,19 @@ WebRtc_Word32 BandwidthManagement::UpdatePacketLoss(
     return 0;
 }
 
+WebRtc_Word32 BandwidthManagement::AvailableBandwidth(
+    WebRtc_UWord32* bandwidthKbit) const {
+  CriticalSectionScoped cs(_critsect);
+  if (_bitRate == 0) {
+    return -1;
+  }
+  if (!bandwidthKbit) {
+    return -1;
+  }
+  *bandwidthKbit = _bitRate;
+  return 0;
+}
+
 /* Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
  * The formula in RFC 3448, Section 3.1, is used.
  */

src/modules/rtp_rtcp/source/bandwidth_management.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -41,7 +41,9 @@ public:
         WebRtc_UWord32* newBitrate,
         WebRtc_Word64 nowMS);
 
-    WebRtc_Word32 AvailableBandwidth(WebRtc_UWord16* bandwidthKbit) const;
+    // If no bandwidth estimate is available or if |bandwidthKbit| is NULL,
+    // -1 is returned.
+    WebRtc_Word32 AvailableBandwidth(WebRtc_UWord32* bandwidthKbit) const;
 
     void SetSendBitrate(const WebRtc_UWord32 startBitrate,
                         const WebRtc_UWord16 minBitrateKbit,

src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -2501,6 +2501,11 @@ void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* totalRate,
         *nackRate = _rtpSender.NackOverheadRate();
 }
 
+int ModuleRtpRtcpImpl::EstimatedBandwidth(
+    WebRtc_UWord32* available_bandwidth) const {
+  return _bandwidthManagement.AvailableBandwidth(available_bandwidth);
+}
+
 // for lip sync
 void ModuleRtpRtcpImpl::OnReceivedNTP() {
     // don't do anything if we are the audio module

src/modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -491,6 +491,8 @@ public:
                              WebRtc_UWord32* fecRate,
                              WebRtc_UWord32* nackRate) const;
 
+    virtual int EstimatedBandwidth(WebRtc_UWord32* available_bandwidth) const;
+
     virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC);
     
     virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);

src/video_engine/include/vie_rtp_rtcp.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -242,6 +242,12 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
                                 unsigned int& fec_bitrate_sent,
                                 unsigned int& nackBitrateSent) const = 0;
 
+  // This function gets the send-side estimated bandwidth available for video,
+  // including overhead, in bits/s.
+  virtual int GetEstimatedBandwidth(
+      const int video_channel,
+      unsigned int* estimated_bandwidth) const = 0;
+
   // This function enables or disables an RTP keep-alive mechanism which can
   // be used to maintain an existing Network Address Translator (NAT) mapping
   // while regular RTP is no longer transmitted.

src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -221,6 +221,11 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     EXPECT_GT(recJitter, 0u);
     EXPECT_GT(recRttMs, 0);
 
+    unsigned int estimated_bandwidth = 0;
+    EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedBandwidth(tbChannel.videoChannel,
+                                                     &estimated_bandwidth));
+    EXPECT_GT(estimated_bandwidth, 0u);
+
     // Check that rec stats extended max is greater than what we've sent.
     EXPECT_GE(recExtendedMax, sentExtendedMax);
     EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));

src/video_engine/vie_encoder.cc

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -560,6 +560,13 @@ WebRtc_Word32 ViEEncoder::SendCodecStatistics(
   return 0;
 }
 
+WebRtc_Word32 ViEEncoder::EstimatedBandwidth(
+    WebRtc_UWord32* available_bandwidth) const {
+  WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
+               __FUNCTION__);
+  return default_rtp_rtcp_.EstimatedBandwidth(available_bandwidth);
+}
+
 WebRtc_Word32 ViEEncoder::UpdateProtectionMethod() {
   bool fec_enabled = false;
   WebRtc_UWord8 dummy_ptype_red = 0;

src/video_engine/vie_encoder.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -88,6 +88,7 @@ class ViEEncoder
   WebRtc_Word32 SendKeyFrame();
   WebRtc_Word32 SendCodecStatistics(WebRtc_UWord32& num_key_frames,
                                     WebRtc_UWord32& num_delta_frames);
+  WebRtc_Word32 EstimatedBandwidth(WebRtc_UWord32* available_bandwidth) const;
   // Loss protection.
   WebRtc_Word32 UpdateProtectionMethod();
 

src/video_engine/vie_rtp_rtcp_impl.cc

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -733,6 +733,26 @@ int ViERTP_RTCPImpl::GetBandwidthUsage(const int video_channel,
   return 0;
 }
 
+int ViERTP_RTCPImpl::GetEstimatedBandwidth(
+    const int video_channel,
+    unsigned int* estimated_bandwidth) const {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+               ViEId(shared_data_->instance_id(), video_channel),
+               "%s(channel: %d)", __FUNCTION__, video_channel);
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+  if (!vie_encoder) {
+    WEBRTC_TRACE(kTraceError, kTraceVideo,
+                 ViEId(shared_data_->instance_id(), video_channel),
+                 "%s: Could not get encoder for channel %d", __FUNCTION__,
+                 video_channel);
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  return vie_encoder->EstimatedBandwidth(
+      static_cast<WebRtc_UWord32*>(estimated_bandwidth));
+}
+
 int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
     const int video_channel,
     bool enable,

src/video_engine/vie_rtp_rtcp_impl.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -87,9 +87,13 @@ class ViERTP_RTCPImpl
                                 unsigned int& video_bitrate_sent,
                                 unsigned int& fec_bitrate_sent,
                                 unsigned int& nackBitrateSent) const;
-  virtual int SetRTPKeepAliveStatus(const int video_channel,
+  virtual int GetEstimatedBandwidth(
-                                    bool enable,
+      const int video_channel,
-                                    const char unknown_payload_type,
+      unsigned int* estimated_bandwidth) const;
+  virtual int SetRTPKeepAliveStatus(
+      const int video_channel,
+      bool enable,
+      const char unknown_payload_type,
       const unsigned int delta_transmit_time_seconds);
   virtual int GetRTPKeepAliveStatus(
       const int video_channel,