Adding packet overhead to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index ac0045ec4dee39dc0127c82df9de25258759673f..2f64df066b36bba2ec8bd963627bf594e3c66dd0 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -13,17 +13,11 @@
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {
 
-namespace {
-// TODO(minyue): consider passing this from a higher layer through
-// SetConstraints().
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-}
-
 BitrateController::Config::Config(int initial_bitrate_bps,
                                   int initial_frame_length_ms)
     : initial_bitrate_bps(initial_bitrate_bps),
@@ -34,10 +28,9 @@ BitrateController::Config::~Config() = default;
 BitrateController::BitrateController(const Config& config)
     : config_(config),
       bitrate_bps_(config_.initial_bitrate_bps),
-      overhead_rate_bps_(kPacketOverheadBits * 1000 /
-                         config_.initial_frame_length_ms) {
+      frame_length_ms_(config_.initial_frame_length_ms) {
   RTC_DCHECK_GT(bitrate_bps_, 0);
-  RTC_DCHECK_GT(overhead_rate_bps_, 0);
+  RTC_DCHECK_GT(frame_length_ms_, 0);
 }
 
 void BitrateController::MakeDecision(
@@ -45,23 +38,18 @@ void BitrateController::MakeDecision(
     AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
   // Decision on |bitrate_bps| should not have been made.
   RTC_DCHECK(!config->bitrate_bps);
-
-  if (metrics.target_audio_bitrate_bps) {
-    int overhead_rate =
-        config->frame_length_ms
-            ? kPacketOverheadBits * 1000 / *config->frame_length_ms
-            : overhead_rate_bps_;
-    // If |metrics.target_audio_bitrate_bps| had included overhead, we would
-    // simply do:
-    //     bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
-    // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
-    // progress regarding this.
-    // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
-    // overhead of most recent packets.
-    bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
-                                   overhead_rate_bps_ - overhead_rate);
-    // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
-    overhead_rate_bps_ = overhead_rate;
+  if (metrics.target_audio_bitrate_bps && metrics.overhead_bytes_per_packet) {
+    // Current implementation of BitrateController can only work when
+    // |metrics.target_audio_bitrate_bps| includes overhead is enabled. This is
+    // currently governed by the following field trial.
+    RTC_DCHECK_EQ("Enabled", webrtc::field_trial::FindFullName(
+                                 "WebRTC-SendSideBwe-WithOverhead"));
+    if (config->frame_length_ms)
+      frame_length_ms_ = *config->frame_length_ms;
+    int overhead_rate_bps =
+        *metrics.overhead_bytes_per_packet * 8 * 1000 / frame_length_ms_;
+    bitrate_bps_ =
+        std::max(0, *metrics.target_audio_bitrate_bps - overhead_rate_bps);
   }
   config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
 }