Add overhead to audio bwe min, max.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 96f9ed70d403c6c464cc9b015f83190630a0b331..3ff0139b49b298c74317390da1ee3570e4ed0c77 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -399,6 +399,24 @@ class WebRtcVoiceCodecs final {
     return 0;
   }
 
+  static rtc::ArrayView<const int> GetPacketSizesMs(
+      const webrtc::CodecInst& codec) {
+    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
+      if (IsCodec(codec, kCodecPrefs[i].name)) {
+        return rtc::ArrayView<const int>(
+            kCodecPrefs[i].packet_sizes_ms,
+            std::distance(

[stefan-webrtc, 2017/01/10 12:21:17]

Hm, isn't this easier to write with a for loop? Pretty hard to read...

[michaelt, 2017/01/10 13:09:02]

I use a loop now, but i'm not sure if its really better to read. 
What do you think ?

+                std::begin(kCodecPrefs[i].packet_sizes_ms),
+                std::find_if(std::begin(kCodecPrefs[i].packet_sizes_ms),
+                             std::end(kCodecPrefs[i].packet_sizes_ms),
+                             [](const int packet_size_ms) {
+                               return packet_size_ms == 0;
+                             })));
+      }
+    }
+    return rtc::ArrayView<const int>();
+  }
+
   // If the AudioCodec param kCodecParamPTime is set, then we will set it to
   // codec pacsize if it's valid, or we will pick the next smallest value we
   // support.
@@ -1420,8 +1438,32 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
         "Enabled") {
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
-      config_.min_bitrate_bps = kOpusMinBitrateBps;
-      config_.max_bitrate_bps = kOpusBitrateFbBps;
+      if (webrtc::field_trial::FindFullName(
+              "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
+        auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
+            config_.send_codec_spec.codec_inst);
+        if (packet_sizes_ms.size() > 0) {
+          const int* max_packet_size_ms =
+              std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
+          const int* min_packet_size_ms =
+              std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
+
+          // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+          constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
+
+          int min_overhead_bps =
+              kOverheadPerPacket * 8 * 1000 / *max_packet_size_ms;
+          int max_overhead_bps =
+              kOverheadPerPacket * 8 * 1000 /
+              std::max(*min_packet_size_ms, kOpusDefaultPTime);

[stefan-webrtc, 2017/01/10 12:21:17]

Add a comment about this max(), so that it's clear that this code was added
because the encoder never uses frames shorter than 20 ms.

[michaelt, 2017/01/10 13:09:02]

Done.

+
+          config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
+          config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
+        }
+      } else {
+        config_.min_bitrate_bps = kOpusMinBitrateBps;
+        config_.max_bitrate_bps = kOpusBitrateFbBps;
+      }
     }
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);