Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-si…

webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
 
 #include <sstream>
 #include <stdio.h>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
-const int kMinBitrateBps = 30000;
-
 namespace flags {
 
 DEFINE_string(extension_type,
               "abs",
               "Extension type, either abs for absolute send time or tsoffset "
               "for timestamp offset.");
 std::string ExtensionType() {
   return static_cast<std::string>(FLAGS_extension_type);
 }
 
@@ skipping 73 matching lines @@
     fprintf(stderr, "Unknown extension type\n");
     return false;
   }
 
   // Setup the RTP header parser and the bitrate estimator.
   *parser = webrtc::RtpHeaderParser::Create();
   (*parser)->RegisterRtpHeaderExtension(extension, flags::ExtensionId());
   if (estimator) {
     switch (extension) {
       case webrtc::kRtpExtensionAbsoluteSendTime: {
-        *estimator = new webrtc::RemoteBitrateEstimatorAbsSendTime(
-            observer, clock, kMinBitrateBps);
+        *estimator =
+            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock);
           *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
           break;
         }
       case webrtc::kRtpExtensionTransmissionTimeOffset: {
-        *estimator = new webrtc::RemoteBitrateEstimatorSingleStream(
-            observer, clock, kMinBitrateBps);
+        *estimator =
+            new webrtc::RemoteBitrateEstimatorSingleStream(observer, clock);
           *estimator_used = "RemoteBitrateEstimator";
           break;
         }
       default:
         assert(false);
     }
   }
   return true;
 }