Added ProbeBitrate(bitrate_bps, num_probes) to BitrateProber.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 575 matching lines @@
     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
   }
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
-  if (bitrate_config.max_bitrate_bps != -1)
-    RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
+  if (bitrate_config.max_bitrate_bps < -1)
+    return;

[stefan-webrtc, 2016/08/12 13:19:58]

Is there a reason not to DCHECK?

[philipel, 2016/08/12 14:08:06]

Thought this might be user input, but i haven't looked into it completely so
I'll undo this change.

   if (config_.bitrate_config.min_bitrate_bps ==
           bitrate_config.min_bitrate_bps &&
       (bitrate_config.start_bitrate_bps <= 0 ||
        config_.bitrate_config.start_bitrate_bps ==
            bitrate_config.start_bitrate_bps) &&
       config_.bitrate_config.max_bitrate_bps ==
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
@@ skipping 293 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc