Merge from Chromium at DEPS revision 03655fd3f6d7

This commit was generated by merge_to_master.py.

Change-Id: I659bdd3b127bb83ee982b3e6117a93ad4fa68d54

86 files changed, 831 insertions, 764 deletions

diff --git a/base/helpers.cc b/base/helpers.cc
index 8b14cdfd..84d1c93b 100644
--- a/base/helpers.cc
+++ b/base/helpers.cc
@@ -47,36 +47,17 @@ class RandomGenerator {
 };
 
 #if defined(SSL_USE_OPENSSL)
-// The OpenSSL RNG. Need to make sure it doesn't run out of entropy.
+// The OpenSSL RNG.
 class SecureRandomGenerator : public RandomGenerator {
  public:
-  SecureRandomGenerator() : inited_(false) {
-  }
-  ~SecureRandomGenerator() {
-  }
+  SecureRandomGenerator() {}
+  ~SecureRandomGenerator() {}
   virtual bool Init(const void* seed, size_t len) {
-    // By default, seed from the system state.
-    if (!inited_) {
-      if (RAND_poll() <= 0) {
-        return false;
-      }
-      inited_ = true;
-    }
-    // Allow app data to be mixed in, if provided.
-    if (seed) {
-      RAND_seed(seed, len);
-    }
     return true;
   }
   virtual bool Generate(void* buf, size_t len) {
-    if (!inited_ && !Init(NULL, 0)) {
-      return false;
-    }
     return (RAND_bytes(reinterpret_cast<unsigned char*>(buf), len) > 0);
   }
-
- private:
-  bool inited_;
 };
 
 #elif defined(SSL_USE_NSS_RNG)
diff --git a/base/openssladapter.cc b/base/openssladapter.cc
index 68a1fcb1..feb01d36 100644
--- a/base/openssladapter.cc
+++ b/base/openssladapter.cc
@@ -34,6 +34,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/openssl.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/sslroots.h"
 #include "webrtc/base/stringutils.h"
 
@@ -141,7 +142,7 @@ static int socket_write(BIO* b, const char* in, int inl) {
 }
 
 static int socket_puts(BIO* b, const char* str) {
-  return socket_write(b, str, strlen(str));
+  return socket_write(b, str, rtc::checked_cast<int>(strlen(str)));
 }
 
 static long socket_ctrl(BIO* b, int cmd, long num, void* ptr) {
@@ -448,7 +449,7 @@ OpenSSLAdapter::Send(const void* pv, size_t cb) {
 
   ssl_write_needs_read_ = false;
 
-  int code = SSL_write(ssl_, pv, cb);
+  int code = SSL_write(ssl_, pv, checked_cast<int>(cb));
   switch (SSL_get_error(ssl_, code)) {
   case SSL_ERROR_NONE:
     //LOG(LS_INFO) << " -- success";
@@ -503,7 +504,7 @@ OpenSSLAdapter::Recv(void* pv, size_t cb) {
 
   ssl_read_needs_write_ = false;
 
-  int code = SSL_read(ssl_, pv, cb);
+  int code = SSL_read(ssl_, pv, checked_cast<int>(cb));
   switch (SSL_get_error(ssl_, code)) {
   case SSL_ERROR_NONE:
     //LOG(LS_INFO) << " -- success";
@@ -843,7 +844,8 @@ bool OpenSSLAdapter::ConfigureTrustedRootCertificates(SSL_CTX* ctx) {
   for (int i = 0; i < ARRAY_SIZE(kSSLCertCertificateList); i++) {
     const unsigned char* cert_buffer = kSSLCertCertificateList[i];
     size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
-    X509* cert = d2i_X509(NULL, &cert_buffer, cert_buffer_len);
+    X509* cert = d2i_X509(NULL, &cert_buffer,
+                          checked_cast<long>(cert_buffer_len));
     if (cert) {
       int return_value = X509_STORE_add_cert(SSL_CTX_get_cert_store(ctx), cert);
       if (return_value == 0) {
diff --git a/base/opensslstreamadapter.cc b/base/opensslstreamadapter.cc
index 133eb72b..d790e4e8 100644
--- a/base/opensslstreamadapter.cc
+++ b/base/opensslstreamadapter.cc
@@ -26,6 +26,7 @@
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/openssl.h"
 #include "webrtc/base/openssladapter.h"
@@ -114,7 +115,7 @@ static int stream_read(BIO* b, char* out, int outl) {
   int error;
   StreamResult result = stream->Read(out, outl, &read, &error);
   if (result == SR_SUCCESS) {
-    return read;
+    return checked_cast<int>(read);
   } else if (result == SR_EOS) {
     b->num = 1;
   } else if (result == SR_BLOCK) {
@@ -132,7 +133,7 @@ static int stream_write(BIO* b, const char* in, int inl) {
   int error;
   StreamResult result = stream->Write(in, inl, &written, &error);
   if (result == SR_SUCCESS) {
-    return written;
+    return checked_cast<int>(written);
   } else if (result == SR_BLOCK) {
     BIO_set_retry_write(b);
   }
@@ -140,7 +141,7 @@ static int stream_write(BIO* b, const char* in, int inl) {
 }
 
 static int stream_puts(BIO* b, const char* str) {
-  return stream_write(b, str, strlen(str));
+  return stream_write(b, str, checked_cast<int>(strlen(str)));
 }
 
 static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) {
@@ -364,7 +365,7 @@ StreamResult OpenSSLStreamAdapter::Write(const void* data, size_t data_len,
 
   ssl_write_needs_read_ = false;
 
-  int code = SSL_write(ssl_, data, data_len);
+  int code = SSL_write(ssl_, data, checked_cast<int>(data_len));
   int ssl_error = SSL_get_error(ssl_, code);
   switch (ssl_error) {
   case SSL_ERROR_NONE:
@@ -425,7 +426,7 @@ StreamResult OpenSSLStreamAdapter::Read(void* data, size_t data_len,
 
   ssl_read_needs_write_ = false;
 
-  int code = SSL_read(ssl_, data, data_len);
+  int code = SSL_read(ssl_, data, checked_cast<int>(data_len));
   int ssl_error = SSL_get_error(ssl_, code);
   switch (ssl_error) {
     case SSL_ERROR_NONE:
diff --git a/base/safe_conversions_impl.h b/base/safe_conversions_impl.h
index 2950f970..77b053a8 100644
--- a/base/safe_conversions_impl.h
+++ b/base/safe_conversions_impl.h
@@ -15,6 +15,8 @@
 
 #include <limits>
 
+#include "webrtc/base/compile_assert.h"
+
 namespace rtc {
 namespace internal {
 
diff --git a/call.h b/call.h
index f21425fb..c6596f86 100644
--- a/call.h
+++ b/call.h
@@ -88,6 +88,14 @@ class Call {
     int stream_start_bitrate_bps;
   };
 
+  struct Stats {
+    Stats() : send_bandwidth_bps(0), recv_bandwidth_bps(0), pacer_delay_ms(0) {}
+
+    int send_bandwidth_bps;
+    int recv_bandwidth_bps;
+    int pacer_delay_ms;
+  };
+
   static Call* Create(const Call::Config& config);
 
   static Call* Create(const Call::Config& config,
@@ -109,13 +117,9 @@ class Call {
   // Call instance exists.
   virtual PacketReceiver* Receiver() = 0;
 
-  // Returns the estimated total send bandwidth. Note: this can differ from the
-  // actual encoded bitrate.
-  virtual uint32_t SendBitrateEstimate() = 0;
-
-  // Returns the total estimated receive bandwidth for the call. Note: this can
-  // differ from the actual receive bitrate.
-  virtual uint32_t ReceiveBitrateEstimate() = 0;
+  // Returns the call statistics, such as estimated send and receive bandwidth,
+  // pacing delay, etc.
+  virtual Stats GetStats() const = 0;
 
   virtual void SignalNetworkState(NetworkState state) = 0;
 
diff --git a/common_types.h b/common_types.h
index 7bcfd6d4..0b4af26e 100644
--- a/common_types.h
+++ b/common_types.h
@@ -271,7 +271,9 @@ class BitrateStatisticsObserver {
  public:
   virtual ~BitrateStatisticsObserver() {}
 
-  virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) = 0;
+  virtual void Notify(const BitrateStatistics& total_stats,
+                      const BitrateStatistics& retransmit_stats,
+                      uint32_t ssrc) = 0;
 };
 
 // Callback, used to notify an observer whenever frame counts have been updated
diff --git a/config.cc b/config.cc
index 70bd8706..357f6367 100644
--- a/config.cc
+++ b/config.cc
@@ -39,13 +39,13 @@ std::string VideoStream::ToString() const {
   ss << ", max_bitrate_bps:" << max_bitrate_bps;
   ss << ", max_qp: " << max_qp;
 
-  ss << ", temporal_layer_thresholds_bps: {";
+  ss << ", temporal_layer_thresholds_bps: [";
   for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
     ss << temporal_layer_thresholds_bps[i];
     if (i != temporal_layer_thresholds_bps.size() - 1)
-      ss << "}, {";
+      ss << ", ";
   }
-  ss << '}';
+  ss << ']';
 
   ss << '}';
   return ss.str();
@@ -54,13 +54,13 @@ std::string VideoStream::ToString() const {
 std::string VideoEncoderConfig::ToString() const {
   std::stringstream ss;
 
-  ss << "{streams: {";
+  ss << "{streams: [";
   for (size_t i = 0; i < streams.size(); ++i) {
     ss << streams[i].ToString();
     if (i != streams.size() - 1)
-      ss << "}, {";
+      ss << ", ";
   }
-  ss << '}';
+  ss << ']';
   ss << ", content_type: ";
   switch (content_type) {
     case kRealtimeVideo:
diff --git a/config.h b/config.h
index 8ea28288..ee1097fc 100644
--- a/config.h
+++ b/config.h
@@ -33,16 +33,19 @@ struct RtpStatistics {
   int extended_max_sequence_number;
 };
 
-struct StreamStats {
-  StreamStats()
+struct SsrcStats {
+  SsrcStats()
       : key_frames(0),
         delta_frames(0),
-        bitrate_bps(0),
+        total_bitrate_bps(0),
+        retransmit_bitrate_bps(0),
         avg_delay_ms(0),
         max_delay_ms(0) {}
   uint32_t key_frames;
   uint32_t delta_frames;
-  int32_t bitrate_bps;
+  // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
+  int total_bitrate_bps;
+  int retransmit_bitrate_bps;
   int avg_delay_ms;
   int max_delay_ms;
   StreamDataCounters rtp_stats;
diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc
index bc20c961..8fa96e50 100644
--- a/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/modules/audio_coding/main/acm2/acm_isac.cc
@@ -277,7 +277,6 @@ ACMISAC::ACMISAC(int16_t codec_id)
     return;
   }
   codec_inst_ptr_->inst = NULL;
-  state_ = codec_inst_ptr_;
 }
 
 ACMISAC::~ACMISAC() {
diff --git a/modules/audio_coding/neteq/audio_decoder.cc b/modules/audio_coding/neteq/audio_decoder.cc
index 04a74eef..d5a27628 100644
--- a/modules/audio_coding/neteq/audio_decoder.cc
+++ b/modules/audio_coding/neteq/audio_decoder.cc
@@ -12,6 +12,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 
 namespace webrtc {
@@ -51,6 +52,11 @@ bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
   return false;
 }
 
+CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
+  FATAL() << "Not a CNG decoder";
+  return NULL;
+}
+
 bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) {
   switch (codec_type) {
     case kDecoderPCMu:
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc
index 07b1b4be..eb078234 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -103,17 +103,17 @@ AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) {
 // iLBC
 #ifdef WEBRTC_CODEC_ILBC
 AudioDecoderIlbc::AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_));
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
 }
 
 AudioDecoderIlbc::~AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_));
+  WebRtcIlbcfix_DecoderFree(dec_state_);
 }
 
 int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_),
+  int16_t ret = WebRtcIlbcfix_Decode(dec_state_,
                                      reinterpret_cast<const int16_t*>(encoded),
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -122,12 +122,11 @@ int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
-  return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_),
-                                decoded, num_frames);
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
 int AudioDecoderIlbc::Init() {
-  return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_));
+  return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 #endif
 
@@ -135,19 +134,18 @@ int AudioDecoderIlbc::Init() {
 #ifdef WEBRTC_CODEC_ISAC
 AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) {
   DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000);
-  WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_));
-  WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_),
-                            decode_sample_rate_hz);
+  WebRtcIsac_Create(&isac_state_);
+  WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz);
 }
 
 AudioDecoderIsac::~AudioDecoderIsac() {
-  WebRtcIsac_Free(static_cast<ISACStruct*>(state_));
+  WebRtcIsac_Free(isac_state_);
 }
 
 int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
+  int16_t ret = WebRtcIsac_Decode(isac_state_,
                                   encoded,
                                   static_cast<int16_t>(encoded_len), decoded,
                                   &temp_type);
@@ -159,7 +157,7 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
                                       size_t encoded_len, int16_t* decoded,
                                       SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
+  int16_t ret = WebRtcIsac_DecodeRcu(isac_state_,
                                      encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -168,12 +166,11 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
 }
 
 int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) {
-  return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_),
-                                 decoded, num_frames);
+  return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames);
 }
 
 int AudioDecoderIsac::Init() {
-  return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_));
+  return WebRtcIsac_DecoderInit(isac_state_);
 }
 
 int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
@@ -181,7 +178,7 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
                                      uint16_t rtp_sequence_number,
                                      uint32_t rtp_timestamp,
                                      uint32_t arrival_timestamp) {
-  return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_),
+  return WebRtcIsac_UpdateBwEstimate(isac_state_,
                                      payload,
                                      static_cast<int32_t>(payload_len),
                                      rtp_sequence_number,
@@ -190,24 +187,24 @@ int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
 }
 
 int AudioDecoderIsac::ErrorCode() {
-  return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_));
+  return WebRtcIsac_GetErrorCode(isac_state_);
 }
 #endif
 
 // iSAC fix
 #ifdef WEBRTC_CODEC_ISACFX
 AudioDecoderIsacFix::AudioDecoderIsacFix() {
-  WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_));
+  WebRtcIsacfix_Create(&isac_state_);
 }
 
 AudioDecoderIsacFix::~AudioDecoderIsacFix() {
-  WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_));
+  WebRtcIsacfix_Free(isac_state_);
 }
 
 int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
                                 int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
+  int16_t ret = WebRtcIsacfix_Decode(isac_state_,
                                      encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
@@ -216,7 +213,7 @@ int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderIsacFix::Init() {
-  return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_));
+  return WebRtcIsacfix_DecoderInit(isac_state_);
 }
 
 int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
@@ -225,32 +222,32 @@ int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
                                         uint32_t rtp_timestamp,
                                         uint32_t arrival_timestamp) {
   return WebRtcIsacfix_UpdateBwEstimate(
-      static_cast<ISACFIX_MainStruct*>(state_),
+      isac_state_,
       payload,
       static_cast<int32_t>(payload_len),
       rtp_sequence_number, rtp_timestamp, arrival_timestamp);
 }
 
 int AudioDecoderIsacFix::ErrorCode() {
-  return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_));
+  return WebRtcIsacfix_GetErrorCode(isac_state_);
 }
 #endif
 
 // G.722
 #ifdef WEBRTC_CODEC_G722
 AudioDecoderG722::AudioDecoderG722() {
-  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_));
+  WebRtcG722_CreateDecoder(&dec_state_);
 }
 
 AudioDecoderG722::~AudioDecoderG722() {
-  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_));
+  WebRtcG722_FreeDecoder(dec_state_);
 }
 
 int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
   int16_t ret = WebRtcG722_Decode(
-      static_cast<G722DecInst*>(state_),
+      dec_state_,
       const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)),
       static_cast<int16_t>(encoded_len), decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -258,7 +255,7 @@ int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderG722::Init() {
-  return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_));
+  return WebRtcG722_DecoderInit(dec_state_);
 }
 
 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@@ -267,18 +264,15 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(2 * encoded_len / channels_);
 }
 
-AudioDecoderG722Stereo::AudioDecoderG722Stereo()
-    : AudioDecoderG722(),
-      state_left_(state_),  // Base member |state_| is used for left channel.
-      state_right_(NULL) {
+AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
   channels_ = 2;
-  // |state_left_| already created by the base class AudioDecoderG722.
-  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_));
+  WebRtcG722_CreateDecoder(&dec_state_left_);
+  WebRtcG722_CreateDecoder(&dec_state_right_);
 }
 
 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
-  // |state_left_| will be freed by the base class AudioDecoderG722.
-  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_));
+  WebRtcG722_FreeDecoder(dec_state_left_);
+  WebRtcG722_FreeDecoder(dec_state_right_);
 }
 
 int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
@@ -289,13 +283,13 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
   SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
   // Decode left and right.
   int16_t ret = WebRtcG722_Decode(
-      static_cast<G722DecInst*>(state_left_),
+      dec_state_left_,
       reinterpret_cast<int16_t*>(encoded_deinterleaved),
       static_cast<int16_t>(encoded_len / 2), decoded, &temp_type);
   if (ret >= 0) {
     int decoded_len = ret;
     ret = WebRtcG722_Decode(
-      static_cast<G722DecInst*>(state_right_),
+      dec_state_right_,
       reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]),
       static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type);
     if (ret == decoded_len) {
@@ -317,11 +311,10 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
 }
 
 int AudioDecoderG722Stereo::Init() {
-  int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_));
-  if (ret != 0) {
-    return ret;
-  }
-  return AudioDecoderG722::Init();
+  int r = WebRtcG722_DecoderInit(dec_state_left_);
+  if (r != 0)
+    return r;
+  return WebRtcG722_DecoderInit(dec_state_right_);
 }
 
 // Split the stereo packet and place left and right channel after each other
@@ -401,18 +394,17 @@ int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) {
 AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
   DCHECK(num_channels == 1 || num_channels == 2);
   channels_ = num_channels;
-  WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_),
-                           static_cast<int>(channels_));
+  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
 }
 
 AudioDecoderOpus::~AudioDecoderOpus() {
-  WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_));
+  WebRtcOpus_DecoderFree(dec_state_);
 }
 
 int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeNew(static_cast<OpusDecInst*>(state_), encoded,
+  int16_t ret = WebRtcOpus_DecodeNew(dec_state_, encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   if (ret > 0)
@@ -425,7 +417,7 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
                                       size_t encoded_len, int16_t* decoded,
                                       SpeechType* speech_type) {
   int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded,
+  int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
                                      static_cast<int16_t>(encoded_len), decoded,
                                      &temp_type);
   if (ret > 0)
@@ -435,12 +427,12 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
 }
 
 int AudioDecoderOpus::Init() {
-  return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_));
+  return WebRtcOpus_DecoderInitNew(dec_state_);
 }
 
 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) {
-  return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_),
+  return WebRtcOpus_DurationEst(dec_state_,
                                 encoded, static_cast<int>(encoded_len));
 }
 
@@ -458,19 +450,15 @@ bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
 #endif
 
 AudioDecoderCng::AudioDecoderCng() {
-  WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_));
-  assert(state_);
+  CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
-  if (state_) {
-    WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_));
-  }
+  WebRtcCng_FreeDec(dec_state_);
 }
 
 int AudioDecoderCng::Init() {
-  assert(state_);
-  return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_));
+  return WebRtcCng_InitDec(dec_state_);
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h
index 214392e7..b30331f3 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -19,6 +19,22 @@
 #include "webrtc/engine_configurations.h"
 #endif
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#endif
 #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
 #include "webrtc/typedefs.h"
 
@@ -109,6 +125,7 @@ class AudioDecoderIlbc : public AudioDecoder {
   virtual int Init();
 
  private:
+  iLBC_decinst_t* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
 };
 #endif
@@ -133,6 +150,7 @@ class AudioDecoderIsac : public AudioDecoder {
   virtual int ErrorCode();
 
  private:
+  ISACStruct* isac_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsac);
 };
 #endif
@@ -153,6 +171,7 @@ class AudioDecoderIsacFix : public AudioDecoder {
   virtual int ErrorCode();
 
  private:
+  ISACFIX_MainStruct* isac_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix);
 };
 #endif
@@ -169,10 +188,11 @@ class AudioDecoderG722 : public AudioDecoder {
   virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len);
 
  private:
+  G722DecInst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
 };
 
-class AudioDecoderG722Stereo : public AudioDecoderG722 {
+class AudioDecoderG722Stereo : public AudioDecoder {
  public:
   AudioDecoderG722Stereo();
   virtual ~AudioDecoderG722Stereo();
@@ -189,8 +209,8 @@ class AudioDecoderG722Stereo : public AudioDecoderG722 {
   void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len,
                          uint8_t* encoded_deinterleaved);
 
-  void* const state_left_;
-  void* state_right_;
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
 };
@@ -229,6 +249,7 @@ class AudioDecoderOpus : public AudioDecoder {
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
  private:
+  OpusDecInst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
 };
 #endif
@@ -252,7 +273,10 @@ class AudioDecoderCng : public AudioDecoder {
                              uint32_t rtp_timestamp,
                              uint32_t arrival_timestamp) { return -1; }
 
+  virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; }
+
  private:
+  CNG_dec_inst* dec_state_;
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
 };
 
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index 31bb40c9..e2be066e 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -36,7 +36,7 @@ int ComfortNoise::UpdateParameters(Packet* packet) {
     return kUnknownPayloadType;
   }
   decoder_database_->SetActiveCngDecoder(packet->header.payloadType);
-  CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
+  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
   int16_t ret = WebRtcCng_UpdateSid(cng_inst,
                                     packet->payload,
                                     packet->payload_length);
@@ -72,7 +72,7 @@ int ComfortNoise::Generate(size_t requested_length,
   if (!cng_decoder) {
     return kUnknownPayloadType;
   }
-  CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
+  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
   // The expression &(*output)[0][0] is a pointer to the first element in
   // the first channel.
   if (WebRtcCng_Generate(cng_inst, &(*output)[0][0],
diff --git a/modules/audio_coding/neteq/interface/audio_decoder.h b/modules/audio_coding/neteq/interface/audio_decoder.h
index 16d78c9e..be85c4dd 100644
--- a/modules/audio_coding/neteq/interface/audio_decoder.h
+++ b/modules/audio_coding/neteq/interface/audio_decoder.h
@@ -14,6 +14,7 @@
 #include <stdlib.h>  // NULL
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -63,7 +64,7 @@ class AudioDecoder {
   // Used by PacketDuration below. Save the value -1 for errors.
   enum { kNotImplemented = -2 };
 
-  AudioDecoder() : channels_(1), state_(NULL) {}
+  AudioDecoder() : channels_(1) {}
   virtual ~AudioDecoder() {}
 
   // Decodes |encode_len| bytes from |encoded| and writes the result in
@@ -114,8 +115,9 @@ class AudioDecoder {
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
-  // Returns the underlying decoder state.
-  void* state() { return state_; }
+  // If this is a CNG decoder, return the underlying CNG_dec_inst*. If this
+  // isn't a CNG decoder, don't call this method.
+  virtual CNG_dec_inst* CngDecoderInstance();
 
   // Returns true if |codec_type| is supported.
   static bool CodecSupported(NetEqDecoder codec_type);
@@ -134,7 +136,6 @@ class AudioDecoder {
   static SpeechType ConvertSpeechType(int16_t type);
 
   size_t channels_;
-  void* state_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 46d03fb8..ca2c1ee5 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -147,9 +147,9 @@ int Normal::Process(const int16_t* input,
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
 
     if (cng_decoder) {
-      CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
       // Generate long enough for 32kHz.
-      if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
+      if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
+                             kCngLength, 0) < 0) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }
diff --git a/modules/audio_device/android/opensles_input.cc b/modules/audio_device/android/opensles_input.cc
index f22d8bf7..e68a6aa2 100644
--- a/modules/audio_device/android/opensles_input.cc
+++ b/modules/audio_device/android/opensles_input.cc
@@ -360,6 +360,24 @@ bool OpenSlesInput::CreateAudioRecorder() {
                                                req),
       false);
 
+  SLAndroidConfigurationItf recorder_config;
+  OPENSL_RETURN_ON_FAILURE(
+      (*sles_recorder_)->GetInterface(sles_recorder_,
+                                      SL_IID_ANDROIDCONFIGURATION,
+                                      &recorder_config),
+      false);
+
+  // Set audio recorder configuration to
+  // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION which ensures that we
+  // use the main microphone tuned for audio communications.
+  SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+  OPENSL_RETURN_ON_FAILURE(
+      (*recorder_config)->SetConfiguration(recorder_config,
+                                           SL_ANDROID_KEY_RECORDING_PRESET,
+                                           &stream_type,
+                                           sizeof(SLint32)),
+      false);
+
   // Realize the recorder in synchronous mode.
   OPENSL_RETURN_ON_FAILURE((*sles_recorder_)->Realize(sles_recorder_,
                                                       SL_BOOLEAN_FALSE),
diff --git a/modules/audio_device/android/opensles_output.cc b/modules/audio_device/android/opensles_output.cc
index 377789b2..487e2840 100644
--- a/modules/audio_device/android/opensles_output.cc
+++ b/modules/audio_device/android/opensles_output.cc
@@ -407,6 +407,24 @@ bool OpenSlesOutput::CreateAudioPlayer() {
                                              &audio_source, &audio_sink,
                                              kNumInterfaces, ids, req),
       false);
+
+  SLAndroidConfigurationItf player_config;
+  OPENSL_RETURN_ON_FAILURE(
+      (*sles_player_)->GetInterface(sles_player_,
+                                    SL_IID_ANDROIDCONFIGURATION,
+                                    &player_config),
+      false);
+
+  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which corresponds
+  // to android.media.AudioManager.STREAM_VOICE_CALL.
+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
+  OPENSL_RETURN_ON_FAILURE(
+      (*player_config)->SetConfiguration(player_config,
+                                         SL_ANDROID_KEY_STREAM_TYPE,
+                                         &stream_type,
+                                         sizeof(SLint32)),
+      false);
+
   // Realize the player in synchronous mode.
   OPENSL_RETURN_ON_FAILURE((*sles_player_)->Realize(sles_player_,
                                                     SL_BOOLEAN_FALSE),
diff --git a/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 47a79ad2..9b55dad7 100644
--- a/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -22,6 +22,7 @@ enum { kBweDecreaseIntervalMs = 300 };
 enum { kLimitNumPackets = 20 };
 enum { kAvgPacketSizeBytes = 1000 };
 enum { kStartPhaseMs = 2000 };
+enum { kBweConverganceTimeMs = 20000 };
 
 // Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
 // The formula in RFC 3448, Section 3.1, is used.
@@ -61,7 +62,8 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
       time_last_decrease_ms_(0),
       first_report_time_ms_(-1),
       initially_lost_packets_(0),
-      uma_updated_(false) {
+      bitrate_at_2_seconds_kbps_(0),
+      uma_update_state_(kNoUpdate) {
 }
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@@ -130,18 +132,35 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
 
   if (first_report_time_ms_ == -1) {
     first_report_time_ms_ = now_ms;
-  } else if (IsInStartPhase(now_ms)) {
-    initially_lost_packets_ += (fraction_loss * number_of_packets) >> 8;
-  } else if (!uma_updated_) {
-    uma_updated_ = true;
+  } else {
+    UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
+                                                 int rtt,
+                                                 int lost_packets) {
+  if (IsInStartPhase(now_ms)) {
+    initially_lost_packets_ += lost_packets;
+  } else if (uma_update_state_ == kNoUpdate) {
+    uma_update_state_ = kFirstDone;
+    bitrate_at_2_seconds_kbps_ = (bitrate_ + 500) / 1000;
     RTC_HISTOGRAM_COUNTS(
         "WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
     RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt, 0, 2000, 50);
     RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
-                         (bitrate_ + 500) / 1000,
+                         bitrate_at_2_seconds_kbps_,
                          0,
                          2000,
                          50);
+  } else if (uma_update_state_ == kFirstDone &&
+             now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
+    uma_update_state_ = kDone;
+    int bitrate_diff_kbps = std::max(
+        bitrate_at_2_seconds_kbps_ - static_cast<int>((bitrate_ + 500) / 1000),
+        0);
+    RTC_HISTOGRAM_COUNTS(
+        "WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
   }
 }
 
diff --git a/modules/bitrate_controller/send_side_bandwidth_estimation.h b/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 0fe3ae67..3361904b 100644
--- a/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -43,8 +43,12 @@ class SendSideBandwidthEstimation {
   void SetMinBitrate(uint32_t min_bitrate);
 
  private:
+  enum UmaState { kNoUpdate, kFirstDone, kDone };
+
   bool IsInStartPhase(int64_t now_ms) const;
 
+  void UpdateUmaStats(int64_t now_ms, int rtt, int lost_packets);
+
   // Returns the input bitrate capped to the thresholds defined by the max,
   // min and incoming bandwidth.
   uint32_t CapBitrateToThresholds(uint32_t bitrate);
@@ -72,7 +76,8 @@ class SendSideBandwidthEstimation {
   uint32_t time_last_decrease_ms_;
   int64_t first_report_time_ms_;
   int initially_lost_packets_;
-  bool uma_updated_;
+  int bitrate_at_2_seconds_kbps_;
+  UmaState uma_update_state_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
diff --git a/modules/pacing/bitrate_prober.cc b/modules/pacing/bitrate_prober.cc
index 04e71c5c..51c87fb2 100644
--- a/modules/pacing/bitrate_prober.cc
+++ b/modules/pacing/bitrate_prober.cc
@@ -55,16 +55,18 @@ void BitrateProber::MaybeInitializeProbe(int bitrate_bps) {
     return;
   probe_bitrates_.clear();
   // Max number of packets used for probing.
-  const int kMaxProbeLength = 15;
-  const int kMaxNumProbes = 3;
-  const int kPacketsPerProbe = kMaxProbeLength / kMaxNumProbes;
-  const float kProbeBitrateMultipliers[kMaxNumProbes] = {2.5, 4, 6};
+  const int kMaxNumProbes = 2;
+  const int kPacketsPerProbe = 5;
+  const float kProbeBitrateMultipliers[kMaxNumProbes] = {3, 6};
   int bitrates_bps[kMaxNumProbes];
   std::stringstream bitrate_log;
   bitrate_log << "Start probing for bandwidth, bitrates:";
   for (int i = 0; i < kMaxNumProbes; ++i) {
     bitrates_bps[i] = kProbeBitrateMultipliers[i] * bitrate_bps;
     bitrate_log << " " << bitrates_bps[i];
+    // We need one extra to get 5 deltas for the first probe.
+    if (i == 0)
+      probe_bitrates_.push_back(bitrates_bps[i]);
     for (int j = 0; j < kPacketsPerProbe; ++j)
       probe_bitrates_.push_back(bitrates_bps[i]);
   }
diff --git a/modules/pacing/bitrate_prober_unittest.cc b/modules/pacing/bitrate_prober_unittest.cc
index 15b1cc58..fac6a721 100644
--- a/modules/pacing/bitrate_prober_unittest.cc
+++ b/modules/pacing/bitrate_prober_unittest.cc
@@ -30,17 +30,11 @@ TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
   EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
   prober.PacketSent(now_ms, 1000);
 
-  for (int i = 0; i < 4; ++i) {
-    EXPECT_EQ(10, prober.TimeUntilNextProbe(now_ms));
-    now_ms += 5;
-    EXPECT_EQ(5, prober.TimeUntilNextProbe(now_ms));
-    now_ms += 5;
-    EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
-    prober.PacketSent(now_ms, 1000);
-  }
   for (int i = 0; i < 5; ++i) {
-    EXPECT_EQ(6, prober.TimeUntilNextProbe(now_ms));
-    now_ms += 6;
+    EXPECT_EQ(8, prober.TimeUntilNextProbe(now_ms));
+    now_ms += 4;
+    EXPECT_EQ(4, prober.TimeUntilNextProbe(now_ms));
+    now_ms += 4;
     EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
     prober.PacketSent(now_ms, 1000);
   }
diff --git a/modules/pacing/paced_sender_unittest.cc b/modules/pacing/paced_sender_unittest.cc
index 34787d16..fe16008c 100644
--- a/modules/pacing/paced_sender_unittest.cc
+++ b/modules/pacing/paced_sender_unittest.cc
@@ -711,13 +711,14 @@ class ProbingPacedSender : public PacedSender {
 };
 
 TEST_F(PacedSenderTest, ProbingWithInitialFrame) {
-  const int kNumPackets = 15;
+  const int kNumPackets = 11;
+  const int kNumDeltas = kNumPackets - 1;
   const int kPacketSize = 1200;
   const int kInitialBitrateKbps = 300;
   uint32_t ssrc = 12346;
   uint16_t sequence_number = 1234;
-  const int expected_deltas[kNumPackets - 1] = {
-      12, 12, 12, 12, 8, 8, 8, 8, 8, 5, 5, 5, 5, 5};
+  const int expected_deltas[kNumDeltas] = {
+      10, 10, 10, 10, 10, 5, 5, 5, 5, 5};
   std::list<int> expected_deltas_list(expected_deltas,
                                       expected_deltas + kNumPackets - 1);
   PacedSenderProbing callback(expected_deltas_list, &clock_);
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index faef7a0b..18225f9b 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -53,12 +53,10 @@ class RtpPacketizer {
 class RtpDepacketizer {
  public:
   struct ParsedPayload {
-    explicit ParsedPayload(WebRtcRTPHeader* rtp_header)
-        : payload(NULL), payload_length(0), header(rtp_header) {}
-
     const uint8_t* payload;
     size_t payload_length;
-    WebRtcRTPHeader* header;
+    FrameType frame_type;
+    RTPTypeHeader type;
   };
 
   static RtpDepacketizer* Create(RtpVideoCodecTypes type);
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.cc b/modules/rtp_rtcp/source/rtp_format_h264.cc
index b6af1ada..0d20b301 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -37,12 +37,15 @@ enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
 // Bit masks for FU (A and B) headers.
 enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
 
-void ParseSingleNalu(WebRtcRTPHeader* rtp_header,
+void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) {
-  rtp_header->type.Video.codec = kRtpVideoH264;
-  rtp_header->type.Video.isFirstPacket = true;
-  RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+  parsed_payload->type.Video.width = 0;
+  parsed_payload->type.Video.height = 0;
+  parsed_payload->type.Video.codec = kRtpVideoH264;
+  parsed_payload->type.Video.isFirstPacket = true;
+  RTPVideoHeaderH264* h264_header =
+      &parsed_payload->type.Video.codecHeader.H264;
   h264_header->single_nalu = true;
   h264_header->stap_a = false;
 
@@ -56,15 +59,15 @@ void ParseSingleNalu(WebRtcRTPHeader* rtp_header,
     case kSps:
     case kPps:
     case kIdr:
-      rtp_header->frameType = kVideoFrameKey;
+      parsed_payload->frame_type = kVideoFrameKey;
       break;
     default:
-      rtp_header->frameType = kVideoFrameDelta;
+      parsed_payload->frame_type = kVideoFrameDelta;
       break;
   }
 }
 
-void ParseFuaNalu(WebRtcRTPHeader* rtp_header,
+void ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                   const uint8_t* payload_data,
                   size_t payload_data_length,
                   size_t* offset) {
@@ -82,13 +85,16 @@ void ParseFuaNalu(WebRtcRTPHeader* rtp_header,
   }
 
   if (original_nal_type == kIdr) {
-    rtp_header->frameType = kVideoFrameKey;
+    parsed_payload->frame_type = kVideoFrameKey;
   } else {
-    rtp_header->frameType = kVideoFrameDelta;
+    parsed_payload->frame_type = kVideoFrameDelta;
   }
-  rtp_header->type.Video.codec = kRtpVideoH264;
-  rtp_header->type.Video.isFirstPacket = first_fragment;
-  RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+  parsed_payload->type.Video.width = 0;
+  parsed_payload->type.Video.height = 0;
+  parsed_payload->type.Video.codec = kRtpVideoH264;
+  parsed_payload->type.Video.isFirstPacket = first_fragment;
+  RTPVideoHeaderH264* h264_header =
+      &parsed_payload->type.Video.codecHeader.H264;
   h264_header->single_nalu = false;
   h264_header->stap_a = false;
 }
@@ -298,12 +304,11 @@ bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
   size_t offset = 0;
   if (nal_type == kFuA) {
     // Fragmented NAL units (FU-A).
-    ParseFuaNalu(
-        parsed_payload->header, payload_data, payload_data_length, &offset);
+    ParseFuaNalu(parsed_payload, payload_data, payload_data_length, &offset);
   } else {
     // We handle STAP-A and single NALU's the same way here. The jitter buffer
     // will depacketize the STAP-A into NAL units later.
-    ParseSingleNalu(parsed_payload->header, payload_data, payload_data_length);
+    ParseSingleNalu(parsed_payload, payload_data, payload_data_length);
   }
 
   parsed_payload->payload = payload_data + offset;
diff --git a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index fb29b5a6..eb690ea8 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -399,17 +399,15 @@ class RtpDepacketizerH264Test : public ::testing::Test {
 
 TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
   uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5.
-
-  WebRtcRTPHeader expected_header;
-  memset(&expected_header, 0, sizeof(expected_header));
-  RtpDepacketizer::ParsedPayload payload(&expected_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(&payload, packet, sizeof(packet));
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
@@ -417,17 +415,15 @@ TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
                         // Length, nal header, payload.
                         0,      0x02, kIdr, 0xFF, 0,    0x03, kIdr, 0xFF,
                         0x00,   0,    0x04, kIdr, 0xFF, 0x00, 0x11};
-
-  WebRtcRTPHeader expected_header;
-  memset(&expected_header, 0, sizeof(expected_header));
-  RtpDepacketizer::ParsedPayload payload(&expected_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(&payload, packet, sizeof(packet));
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
@@ -435,17 +431,15 @@ TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
                         // Length, nal header, payload.
                         0,      0x02, kSlice, 0xFF,   0,    0x03, kSlice, 0xFF,
                         0x00,   0,    0x04,   kSlice, 0xFF, 0x00, 0x11};
-
-  WebRtcRTPHeader expected_header;
-  memset(&expected_header, 0, sizeof(expected_header));
-  RtpDepacketizer::ParsedPayload payload(&expected_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(&payload, packet, sizeof(packet));
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestFuA) {
@@ -470,33 +464,36 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
   };
   const uint8_t kExpected3[1] = {0x03};
 
-  WebRtcRTPHeader expected_header;
-  memset(&expected_header, 0, sizeof(expected_header));
-  RtpDepacketizer::ParsedPayload payload(&expected_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   // We expect that the first packet is one byte shorter since the FU-A header
   // has been replaced by the original nal header.
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet1, sizeof(packet1)));
   ExpectPacket(&payload, kExpected1, sizeof(kExpected1));
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
 
   // Following packets will be 2 bytes shorter since they will only be appended
   // onto the first packet.
+  payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet2, sizeof(packet2)));
   ExpectPacket(&payload, kExpected2, sizeof(kExpected2));
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  EXPECT_FALSE(payload.header->type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_FALSE(payload.type.Video.isFirstPacket);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
 
+  payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet3, sizeof(packet3)));
   ExpectPacket(&payload, kExpected3, sizeof(kExpected3));
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  EXPECT_FALSE(payload.header->type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_FALSE(payload.type.Video.isFirstPacket);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
 }
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 4907846f..ab210ecb 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -90,17 +90,19 @@ bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
                                    const uint8_t* payload_data,
                                    size_t payload_data_length) {
   assert(parsed_payload != NULL);
-  assert(parsed_payload->header != NULL);
 
   uint8_t generic_header = *payload_data++;
   --payload_data_length;
 
-  parsed_payload->header->frameType =
+  parsed_payload->frame_type =
       ((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
           ? kVideoFrameKey
           : kVideoFrameDelta;
-  parsed_payload->header->type.Video.isFirstPacket =
+  parsed_payload->type.Video.isFirstPacket =
       (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
+  parsed_payload->type.Video.codec = kRtpVideoGeneric;
+  parsed_payload->type.Video.width = 0;
+  parsed_payload->type.Video.height = 0;
 
   parsed_payload->payload = payload_data;
   parsed_payload->payload_length = payload_data_length;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.cc b/modules/rtp_rtcp/source/rtp_format_vp8.cc
index 86bdd8bd..d74e04f8 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -121,11 +121,11 @@ int ParseVP8Extension(RTPVideoHeaderVP8* vp8,
   return parsed_bytes;
 }
 
-int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header,
+int ParseVP8FrameSize(RtpDepacketizer::ParsedPayload* parsed_payload,
                       const uint8_t* data,
                       int data_length) {
-  assert(rtp_header != NULL);
-  if (rtp_header->frameType != kVideoFrameKey) {
+  assert(parsed_payload != NULL);
+  if (parsed_payload->frame_type != kVideoFrameKey) {
     // Included in payload header for I-frames.
     return 0;
   }
@@ -134,8 +134,8 @@ int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header,
     // in the beginning of the partition.
     return -1;
   }
-  rtp_header->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
-  rtp_header->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
+  parsed_payload->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
+  parsed_payload->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
   return 0;
 }
 }  // namespace
@@ -664,27 +664,27 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
                                const uint8_t* payload_data,
                                size_t payload_data_length) {
   assert(parsed_payload != NULL);
-  assert(parsed_payload->header != NULL);
 
   // Parse mandatory first byte of payload descriptor.
   bool extension = (*payload_data & 0x80) ? true : false;               // X bit
   bool beginning_of_partition = (*payload_data & 0x10) ? true : false;  // S bit
   int partition_id = (*payload_data & 0x0F);  // PartID field
 
-  parsed_payload->header->type.Video.isFirstPacket =
+  parsed_payload->type.Video.width = 0;
+  parsed_payload->type.Video.height = 0;
+  parsed_payload->type.Video.isFirstPacket =
       beginning_of_partition && (partition_id == 0);
-
-  parsed_payload->header->type.Video.codecHeader.VP8.nonReference =
+  parsed_payload->type.Video.codec = kRtpVideoVp8;
+  parsed_payload->type.Video.codecHeader.VP8.nonReference =
       (*payload_data & 0x20) ? true : false;  // N bit
-  parsed_payload->header->type.Video.codecHeader.VP8.partitionId = partition_id;
-  parsed_payload->header->type.Video.codecHeader.VP8.beginningOfPartition =
+  parsed_payload->type.Video.codecHeader.VP8.partitionId = partition_id;
+  parsed_payload->type.Video.codecHeader.VP8.beginningOfPartition =
       beginning_of_partition;
-  parsed_payload->header->type.Video.codecHeader.VP8.pictureId = kNoPictureId;
-  parsed_payload->header->type.Video.codecHeader.VP8.tl0PicIdx = kNoTl0PicIdx;
-  parsed_payload->header->type.Video.codecHeader.VP8.temporalIdx =
-      kNoTemporalIdx;
-  parsed_payload->header->type.Video.codecHeader.VP8.layerSync = false;
-  parsed_payload->header->type.Video.codecHeader.VP8.keyIdx = kNoKeyIdx;
+  parsed_payload->type.Video.codecHeader.VP8.pictureId = kNoPictureId;
+  parsed_payload->type.Video.codecHeader.VP8.tl0PicIdx = kNoTl0PicIdx;
+  parsed_payload->type.Video.codecHeader.VP8.temporalIdx = kNoTemporalIdx;
+  parsed_payload->type.Video.codecHeader.VP8.layerSync = false;
+  parsed_payload->type.Video.codecHeader.VP8.keyIdx = kNoKeyIdx;
 
   if (partition_id > 8) {
     // Weak check for corrupt payload_data: PartID MUST NOT be larger than 8.
@@ -697,7 +697,7 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
 
   if (extension) {
     const int parsed_bytes =
-        ParseVP8Extension(&parsed_payload->header->type.Video.codecHeader.VP8,
+        ParseVP8Extension(&parsed_payload->type.Video.codecHeader.VP8,
                           payload_data,
                           payload_data_length);
     if (parsed_bytes < 0)
@@ -713,14 +713,14 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
 
   // Read P bit from payload header (only at beginning of first partition).
   if (payload_data_length > 0 && beginning_of_partition && partition_id == 0) {
-    parsed_payload->header->frameType =
+    parsed_payload->frame_type =
         (*payload_data & 0x01) ? kVideoFrameDelta : kVideoFrameKey;
   } else {
-    parsed_payload->header->frameType = kVideoFrameDelta;
+    parsed_payload->frame_type = kVideoFrameDelta;
   }
 
-  if (0 != ParseVP8FrameSize(
-               parsed_payload->header, payload_data, payload_data_length)) {
+  if (ParseVP8FrameSize(parsed_payload, payload_data, payload_data_length) !=
+      0) {
     return false;
   }
 
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index b13f8793..4382ac2c 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -56,24 +56,23 @@ namespace {
 //      | padding       |
 //      :               :
 //      +-+-+-+-+-+-+-+-+
-
-void VerifyBasicHeader(WebRtcRTPHeader* header, bool N, bool S, int part_id) {
-  ASSERT_TRUE(header != NULL);
-  EXPECT_EQ(N, header->type.Video.codecHeader.VP8.nonReference);
-  EXPECT_EQ(S, header->type.Video.codecHeader.VP8.beginningOfPartition);
-  EXPECT_EQ(part_id, header->type.Video.codecHeader.VP8.partitionId);
+void VerifyBasicHeader(RTPTypeHeader* type, bool N, bool S, int part_id) {
+  ASSERT_TRUE(type != NULL);
+  EXPECT_EQ(N, type->Video.codecHeader.VP8.nonReference);
+  EXPECT_EQ(S, type->Video.codecHeader.VP8.beginningOfPartition);
+  EXPECT_EQ(part_id, type->Video.codecHeader.VP8.partitionId);
 }
 
-void VerifyExtensions(WebRtcRTPHeader* header,
+void VerifyExtensions(RTPTypeHeader* type,
                       int16_t picture_id,   /* I */
                       int16_t tl0_pic_idx,  /* L */
                       uint8_t temporal_idx, /* T */
                       int key_idx /* K */) {
-  ASSERT_TRUE(header != NULL);
-  EXPECT_EQ(picture_id, header->type.Video.codecHeader.VP8.pictureId);
-  EXPECT_EQ(tl0_pic_idx, header->type.Video.codecHeader.VP8.tl0PicIdx);
-  EXPECT_EQ(temporal_idx, header->type.Video.codecHeader.VP8.temporalIdx);
-  EXPECT_EQ(key_idx, header->type.Video.codecHeader.VP8.keyIdx);
+  ASSERT_TRUE(type != NULL);
+  EXPECT_EQ(picture_id, type->Video.codecHeader.VP8.pictureId);
+  EXPECT_EQ(tl0_pic_idx, type->Video.codecHeader.VP8.tl0PicIdx);
+  EXPECT_EQ(temporal_idx, type->Video.codecHeader.VP8.temporalIdx);
+  EXPECT_EQ(key_idx, type->Video.codecHeader.VP8.keyIdx);
 }
 }  // namespace
 
@@ -405,18 +404,16 @@ TEST_F(RtpDepacketizerVp8Test, BasicHeader) {
   uint8_t packet[4] = {0};
   packet[0] = 0x14;  // Binary 0001 0100; S = 1, PartID = 4.
   packet[1] = 0x01;  // P frame.
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 0, 1, 4);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 0, 1, 4);
   VerifyExtensions(
-      payload.header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+      &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
 }
 
 TEST_F(RtpDepacketizerVp8Test, PictureID) {
@@ -427,29 +424,27 @@ TEST_F(RtpDepacketizerVp8Test, PictureID) {
   packet[0] = 0xA0;
   packet[1] = 0x80;
   packet[2] = kPictureId;
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength1, sizeof(packet) - kHeaderLength1);
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 1, 0, 0);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 1, 0, 0);
   VerifyExtensions(
-      payload.header, kPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+      &payload.type, kPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
 
   // Re-use packet, but change to long PictureID.
   packet[2] = 0x80 | kPictureId;
   packet[3] = kPictureId;
-  memset(payload.header, 0, sizeof(rtp_header));
 
+  payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength2, sizeof(packet) - kHeaderLength2);
-  VerifyBasicHeader(payload.header, 1, 0, 0);
-  VerifyExtensions(payload.header,
+  VerifyBasicHeader(&payload.type, 1, 0, 0);
+  VerifyExtensions(&payload.type,
                    (kPictureId << 8) + kPictureId,
                    kNoTl0PicIdx,
                    kNoTemporalIdx,
@@ -463,18 +458,16 @@ TEST_F(RtpDepacketizerVp8Test, Tl0PicIdx) {
   packet[0] = 0x90;
   packet[1] = 0x40;
   packet[2] = kTl0PicIdx;
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 0, 1, 0);
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 0, 1, 0);
   VerifyExtensions(
-      payload.header, kNoPictureId, kTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+      &payload.type, kNoPictureId, kTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
 }
 
 TEST_F(RtpDepacketizerVp8Test, TIDAndLayerSync) {
@@ -483,18 +476,16 @@ TEST_F(RtpDepacketizerVp8Test, TIDAndLayerSync) {
   packet[0] = 0x88;
   packet[1] = 0x20;
   packet[2] = 0x80;  // TID(2) + LayerSync(false)
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 0, 0, 8);
-  VerifyExtensions(payload.header, kNoPictureId, kNoTl0PicIdx, 2, kNoKeyIdx);
-  EXPECT_FALSE(payload.header->type.Video.codecHeader.VP8.layerSync);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 0, 0, 8);
+  VerifyExtensions(&payload.type, kNoPictureId, kNoTl0PicIdx, 2, kNoKeyIdx);
+  EXPECT_FALSE(payload.type.Video.codecHeader.VP8.layerSync);
 }
 
 TEST_F(RtpDepacketizerVp8Test, KeyIdx) {
@@ -504,18 +495,16 @@ TEST_F(RtpDepacketizerVp8Test, KeyIdx) {
   packet[0] = 0x88;
   packet[1] = 0x10;  // K = 1.
   packet[2] = kKeyIdx;
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 0, 0, 8);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 0, 0, 8);
   VerifyExtensions(
-      payload.header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kKeyIdx);
+      &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kKeyIdx);
 }
 
 TEST_F(RtpDepacketizerVp8Test, MultipleExtensions) {
@@ -527,17 +516,15 @@ TEST_F(RtpDepacketizerVp8Test, MultipleExtensions) {
   packet[3] = 17;                  // PictureID, low 8 bits.
   packet[4] = 42;                  // Tl0PicIdx.
   packet[5] = 0x40 | 0x20 | 0x11;  // TID(1) + LayerSync(true) + KEYIDX(17).
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 0, 0, 8);
-  VerifyExtensions(payload.header, (17 << 8) + 17, 42, 1, 17);
+  EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 0, 0, 8);
+  VerifyExtensions(&payload.type, (17 << 8) + 17, 42, 1, 17);
 }
 
 TEST_F(RtpDepacketizerVp8Test, TooShortHeader) {
@@ -546,10 +533,7 @@ TEST_F(RtpDepacketizerVp8Test, TooShortHeader) {
   packet[1] = 0x80 | 0x40 | 0x20 | 0x10;  // All extensions are enabled...
   packet[2] = 0x80 | 17;  // ... but only 2 bytes PictureID is provided.
   packet[3] = 17;         // PictureID, low 8 bits.
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   EXPECT_FALSE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
 }
@@ -571,23 +555,20 @@ TEST_F(RtpDepacketizerVp8Test, TestWithPacketizer) {
   size_t send_bytes;
   ASSERT_TRUE(packetizer.NextPacket(packet, &send_bytes, &last));
   ASSERT_TRUE(last);
-
-  WebRtcRTPHeader rtp_header;
-  memset(&rtp_header, 0, sizeof(rtp_header));
-  RtpDepacketizer::ParsedPayload payload(&rtp_header);
+  RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(
       &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
-  EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
-  VerifyBasicHeader(payload.header, 1, 1, 0);
-  VerifyExtensions(payload.header,
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+  VerifyBasicHeader(&payload.type, 1, 1, 0);
+  VerifyExtensions(&payload.type,
                    input_header.pictureId,
                    input_header.tl0PicIdx,
                    input_header.temporalIdx,
                    input_header.keyIdx);
-  EXPECT_EQ(payload.header->type.Video.codecHeader.VP8.layerSync,
+  EXPECT_EQ(payload.type.Video.codecHeader.VP8.layerSync,
             input_header.layerSync);
 }
-
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc
index dfbf35ae..6f6d6470 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -79,13 +79,15 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
   }
 
   rtp_header->type.Video.isFirstPacket = is_first_packet;
-  RtpDepacketizer::ParsedPayload parsed_payload(rtp_header);
+  RtpDepacketizer::ParsedPayload parsed_payload;
   if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
     return -1;
 
+  rtp_header->frameType = parsed_payload.frame_type;
+  rtp_header->type = parsed_payload.type;
   return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
                                                parsed_payload.payload_length,
-                                               parsed_payload.header) == 0
+                                               rtp_header) == 0
              ? 0
              : -1;
 }
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 0438b9f7..677f3fc4 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -40,6 +40,57 @@ const char* FrameTypeToString(const FrameType frame_type) {
 
 }  // namespace
 
+class BitrateAggregator {
+ public:
+  explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback)
+      : callback_(bitrate_callback),
+        total_bitrate_observer_(*this),
+        retransmit_bitrate_observer_(*this),
+        ssrc_(0) {}
+
+  void OnStatsUpdated() const {
+    if (callback_)
+      callback_->Notify(total_bitrate_observer_.statistics(),
+                        retransmit_bitrate_observer_.statistics(),
+                        ssrc_);
+  }
+
+  Bitrate::Observer* total_bitrate_observer() {
+    return &total_bitrate_observer_;
+  }
+  Bitrate::Observer* retransmit_bitrate_observer() {
+    return &retransmit_bitrate_observer_;
+  }
+
+  void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
+
+ private:
+  // We assume that these observers are called on the same thread, which is
+  // true for RtpSender as they are called on the Process thread.
+  class BitrateObserver : public Bitrate::Observer {
+   public:
+    explicit BitrateObserver(const BitrateAggregator& aggregator)
+        : aggregator_(aggregator) {}
+
+    // Implements Bitrate::Observer.
+    virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE {
+      statistics_ = stats;
+      aggregator_.OnStatsUpdated();
+    }
+
+    BitrateStatistics statistics() const { return statistics_; }
+
+   private:
+    BitrateStatistics statistics_;
+    const BitrateAggregator& aggregator_;
+  };
+
+  BitrateStatisticsObserver* const callback_;
+  BitrateObserver total_bitrate_observer_;
+  BitrateObserver retransmit_bitrate_observer_;
+  uint32_t ssrc_;
+};
+
 RTPSender::RTPSender(const int32_t id,
                      const bool audio,
                      Clock* clock,
@@ -54,7 +105,8 @@ RTPSender::RTPSender(const int32_t id,
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
-      bitrate_sent_(clock, this),
+      bitrates_(new BitrateAggregator(bitrate_callback)),
+      total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
       id_(id),
       audio_configured_(audio),
       audio_(NULL),
@@ -74,12 +126,11 @@ RTPSender::RTPSender(const int32_t id,
       // NACK.
       nack_byte_count_times_(),
       nack_byte_count_(),
-      nack_bitrate_(clock, NULL),
+      nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()),
       packet_history_(clock),
       // Statistics
       statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_stats_callback_(NULL),
-      bitrate_callback_(bitrate_callback),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       // RTP variables
@@ -108,6 +159,7 @@ RTPSender::RTPSender(const int32_t id,
   srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
   ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
   ssrc_rtx_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+  bitrates_->set_ssrc(ssrc_);
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
   sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
@@ -149,7 +201,7 @@ uint32_t RTPSender::GetTargetBitrate() {
 }
 
 uint16_t RTPSender::ActualSendBitrateKbit() const {
-  return (uint16_t)(bitrate_sent_.BitrateNow() / 1000);
+  return (uint16_t)(total_bitrate_sent_.BitrateNow() / 1000);
 }
 
 uint32_t RTPSender::VideoBitrateSent() const {
@@ -864,7 +916,7 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer,
     counters = &rtp_stats_;
   }
 
-  bitrate_sent_.Update(size);
+  total_bitrate_sent_.Update(size);
   ++counters->packets;
   if (IsFecPacket(buffer, header)) {
     ++counters->fec_packets;
@@ -997,7 +1049,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
 
 void RTPSender::ProcessBitrate() {
   CriticalSectionScoped cs(send_critsect_);
-  bitrate_sent_.Process();
+  total_bitrate_sent_.Process();
   nack_bitrate_.Process();
   if (audio_configured_) {
     return;
@@ -1420,6 +1472,7 @@ void RTPSender::SetSendingStatus(bool enabled) {
       // Generate a new SSRC.
       ssrc_db_.ReturnSSRC(ssrc_);
       ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+      bitrates_->set_ssrc(ssrc_);
     }
     // Don't initialize seq number if SSRC passed externally.
     if (!sequence_number_forced_ && !ssrc_forced_) {
@@ -1470,6 +1523,7 @@ uint32_t RTPSender::GenerateNewSSRC() {
     return 0;
   }
   ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+  bitrates_->set_ssrc(ssrc_);
   return ssrc_;
 }
 
@@ -1484,6 +1538,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
   ssrc_db_.ReturnSSRC(ssrc_);
   ssrc_db_.RegisterSSRC(ssrc);
   ssrc_ = ssrc;
+  bitrates_->set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
     sequence_number_ =
         rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);  // NOLINT
@@ -1681,17 +1736,8 @@ StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
   return rtp_stats_callback_;
 }
 
-uint32_t RTPSender::BitrateSent() const { return bitrate_sent_.BitrateLast(); }
-
-void RTPSender::BitrateUpdated(const BitrateStatistics& stats) {
-  uint32_t ssrc;
-  {
-    CriticalSectionScoped ssrc_lock(send_critsect_);
-    ssrc = ssrc_;
-  }
-  if (bitrate_callback_) {
-    bitrate_callback_->Notify(stats, ssrc);
-  }
+uint32_t RTPSender::BitrateSent() const {
+  return total_bitrate_sent_.BitrateLast();
 }
 
 void RTPSender::SetRtpState(const RtpState& rtp_state) {
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 780baa1f..6564d47b 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -31,6 +31,7 @@
 
 namespace webrtc {
 
+class BitrateAggregator;
 class CriticalSectionWrapper;
 class RTPSenderAudio;
 class RTPSenderVideo;
@@ -65,7 +66,7 @@ class RTPSenderInterface {
       PacedSender::Priority priority) = 0;
 };
 
-class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
+class RTPSender : public RTPSenderInterface {
  public:
   RTPSender(const int32_t id, const bool audio, Clock *clock,
             Transport *transport, RtpAudioFeedback *audio_feedback,
@@ -276,8 +277,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
 
   uint32_t BitrateSent() const;
 
-  virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE;
-
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
@@ -337,7 +336,9 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
 
   Clock* clock_;
   int64_t clock_delta_ms_;
-  Bitrate bitrate_sent_;
+
+  scoped_ptr<BitrateAggregator> bitrates_;
+  Bitrate total_bitrate_sent_;
 
   int32_t id_;
   const bool audio_configured_;
@@ -375,7 +376,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
-  BitrateStatisticsObserver* const bitrate_callback_;
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 9c6a7207..2a494778 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -855,20 +855,22 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
 TEST_F(RtpSenderTest, BitrateCallbacks) {
   class TestCallback : public BitrateStatisticsObserver {
    public:
-    TestCallback()
-        : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0), bitrate_() {}
+    TestCallback() : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0) {}
     virtual ~TestCallback() {}
 
-    virtual void Notify(const BitrateStatistics& stats,
+    virtual void Notify(const BitrateStatistics& total_stats,
+                        const BitrateStatistics& retransmit_stats,
                         uint32_t ssrc) OVERRIDE {
       ++num_calls_;
       ssrc_ = ssrc;
-      bitrate_ = stats;
+      total_stats_ = total_stats;
+      retransmit_stats_ = retransmit_stats;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
-    BitrateStatistics bitrate_;
+    BitrateStatistics total_stats_;
+    BitrateStatistics retransmit_stats_;
   } callback;
   rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
                                   &mock_paced_sender_, &callback, NULL, NULL));
@@ -909,13 +911,15 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
 
   const uint32_t expected_packet_rate = 1000 / kPacketInterval;
 
-  EXPECT_EQ(1U, callback.num_calls_);
+  // We get one call for every stats updated, thus two calls since both the
+  // stream stats and the retransmit stats are updated once.
+  EXPECT_EQ(2u, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(start_time + (kNumPackets * kPacketInterval),
-            callback.bitrate_.timestamp_ms);
-  EXPECT_EQ(expected_packet_rate, callback.bitrate_.packet_rate);
+            callback.total_stats_.timestamp_ms);
+  EXPECT_EQ(expected_packet_rate, callback.total_stats_.packet_rate);
   EXPECT_EQ((kPacketOverhead + sizeof(payload)) * 8 * expected_packet_rate,
-            callback.bitrate_.bitrate_bps);
+            callback.total_stats_.bitrate_bps);
 
   rtp_sender_.reset();
 }
diff --git a/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 73f774d6..420ef595 100644
--- a/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -604,7 +604,7 @@ TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossVP9) {
                      false, true, false);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 37.5, 36.0, 0.94, 0.93);
+  SetQualityMetrics(&quality_metrics, 37.0, 36.0, 0.93, 0.92);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
   SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
@@ -657,12 +657,12 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeBitRateVP9) {
                      false, true, false);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 36.0, 32.0, 0.90, 0.85);
+  SetQualityMetrics(&quality_metrics, 36.0, 31.8, 0.90, 0.85);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
   SetRateControlMetrics(rc_metrics, 0, 0, 30, 20, 20, 20, 0);
   SetRateControlMetrics(rc_metrics, 1, 2, 0, 20, 20, 60, 0);
-  SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 20, 30, 0);
+  SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 20, 40, 0);
   ProcessFramesAndVerify(quality_metrics,
                          rate_profile,
                          process_settings,
@@ -692,11 +692,11 @@ TEST_F(VideoProcessorIntegrationTest,
                      false, true, false);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 30.0, 18.0, 0.80, 0.40);
+  SetQualityMetrics(&quality_metrics, 29.0, 17.0, 0.80, 0.40);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 35, 55, 70, 15, 40, 0);
-  SetRateControlMetrics(rc_metrics, 1, 15, 0, 50, 10, 30, 0);
+  SetRateControlMetrics(rc_metrics, 0, 50, 60, 100, 15, 45, 0);
+  SetRateControlMetrics(rc_metrics, 1, 30, 0, 65, 10, 35, 0);
   SetRateControlMetrics(rc_metrics, 2, 5, 0, 38, 10, 30, 0);
   ProcessFramesAndVerify(quality_metrics,
                          rate_profile,
diff --git a/modules/video_coding/codecs/vp9/vp9_impl.cc b/modules/video_coding/codecs/vp9/vp9_impl.cc
index 33f11a37..734e73d2 100644
--- a/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -189,11 +189,8 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
   // Only positive speeds, currently: 0 - 7.
-  // O means slowest/best quality, 7 means fastest/lowest quality.
-  // TODO(marpan): Speeds 5-7 are speed settings for real-time mode, on desktop.
-  // Currently set to 5, update to 6 (for faster encoding) after some subjective
-  // quality tests.
-  cpu_speed_ = 5;
+  // O means slowest/best quality, 7 means fastest/lower quality.
+  cpu_speed_ = 6;
   // Note: some of these codec controls still use "VP8" in the control name.
   // TODO(marpan): Update this in the next/future libvpx version.
   vpx_codec_control(encoder_, VP8E_SET_CPUUSED, cpu_speed_);
diff --git a/modules/video_coding/main/source/media_optimization.cc b/modules/video_coding/main/source/media_optimization.cc
index 0d9a4bdf..57894807 100644
--- a/modules/video_coding/main/source/media_optimization.cc
+++ b/modules/video_coding/main/source/media_optimization.cc
@@ -542,7 +542,7 @@ void MediaOptimization::UpdateSentBitrate(int64_t now_ms) {
       now_ms - encoded_frame_samples_.front().time_complete_ms);
   if (denom >= 1.0f) {
     avg_sent_bit_rate_bps_ =
-        static_cast<uint32_t>(framesize_sum * 8 * 1000 / denom + 0.5f);
+        static_cast<uint32_t>(framesize_sum * 8.0f * 1000.0f / denom + 0.5f);
   } else {
     avg_sent_bit_rate_bps_ = framesize_sum * 8;
   }
diff --git a/modules/video_render/android/video_render_android_native_opengl2.h b/modules/video_render/android/video_render_android_native_opengl2.h
index 66c53641..f5e5b57e 100644
--- a/modules/video_render/android/video_render_android_native_opengl2.h
+++ b/modules/video_render/android/video_render_android_native_opengl2.h
@@ -41,7 +41,7 @@ class AndroidNativeOpenGl2Channel: public AndroidStream {
   virtual void DeliverFrame(JNIEnv* jniEnv);
 
  private:
-  static jint CreateOpenGLNativeStatic(
+  static jint JNICALL CreateOpenGLNativeStatic(
       JNIEnv * env,
       jobject,
       jlong context,
@@ -49,7 +49,7 @@ class AndroidNativeOpenGl2Channel: public AndroidStream {
       jint height);
   jint CreateOpenGLNative(int width, int height);
 
-  static void DrawNativeStatic(JNIEnv * env,jobject, jlong context);
+  static void JNICALL DrawNativeStatic(JNIEnv * env,jobject, jlong context);
   void DrawNative();
   uint32_t _id;
   CriticalSectionWrapper& _renderCritSect;
diff --git a/p2p/base/basicpacketsocketfactory.cc b/p2p/base/basicpacketsocketfactory.cc
index 06dfe76e..9b12e78d 100644
--- a/p2p/base/basicpacketsocketfactory.cc
+++ b/p2p/base/basicpacketsocketfactory.cc
@@ -44,7 +44,7 @@ BasicPacketSocketFactory::~BasicPacketSocketFactory() {
 }
 
 AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
-    const SocketAddress& address, int min_port, int max_port) {
+    const SocketAddress& address, uint16 min_port, uint16 max_port) {
   // UDP sockets are simple.
   rtc::AsyncSocket* socket =
       socket_factory()->CreateAsyncSocket(
@@ -62,7 +62,8 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
 }
 
 AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
-    const SocketAddress& local_address, int min_port, int max_port, int opts) {
+    const SocketAddress& local_address, uint16 min_port, uint16 max_port,
+    int opts) {
 
   // Fail if TLS is required.
   if (opts & PacketSocketFactory::OPT_TLS) {
@@ -177,7 +178,7 @@ AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
 
 int BasicPacketSocketFactory::BindSocket(
     AsyncSocket* socket, const SocketAddress& local_address,
-    int min_port, int max_port) {
+    uint16 min_port, uint16 max_port) {
   int ret = -1;
   if (min_port == 0 && max_port == 0) {
     // If there's no port range, let the OS pick a port for us.
diff --git a/p2p/base/basicpacketsocketfactory.h b/p2p/base/basicpacketsocketfactory.h
index fb3a5269..b23a6772 100644
--- a/p2p/base/basicpacketsocketfactory.h
+++ b/p2p/base/basicpacketsocketfactory.h
@@ -24,21 +24,28 @@ class BasicPacketSocketFactory : public PacketSocketFactory {
   BasicPacketSocketFactory();
   explicit BasicPacketSocketFactory(Thread* thread);
   explicit BasicPacketSocketFactory(SocketFactory* socket_factory);
-  virtual ~BasicPacketSocketFactory();
-
-  virtual AsyncPacketSocket* CreateUdpSocket(
-      const SocketAddress& local_address, int min_port, int max_port);
-  virtual AsyncPacketSocket* CreateServerTcpSocket(
-      const SocketAddress& local_address, int min_port, int max_port, int opts);
-  virtual AsyncPacketSocket* CreateClientTcpSocket(
-      const SocketAddress& local_address, const SocketAddress& remote_address,
-      const ProxyInfo& proxy_info, const std::string& user_agent, int opts);
-
-  virtual AsyncResolverInterface* CreateAsyncResolver();
+  ~BasicPacketSocketFactory() override;
+
+  AsyncPacketSocket* CreateUdpSocket(const SocketAddress& local_address,
+                                     uint16 min_port,
+                                     uint16 max_port) override;
+  AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
+                                           uint16 min_port,
+                                           uint16 max_port,
+                                           int opts) override;
+  AsyncPacketSocket* CreateClientTcpSocket(const SocketAddress& local_address,
+                                           const SocketAddress& remote_address,
+                                           const ProxyInfo& proxy_info,
+                                           const std::string& user_agent,
+                                           int opts) override;
+
+  AsyncResolverInterface* CreateAsyncResolver() override;
 
  private:
-  int BindSocket(AsyncSocket* socket, const SocketAddress& local_address,
-                 int min_port, int max_port);
+  int BindSocket(AsyncSocket* socket,
+                 const SocketAddress& local_address,
+                 uint16 min_port,
+                 uint16 max_port);
 
   SocketFactory* socket_factory();
 
diff --git a/p2p/base/packetsocketfactory.h b/p2p/base/packetsocketfactory.h
index 1f45feca..d2d7b1b1 100644
--- a/p2p/base/packetsocketfactory.h
+++ b/p2p/base/packetsocketfactory.h
@@ -29,17 +29,23 @@ class PacketSocketFactory {
   PacketSocketFactory() { }
   virtual ~PacketSocketFactory() { }
 
-  virtual AsyncPacketSocket* CreateUdpSocket(
-      const SocketAddress& address, int min_port, int max_port) = 0;
+  virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
+                                             uint16 min_port,
+                                             uint16 max_port) = 0;
   virtual AsyncPacketSocket* CreateServerTcpSocket(
-      const SocketAddress& local_address, int min_port, int max_port,
+      const SocketAddress& local_address,
+      uint16 min_port,
+      uint16 max_port,
       int opts) = 0;
 
   // TODO: |proxy_info| and |user_agent| should be set
   // per-factory and not when socket is created.
   virtual AsyncPacketSocket* CreateClientTcpSocket(
-      const SocketAddress& local_address, const SocketAddress& remote_address,
-      const ProxyInfo& proxy_info, const std::string& user_agent, int opts) = 0;
+      const SocketAddress& local_address,
+      const SocketAddress& remote_address,
+      const ProxyInfo& proxy_info,
+      const std::string& user_agent,
+      int opts) = 0;
 
   virtual AsyncResolverInterface* CreateAsyncResolver() = 0;
 
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index f569d9f5..a8357ad6 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -152,9 +152,12 @@ static std::string ComputeFoundation(
   return rtc::ToString<uint32>(rtc::ComputeCrc32(ost.str()));
 }
 
-Port::Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-           rtc::Network* network, const rtc::IPAddress& ip,
-           const std::string& username_fragment, const std::string& password)
+Port::Port(rtc::Thread* thread,
+           rtc::PacketSocketFactory* factory,
+           rtc::Network* network,
+           const rtc::IPAddress& ip,
+           const std::string& username_fragment,
+           const std::string& password)
     : thread_(thread),
       factory_(factory),
       send_retransmit_count_attribute_(false),
@@ -176,10 +179,14 @@ Port::Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
   Construct();
 }
 
-Port::Port(rtc::Thread* thread, const std::string& type,
+Port::Port(rtc::Thread* thread,
+           const std::string& type,
            rtc::PacketSocketFactory* factory,
-           rtc::Network* network, const rtc::IPAddress& ip,
-           int min_port, int max_port, const std::string& username_fragment,
+           rtc::Network* network,
+           const rtc::IPAddress& ip,
+           uint16 min_port,
+           uint16 max_port,
+           const std::string& username_fragment,
            const std::string& password)
     : thread_(thread),
       factory_(factory),
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 48b85302..87072e67 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -107,13 +107,20 @@ typedef std::set<rtc::SocketAddress> ServerAddresses;
 class Port : public PortInterface, public rtc::MessageHandler,
              public sigslot::has_slots<> {
  public:
-  Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-       rtc::Network* network, const rtc::IPAddress& ip,
-       const std::string& username_fragment, const std::string& password);
-  Port(rtc::Thread* thread, const std::string& type,
+  Port(rtc::Thread* thread,
        rtc::PacketSocketFactory* factory,
-       rtc::Network* network, const rtc::IPAddress& ip,
-       int min_port, int max_port, const std::string& username_fragment,
+       rtc::Network* network,
+       const rtc::IPAddress& ip,
+       const std::string& username_fragment,
+       const std::string& password);
+  Port(rtc::Thread* thread,
+       const std::string& type,
+       rtc::PacketSocketFactory* factory,
+       rtc::Network* network,
+       const rtc::IPAddress& ip,
+       uint16 min_port,
+       uint16 max_port,
+       const std::string& username_fragment,
        const std::string& password);
   virtual ~Port();
 
@@ -256,8 +263,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // Debugging description of this port
   virtual std::string ToString() const;
   rtc::IPAddress& ip() { return ip_; }
-  int min_port() { return min_port_; }
-  int max_port() { return max_port_; }
+  uint16 min_port() { return min_port_; }
+  uint16 max_port() { return max_port_; }
 
   // Timeout shortening function to speed up unit tests.
   void set_timeout_delay(int delay) { timeout_delay_ = delay; }
@@ -354,8 +361,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
   bool send_retransmit_count_attribute_;
   rtc::Network* network_;
   rtc::IPAddress ip_;
-  int min_port_;
-  int max_port_;
+  uint16 min_port_;
+  uint16 max_port_;
   std::string content_name_;
   int component_;
   uint32 generation_;
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 8805709a..f09db284 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -100,12 +100,17 @@ static bool WriteStunMessage(const StunMessage* msg, ByteBuffer* buf) {
 // Stub port class for testing STUN generation and processing.
 class TestPort : public Port {
  public:
-  TestPort(rtc::Thread* thread, const std::string& type,
-           rtc::PacketSocketFactory* factory, rtc::Network* network,
-           const rtc::IPAddress& ip, int min_port, int max_port,
-           const std::string& username_fragment, const std::string& password)
-      : Port(thread, type, factory, network, ip,
-             min_port, max_port, username_fragment, password) {
+  TestPort(rtc::Thread* thread,
+           const std::string& type,
+           rtc::PacketSocketFactory* factory,
+           rtc::Network* network,
+           const rtc::IPAddress& ip,
+           uint16 min_port,
+           uint16 max_port,
+           const std::string& username_fragment,
+           const std::string& password)
+      : Port(thread, type, factory, network, ip, min_port, max_port,
+             username_fragment, password) {
   }
   ~TestPort() {}
 
@@ -762,19 +767,21 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
         next_server_tcp_socket_(NULL),
         next_client_tcp_socket_(NULL) {
   }
-  virtual ~FakePacketSocketFactory() { }
+  ~FakePacketSocketFactory() override { }
 
-  virtual AsyncPacketSocket* CreateUdpSocket(
-      const SocketAddress& address, int min_port, int max_port) {
+  AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
+                                     uint16 min_port,
+                                     uint16 max_port) override {
     EXPECT_TRUE(next_udp_socket_ != NULL);
     AsyncPacketSocket* result = next_udp_socket_;
     next_udp_socket_ = NULL;
     return result;
   }
 
-  virtual AsyncPacketSocket* CreateServerTcpSocket(
-      const SocketAddress& local_address, int min_port, int max_port,
-      int opts) {
+  AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
+                                           uint16 min_port,
+                                           uint16 max_port,
+                                           int opts) override {
     EXPECT_TRUE(next_server_tcp_socket_ != NULL);
     AsyncPacketSocket* result = next_server_tcp_socket_;
     next_server_tcp_socket_ = NULL;
@@ -783,10 +790,11 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
 
   // TODO: |proxy_info| and |user_agent| should be set
   // per-factory and not when socket is created.
-  virtual AsyncPacketSocket* CreateClientTcpSocket(
-      const SocketAddress& local_address, const SocketAddress& remote_address,
-      const rtc::ProxyInfo& proxy_info,
-      const std::string& user_agent, int opts) {
+  AsyncPacketSocket* CreateClientTcpSocket(const SocketAddress& local_address,
+                                           const SocketAddress& remote_address,
+                                           const rtc::ProxyInfo& proxy_info,
+                                           const std::string& user_agent,
+                                           int opts) override {
     EXPECT_TRUE(next_client_tcp_socket_ != NULL);
     AsyncPacketSocket* result = next_client_tcp_socket_;
     next_client_tcp_socket_ = NULL;
diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc
index 4c40b3da..1a07f8fe 100644
--- a/p2p/base/relayport.cc
+++ b/p2p/base/relayport.cc
@@ -172,11 +172,14 @@ class AllocateRequest : public StunRequest {
   uint32 start_time_;
 };
 
-RelayPort::RelayPort(
-    rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-    rtc::Network* network, const rtc::IPAddress& ip,
-    int min_port, int max_port, const std::string& username,
-    const std::string& password)
+RelayPort::RelayPort(rtc::Thread* thread,
+                     rtc::PacketSocketFactory* factory,
+                     rtc::Network* network,
+                     const rtc::IPAddress& ip,
+                     uint16 min_port,
+                     uint16 max_port,
+                     const std::string& username,
+                     const std::string& password)
     : Port(thread, RELAY_PORT_TYPE, factory, network, ip, min_port, max_port,
            username, password),
       ready_(false),
diff --git a/p2p/base/relayport.h b/p2p/base/relayport.h
index 3d9538da..62971426 100644
--- a/p2p/base/relayport.h
+++ b/p2p/base/relayport.h
@@ -36,9 +36,13 @@ class RelayPort : public Port {
 
   // RelayPort doesn't yet do anything fancy in the ctor.
   static RelayPort* Create(
-      rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-      rtc::Network* network, const rtc::IPAddress& ip,
-      int min_port, int max_port, const std::string& username,
+      rtc::Thread* thread,
+      rtc::PacketSocketFactory* factory,
+      rtc::Network* network,
+      const rtc::IPAddress& ip,
+      uint16 min_port,
+      uint16 max_port,
+      const std::string& username,
       const std::string& password) {
     return new RelayPort(thread, factory, network, ip, min_port, max_port,
                          username, password);
@@ -66,9 +70,13 @@ class RelayPort : public Port {
   sigslot::signal1<const ProtocolAddress*> SignalSoftTimeout;
 
  protected:
-  RelayPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-            rtc::Network*, const rtc::IPAddress& ip,
-            int min_port, int max_port, const std::string& username,
+  RelayPort(rtc::Thread* thread,
+            rtc::PacketSocketFactory* factory,
+            rtc::Network*,
+            const rtc::IPAddress& ip,
+            uint16 min_port,
+            uint16 max_port,
+            const std::string& username,
             const std::string& password);
   bool Init();
 
diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc
index ec6232a6..5ef9e9ea 100644
--- a/p2p/base/stunport.cc
+++ b/p2p/base/stunport.cc
@@ -162,7 +162,8 @@ UDPPort::UDPPort(rtc::Thread* thread,
                  rtc::PacketSocketFactory* factory,
                  rtc::Network* network,
                  rtc::AsyncPacketSocket* socket,
-                 const std::string& username, const std::string& password)
+                 const std::string& username,
+                 const std::string& password)
     : Port(thread, factory, network, socket->GetLocalAddress().ipaddr(),
            username, password),
       requests_(thread),
@@ -175,8 +176,11 @@ UDPPort::UDPPort(rtc::Thread* thread,
 UDPPort::UDPPort(rtc::Thread* thread,
                  rtc::PacketSocketFactory* factory,
                  rtc::Network* network,
-                 const rtc::IPAddress& ip, int min_port, int max_port,
-                 const std::string& username, const std::string& password)
+                 const rtc::IPAddress& ip,
+                 uint16 min_port,
+                 uint16 max_port,
+                 const std::string& username,
+                 const std::string& password)
     : Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
            username, password),
       requests_(thread),
diff --git a/p2p/base/stunport.h b/p2p/base/stunport.h
index eda7bb90..9ca60462 100644
--- a/p2p/base/stunport.h
+++ b/p2p/base/stunport.h
@@ -34,8 +34,8 @@ class UDPPort : public Port {
                          rtc::AsyncPacketSocket* socket,
                          const std::string& username,
                          const std::string& password) {
-    UDPPort* port = new UDPPort(thread, factory, network, socket,
-                                username, password);
+    UDPPort* port =
+        new UDPPort(thread, factory, network, socket, username, password);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -47,12 +47,12 @@ class UDPPort : public Port {
                          rtc::PacketSocketFactory* factory,
                          rtc::Network* network,
                          const rtc::IPAddress& ip,
-                         int min_port, int max_port,
+                         uint16 min_port,
+                         uint16 max_port,
                          const std::string& username,
                          const std::string& password) {
-    UDPPort* port = new UDPPort(thread, factory, network,
-                                ip, min_port, max_port,
-                                username, password);
+    UDPPort* port = new UDPPort(thread, factory, network, ip, min_port,
+                                max_port, username, password);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -98,14 +98,21 @@ class UDPPort : public Port {
   }
 
  protected:
-  UDPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-          rtc::Network* network, const rtc::IPAddress& ip,
-          int min_port, int max_port,
-          const std::string& username, const std::string& password);
-
-  UDPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-          rtc::Network* network, rtc::AsyncPacketSocket* socket,
-          const std::string& username, const std::string& password);
+  UDPPort(rtc::Thread* thread,
+          rtc::PacketSocketFactory* factory,
+          rtc::Network* network,
+          const rtc::IPAddress& ip,
+          uint16 min_port,
+          uint16 max_port,
+          const std::string& username,
+          const std::string& password);
+
+  UDPPort(rtc::Thread* thread,
+          rtc::PacketSocketFactory* factory,
+          rtc::Network* network,
+          rtc::AsyncPacketSocket* socket,
+          const std::string& username,
+          const std::string& password);
 
   bool Init();
 
@@ -194,18 +201,16 @@ class UDPPort : public Port {
 
 class StunPort : public UDPPort {
  public:
-  static StunPort* Create(
-      rtc::Thread* thread,
-      rtc::PacketSocketFactory* factory,
-      rtc::Network* network,
-      const rtc::IPAddress& ip,
-      int min_port, int max_port,
-      const std::string& username,
-      const std::string& password,
-      const ServerAddresses& servers) {
-    StunPort* port = new StunPort(thread, factory, network,
-                                  ip, min_port, max_port,
-                                  username, password, servers);
+  static StunPort* Create(rtc::Thread* thread,
+                          rtc::PacketSocketFactory* factory,
+                          rtc::Network* network,
+                          const rtc::IPAddress& ip,
+                          uint16 min_port, uint16 max_port,
+                          const std::string& username,
+                          const std::string& password,
+                          const ServerAddresses& servers) {
+    StunPort* port = new StunPort(thread, factory, network, ip, min_port,
+                                  max_port, username, password, servers);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -220,10 +225,14 @@ class StunPort : public UDPPort {
   }
 
  protected:
-  StunPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-           rtc::Network* network, const rtc::IPAddress& ip,
-           int min_port, int max_port,
-           const std::string& username, const std::string& password,
+  StunPort(rtc::Thread* thread,
+           rtc::PacketSocketFactory* factory,
+           rtc::Network* network,
+           const rtc::IPAddress& ip,
+           uint16 min_port,
+           uint16 max_port,
+           const std::string& username,
+           const std::string& password,
            const ServerAddresses& servers)
      : UDPPort(thread, factory, network, ip, min_port, max_port, username,
                password) {
diff --git a/p2p/base/tcpport.cc b/p2p/base/tcpport.cc
index be3068be..b37f4d3f 100644
--- a/p2p/base/tcpport.cc
+++ b/p2p/base/tcpport.cc
@@ -18,9 +18,13 @@ namespace cricket {
 
 TCPPort::TCPPort(rtc::Thread* thread,
                  rtc::PacketSocketFactory* factory,
-                 rtc::Network* network, const rtc::IPAddress& ip,
-                 int min_port, int max_port, const std::string& username,
-                 const std::string& password, bool allow_listen)
+                 rtc::Network* network,
+                 const rtc::IPAddress& ip,
+                 uint16 min_port,
+                 uint16 max_port,
+                 const std::string& username,
+                 const std::string& password,
+                 bool allow_listen)
     : Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
            username, password),
       incoming_only_(false),
diff --git a/p2p/base/tcpport.h b/p2p/base/tcpport.h
index 43e49366..b3655a80 100644
--- a/p2p/base/tcpport.h
+++ b/p2p/base/tcpport.h
@@ -32,13 +32,13 @@ class TCPPort : public Port {
                          rtc::PacketSocketFactory* factory,
                          rtc::Network* network,
                          const rtc::IPAddress& ip,
-                         int min_port, int max_port,
+                         uint16 min_port,
+                         uint16 max_port,
                          const std::string& username,
                          const std::string& password,
                          bool allow_listen) {
-    TCPPort* port = new TCPPort(thread, factory, network,
-                                ip, min_port, max_port,
-                                username, password, allow_listen);
+    TCPPort* port = new TCPPort(thread, factory, network, ip, min_port,
+                                max_port, username, password, allow_listen);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -57,10 +57,15 @@ class TCPPort : public Port {
   virtual int GetError();
 
  protected:
-  TCPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
-          rtc::Network* network, const rtc::IPAddress& ip,
-          int min_port, int max_port, const std::string& username,
-          const std::string& password, bool allow_listen);
+  TCPPort(rtc::Thread* thread,
+          rtc::PacketSocketFactory* factory,
+          rtc::Network* network,
+          const rtc::IPAddress& ip,
+          uint16 min_port,
+          uint16 max_port,
+          const std::string& username,
+          const std::string& password,
+          bool allow_listen);
   bool Init();
 
   // Handles sending using the local TCP socket.
diff --git a/p2p/base/turnport.cc b/p2p/base/turnport.cc
index e7626fe0..fbdcfeb6 100644
--- a/p2p/base/turnport.cc
+++ b/p2p/base/turnport.cc
@@ -184,7 +184,8 @@ TurnPort::TurnPort(rtc::Thread* thread,
                    rtc::PacketSocketFactory* factory,
                    rtc::Network* network,
                    const rtc::IPAddress& ip,
-                   int min_port, int max_port,
+                   uint16 min_port,
+                   uint16 max_port,
                    const std::string& username,
                    const std::string& password,
                    const ProtocolAddress& server_address,
diff --git a/p2p/base/turnport.h b/p2p/base/turnport.h
index 17fad176..4ed77a0c 100644
--- a/p2p/base/turnport.h
+++ b/p2p/base/turnport.h
@@ -42,16 +42,16 @@ class TurnPort : public Port {
                           const ProtocolAddress& server_address,
                           const RelayCredentials& credentials,
                           int server_priority) {
-    return new TurnPort(thread, factory, network, socket,
-                        username, password, server_address,
-                        credentials, server_priority);
+    return new TurnPort(thread, factory, network, socket, username, password,
+                        server_address, credentials, server_priority);
   }
 
   static TurnPort* Create(rtc::Thread* thread,
                           rtc::PacketSocketFactory* factory,
                           rtc::Network* network,
                           const rtc::IPAddress& ip,
-                          int min_port, int max_port,
+                          uint16 min_port,
+                          uint16 max_port,
                           const std::string& username,  // ice username.
                           const std::string& password,  // ice password.
                           const ProtocolAddress& server_address,
@@ -135,7 +135,8 @@ class TurnPort : public Port {
            rtc::PacketSocketFactory* factory,
            rtc::Network* network,
            const rtc::IPAddress& ip,
-           int min_port, int max_port,
+           uint16 min_port,
+           uint16 max_port,
            const std::string& username,
            const std::string& password,
            const ProtocolAddress& server_address,
diff --git a/system_wrappers/source/trace_impl.cc b/system_wrappers/source/trace_impl.cc
index 13c63acf..d05b928a 100644
--- a/system_wrappers/source/trace_impl.cc
+++ b/system_wrappers/source/trace_impl.cc
@@ -553,15 +553,6 @@ void TraceImpl::WriteToFile() {
           trace_file_.Write(message, length);
           row_count_text_++;
         }
-        length = AddBuildInfo(message);
-        if (length != -1) {
-          message[length + 1] = 0;
-          message[length] = '\n';
-          message[length - 1] = '\n';
-          trace_file_.Write(message, length + 1);
-          row_count_text_++;
-          row_count_text_++;
-        }
       }
       uint16_t length = length_[local_queue_active][idx];
       message_queue_[local_queue_active][idx][length] = 0;
diff --git a/system_wrappers/source/trace_impl.h b/system_wrappers/source/trace_impl.h
index 5548e984..5fa6ce38 100644
--- a/system_wrappers/source/trace_impl.h
+++ b/system_wrappers/source/trace_impl.h
@@ -72,7 +72,6 @@ class TraceImpl : public Trace {
   virtual int32_t AddTime(char* trace_message,
                           const TraceLevel level) const = 0;
 
-  virtual int32_t AddBuildInfo(char* trace_message) const = 0;
   virtual int32_t AddDateTimeInfo(char* trace_message) const = 0;
 
   static bool Run(void* obj);
diff --git a/system_wrappers/source/trace_posix.cc b/system_wrappers/source/trace_posix.cc
index d14704b6..1e0e1c81 100644
--- a/system_wrappers/source/trace_posix.cc
+++ b/system_wrappers/source/trace_posix.cc
@@ -17,20 +17,6 @@
 #include <sys/time.h>
 #include <time.h>
 
-#if defined(_DEBUG)
-#define BUILDMODE "d"
-#elif defined(DEBUG)
-#define BUILDMODE "d"
-#elif defined(NDEBUG)
-#define BUILDMODE "r"
-#else
-#define BUILDMODE "?"
-#endif
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-// example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
 namespace webrtc {
 
 TracePosix::TracePosix()
@@ -86,12 +72,6 @@ int32_t TracePosix::AddTime(char* trace_message, const TraceLevel level) const {
   return 22;
 }
 
-int32_t TracePosix::AddBuildInfo(char* trace_message) const {
-  sprintf(trace_message, "Build info: %s", BUILDINFO);
-  // Include NULL termination (hence + 1).
-  return strlen(trace_message) + 1;
-}
-
 int32_t TracePosix::AddDateTimeInfo(char* trace_message) const {
   time_t t;
   time(&t);
diff --git a/system_wrappers/source/trace_posix.h b/system_wrappers/source/trace_posix.h
index 2056c704..2f0abc6e 100644
--- a/system_wrappers/source/trace_posix.h
+++ b/system_wrappers/source/trace_posix.h
@@ -26,7 +26,6 @@ class TracePosix : public TraceImpl {
   virtual int32_t AddTime(char* trace_message, const TraceLevel level) const
       OVERRIDE;
 
-  virtual int32_t AddBuildInfo(char* trace_message) const OVERRIDE;
   virtual int32_t AddDateTimeInfo(char* trace_message) const OVERRIDE;
 
  private:
diff --git a/system_wrappers/source/trace_win.cc b/system_wrappers/source/trace_win.cc
index f1a03a54..3752659c 100644
--- a/system_wrappers/source/trace_win.cc
+++ b/system_wrappers/source/trace_win.cc
@@ -15,20 +15,6 @@
 
 #include "Mmsystem.h"
 
-#if defined(_DEBUG)
-#define BUILDMODE "d"
-#elif defined(DEBUG)
-#define BUILDMODE "d"
-#elif defined(NDEBUG)
-#define BUILDMODE "r"
-#else
-#define BUILDMODE "?"
-#endif
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-// Example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
 namespace webrtc {
 TraceWindows::TraceWindows()
     : prev_api_tick_count_(0),
@@ -84,13 +70,6 @@ int32_t TraceWindows::AddTime(char* trace_message,
   return 22;
 }
 
-int32_t TraceWindows::AddBuildInfo(char* trace_message) const {
-  // write data and time to text file
-  sprintf(trace_message, "Build info: %s", BUILDINFO);
-  // Include NULL termination (hence + 1).
-  return static_cast<int32_t>(strlen(trace_message) + 1);
-}
-
 int32_t TraceWindows::AddDateTimeInfo(char* trace_message) const {
   prev_api_tick_count_ = timeGetTime();
   prev_tick_count_ = prev_api_tick_count_;
diff --git a/system_wrappers/source/trace_win.h b/system_wrappers/source/trace_win.h
index 1a871074..1311b23c 100644
--- a/system_wrappers/source/trace_win.h
+++ b/system_wrappers/source/trace_win.h
@@ -25,7 +25,6 @@ class TraceWindows : public TraceImpl {
 
   virtual int32_t AddTime(char* trace_message, const TraceLevel level) const;
 
-  virtual int32_t AddBuildInfo(char* trace_message) const;
   virtual int32_t AddDateTimeInfo(char* trace_message) const;
  private:
   volatile mutable uint32_t prev_api_tick_count_;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index db17c31e..870404e7 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -67,10 +67,7 @@ source_set("test_support") {
   ]
 
   if (is_android) {
-    sources += [ "testsupport/android/root_path_android_chromium.cc" ]
     deps += [ "//base:base" ]
-  } else {
-    sources += [ "testsupport/android/root_path_android.cc" ]
   }
 
   configs += [ "..:common_config" ]
diff --git a/test/encoder_settings.cc b/test/encoder_settings.cc
index db064bb8..bae13505 100644
--- a/test/encoder_settings.cc
+++ b/test/encoder_settings.cc
@@ -60,6 +60,8 @@ VideoReceiveStream::Decoder CreateMatchingDecoder(
   decoder.payload_name = encoder_settings.payload_name;
   if (encoder_settings.payload_name == "VP8") {
     decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp8);
+  } else if (encoder_settings.payload_name == "VP9") {
+    decoder.decoder = VideoDecoder::Create(VideoDecoder::kVp9);
   } else {
     decoder.decoder = new FakeDecoder();
   }
diff --git a/test/fake_encoder.cc b/test/fake_encoder.cc
index 9551c820..0573c8af 100644
--- a/test/fake_encoder.cc
+++ b/test/fake_encoder.cc
@@ -51,7 +51,8 @@ int32_t FakeEncoder::Encode(
   assert(config_.maxFramerate > 0);
   int time_since_last_encode_ms = 1000 / config_.maxFramerate;
   int64_t time_now_ms = clock_->TimeInMilliseconds();
-  if (last_encode_time_ms_ > 0) {
+  const bool first_encode = last_encode_time_ms_ == 0;
+  if (!first_encode) {
     // For all frames but the first we can estimate the display time by looking
     // at the display time of the previous frame.
     time_since_last_encode_ms = time_now_ms - last_encode_time_ms_;
@@ -80,6 +81,12 @@ int32_t FakeEncoder::Encode(
     int stream_bits = (bits_available > max_stream_bits) ? max_stream_bits :
         bits_available;
     int stream_bytes = (stream_bits + 7) / 8;
+    if (first_encode) {
+      // The first frame is a key frame and should be larger.
+      // TODO(holmer): The FakeEncoder should store the bits_available between
+      // encodes so that it can compensate for oversized frames.
+      stream_bytes *= 10;
+    }
     if (static_cast<size_t>(stream_bytes) > sizeof(encoded_buffer_))
       stream_bytes = sizeof(encoded_buffer_);
 
@@ -96,7 +103,6 @@ int32_t FakeEncoder::Encode(
     assert(callback_ != NULL);
     if (callback_->Encoded(encoded, &specifics, NULL) != 0)
       return -1;
-
     bits_available -= encoded._length * 8;
   }
   return 0;
diff --git a/test/test.gyp b/test/test.gyp
index 5c959b9e..7aba2f57 100644
--- a/test/test.gyp
+++ b/test/test.gyp
@@ -98,8 +98,6 @@
         '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'testsupport/android/root_path_android.cc',
-        'testsupport/android/root_path_android_chromium.cc',
         'testsupport/fileutils.cc',
         'testsupport/fileutils.h',
         'testsupport/frame_reader.cc',
@@ -117,20 +115,6 @@
         'testsupport/trace_to_stderr.cc',
         'testsupport/trace_to_stderr.h',
       ],
-      'conditions': [
-        ['OS=="android"', {
-          'dependencies': [
-            '<(DEPTH)/base/base.gyp:base',
-          ],
-          'sources!': [
-            'testsupport/android/root_path_android.cc',
-          ],
-        }, {
-          'sources!': [
-            'testsupport/android/root_path_android_chromium.cc',
-          ],
-        }],
-      ],
     },
     {
       # Depend on this target when you want to have test_support but also the
diff --git a/test/testsupport/android/root_path_android.cc b/test/testsupport/android/root_path_android.cc
deleted file mode 100644
index 6eca5f57..00000000
--- a/test/testsupport/android/root_path_android.cc
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- *  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.
- */
-
-#include <string>
-
-namespace webrtc {
-namespace test {
-
-static const char* kRootDirName = "/sdcard/";
-std::string ProjectRootPathAndroid() {
-  return kRootDirName;
-}
-
-std::string OutputPathAndroid() {
-  return kRootDirName;
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/test/testsupport/android/root_path_android_chromium.cc b/test/testsupport/android/root_path_android_chromium.cc
deleted file mode 100644
index f8e65f4f..00000000
--- a/test/testsupport/android/root_path_android_chromium.cc
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  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.
- */
-
-#include "base/android/path_utils.h"
-#include "base/files/file_path.h"
-
-namespace webrtc {
-namespace test {
-
-std::string OutputPathImpl();
-
-// This file is only compiled when running WebRTC tests in a Chromium workspace.
-// The Android testing framework will push files relative to the root path of
-// the Chromium workspace. The root path for webrtc is one directory up from
-// trunk/webrtc (in standalone) or src/third_party/webrtc (in Chromium).
-std::string ProjectRootPathAndroid() {
-  base::FilePath root_path;
-  base::android::GetExternalStorageDirectory(&root_path);
-  return root_path.value() + "/";
-}
-
-std::string OutputPathAndroid() {
-  return OutputPathImpl();
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/test/testsupport/fileutils.cc b/test/testsupport/fileutils.cc
index a3e66201..36ca391c 100644
--- a/test/testsupport/fileutils.cc
+++ b/test/testsupport/fileutils.cc
@@ -50,15 +50,15 @@ const char* kPathDelimiter = "/";
 #endif
 
 #ifdef WEBRTC_ANDROID
-const char* kResourcesDirName = "resources";
+const char* kRootDirName = "/sdcard/";
 #else
 // The file we're looking for to identify the project root dir.
 const char* kProjectRootFileName = "DEPS";
-const char* kResourcesDirName = "resources";
+const char* kOutputDirName = "out";
+const char* kFallbackPath = "./";
 #endif
+const char* kResourcesDirName = "resources";
 
-const char* kFallbackPath = "./";
-const char* kOutputDirName = "out";
 char relative_dir_path[FILENAME_MAX];
 bool relative_dir_path_set = false;
 
@@ -66,9 +66,6 @@ bool relative_dir_path_set = false;
 
 const char* kCannotFindProjectRootDir = "ERROR_CANNOT_FIND_PROJECT_ROOT_DIR";
 
-std::string OutputPathAndroid();
-std::string ProjectRootPathAndroid();
-
 void SetExecutablePath(const std::string& path) {
   std::string working_dir = WorkingDir();
   std::string temp_path = path;
@@ -95,30 +92,18 @@ bool FileExists(std::string& file_name) {
   return stat(file_name.c_str(), &file_info) == 0;
 }
 
-std::string OutputPathImpl() {
-  std::string path = ProjectRootPath();
-  if (path == kCannotFindProjectRootDir) {
-    return kFallbackPath;
-  }
-  path += kOutputDirName;
-  if (!CreateDir(path)) {
-    return kFallbackPath;
-  }
-  return path + kPathDelimiter;
-}
-
 #ifdef WEBRTC_ANDROID
 
 std::string ProjectRootPath() {
-  return ProjectRootPathAndroid();
+  return kRootDirName;
 }
 
 std::string OutputPath() {
-  return OutputPathAndroid();
+  return kRootDirName;
 }
 
 std::string WorkingDir() {
-  return ProjectRootPath();
+  return kRootDirName;
 }
 
 #else // WEBRTC_ANDROID
@@ -148,7 +133,15 @@ std::string ProjectRootPath() {
 }
 
 std::string OutputPath() {
-  return OutputPathImpl();
+  std::string path = ProjectRootPath();
+  if (path == kCannotFindProjectRootDir) {
+    return kFallbackPath;
+  }
+  path += kOutputDirName;
+  if (!CreateDir(path)) {
+    return kFallbackPath;
+  }
+  return path + kPathDelimiter;
 }
 
 std::string WorkingDir() {
diff --git a/test/testsupport/fileutils_unittest.cc b/test/testsupport/fileutils_unittest.cc
index 114f1083..dff7f224 100644
--- a/test/testsupport/fileutils_unittest.cc
+++ b/test/testsupport/fileutils_unittest.cc
@@ -66,7 +66,7 @@ TEST_F(FileUtilsTest, ProjectRootPath) {
 }
 
 // Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) {
+TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) {
   std::string path = webrtc::test::OutputPath();
   std::string expected_end = "out";
   expected_end = kPathDelimiter + expected_end + kPathDelimiter;
@@ -80,10 +80,11 @@ TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromRootWorkingDir)) {
   ASSERT_EQ("./", webrtc::test::OutputPath());
 }
 
-TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(TempFilename)) {
+TEST_F(FileUtilsTest, TempFilename) {
   std::string temp_filename = webrtc::test::TempFilename(
       webrtc::test::OutputPath(), "TempFilenameTest");
-  ASSERT_TRUE(webrtc::test::FileExists(temp_filename));
+  ASSERT_TRUE(webrtc::test::FileExists(temp_filename))
+      << "Couldn't find file: " << temp_filename;
   remove(temp_filename.c_str());
 }
 
diff --git a/video/call.cc b/video/call.cc
index fd41d75f..2b4f76f4 100644
--- a/video/call.cc
+++ b/video/call.cc
@@ -58,6 +58,8 @@ VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
   switch (codec_type) {
     case kVp8:
       return VP8Decoder::Create();
+    case kVp9:
+      return VP9Decoder::Create();
   }
   assert(false);
   return NULL;
@@ -112,8 +114,7 @@ class Call : public webrtc::Call, public PacketReceiver {
   virtual void DestroyVideoReceiveStream(
       webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
 
-  virtual uint32_t SendBitrateEstimate() OVERRIDE;
-  virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
+  virtual Stats GetStats() const OVERRIDE;
 
   virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
                                        size_t length) OVERRIDE;
@@ -319,14 +320,26 @@ void Call::DestroyVideoReceiveStream(
   delete receive_stream_impl;
 }
 
-uint32_t Call::SendBitrateEstimate() {
-  // TODO(pbos): Return send-bitrate estimate
-  return 0;
-}
-
-uint32_t Call::ReceiveBitrateEstimate() {
-  // TODO(pbos): Return receive-bitrate estimate
-  return 0;
+Call::Stats Call::GetStats() const {
+  Stats stats;
+  // Ignoring return values.
+  uint32_t send_bandwidth = 0;
+  rtp_rtcp_->GetEstimatedSendBandwidth(base_channel_id_, &send_bandwidth);
+  stats.send_bandwidth_bps = send_bandwidth;
+  uint32_t recv_bandwidth = 0;
+  rtp_rtcp_->GetEstimatedReceiveBandwidth(base_channel_id_, &recv_bandwidth);
+  stats.recv_bandwidth_bps = recv_bandwidth;
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    for (std::map<uint32_t, VideoSendStream*>::const_iterator it =
+             send_ssrcs_.begin();
+         it != send_ssrcs_.end();
+         ++it) {
+      stats.pacer_delay_ms =
+          std::max(it->second->GetPacerQueuingDelayMs(), stats.pacer_delay_ms);
+    }
+  }
+  return stats;
 }
 
 void Call::SignalNetworkState(NetworkState state) {
diff --git a/video/call_perf_tests.cc b/video/call_perf_tests.cc
index 9776fb7e..9194b088 100644
--- a/video/call_perf_tests.cc
+++ b/video/call_perf_tests.cc
@@ -502,7 +502,8 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
       VideoSendStream::Stats stats = send_stream_->GetStats();
       if (stats.substreams.size() > 0) {
         assert(stats.substreams.size() == 1);
-        int bitrate_kbps = stats.substreams.begin()->second.bitrate_bps / 1000;
+        int bitrate_kbps =
+            stats.substreams.begin()->second.total_bitrate_bps / 1000;
         if (bitrate_kbps > 0) {
           test::PrintResult(
               "bitrate_stats_",
diff --git a/video/end_to_end_tests.cc b/video/end_to_end_tests.cc
index 96249c3c..2b3c00f5 100644
--- a/video/end_to_end_tests.cc
+++ b/video/end_to_end_tests.cc
@@ -225,8 +225,7 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) {
   DestroyStreams();
 }
 
-// TODO(marpan): Re-enable this test on the next libvpx roll.
-TEST_F(EndToEndTest, DISABLED_SendsAndReceivesVP9) {
+TEST_F(EndToEndTest, SendsAndReceivesVP9) {
   class VP9Observer : public test::EndToEndTest, public VideoRenderer {
    public:
     VP9Observer()
@@ -1202,6 +1201,53 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, VerifyBandwidthStats) {
+  class RtcpObserver : public test::EndToEndTest, public PacketReceiver {
+   public:
+    RtcpObserver()
+        : EndToEndTest(kDefaultTimeoutMs),
+          sender_call_(NULL),
+          receiver_call_(NULL),
+          has_seen_pacer_delay_(false) {}
+
+    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                         size_t length) OVERRIDE {
+      Call::Stats sender_stats = sender_call_->GetStats();
+      Call::Stats receiver_stats = receiver_call_->GetStats();
+      if (!has_seen_pacer_delay_)
+        has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
+      if (sender_stats.send_bandwidth_bps > 0 &&
+          receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_)
+        observation_complete_->Set();
+      return receiver_call_->Receiver()->DeliverPacket(packet, length);
+    }
+
+    virtual void OnCallsCreated(Call* sender_call,
+                                Call* receiver_call) OVERRIDE {
+      sender_call_ = sender_call;
+      receiver_call_ = receiver_call;
+    }
+
+    virtual void PerformTest() OVERRIDE {
+      EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
+                                           "non-zero bandwidth stats.";
+    }
+
+    virtual void SetReceivers(
+        PacketReceiver* send_transport_receiver,
+        PacketReceiver* receive_transport_receiver) OVERRIDE {
+      test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
+    }
+
+   private:
+    Call* sender_call_;
+    Call* receiver_call_;
+    bool has_seen_pacer_delay_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
 void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
   static const int kNumRtcpReportPacketsToObserve = 5;
   class RtcpXrObserver : public test::EndToEndTest {
@@ -1434,7 +1480,7 @@ TEST_F(EndToEndTest, GetStats) {
       // Make sure all fields have been populated.
 
       receive_stats_filled_["IncomingRate"] |=
-          stats.network_frame_rate != 0 || stats.bitrate_bps != 0;
+          stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
 
       receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
 
@@ -1465,7 +1511,7 @@ TEST_F(EndToEndTest, GetStats) {
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
-      for (std::map<uint32_t, StreamStats>::const_iterator it =
+      for (std::map<uint32_t, SsrcStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end();
            ++it) {
@@ -1475,7 +1521,7 @@ TEST_F(EndToEndTest, GetStats) {
         send_stats_filled_[CompoundKey("IncomingRate", it->first)] |=
             stats.input_frame_rate != 0;
 
-        const StreamStats& stream_stats = it->second;
+        const SsrcStats& stream_stats = it->second;
 
         send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
             stream_stats.rtcp_stats.cumulative_lost != 0 ||
@@ -1490,7 +1536,7 @@ TEST_F(EndToEndTest, GetStats) {
 
         send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
                                        it->first)] |=
-            stream_stats.bitrate_bps != 0;
+            stream_stats.total_bitrate_bps != 0;
 
         send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
             stream_stats.delta_frames != 0 || stream_stats.key_frames != 0;
diff --git a/video/loopback.cc b/video/loopback.cc
index 4b49c31e..8013833d 100644
--- a/video/loopback.cc
+++ b/video/loopback.cc
@@ -142,6 +142,8 @@ void Loopback() {
   scoped_ptr<VideoEncoder> encoder;
   if (flags::Codec() == "VP8") {
     encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
+  } else if (flags::Codec() == "VP9") {
+    encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
   } else {
     // Codec not supported.
     assert(false && "Codec not supported!");
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index 38262834..ca0bcdfb 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -58,10 +58,10 @@ std::string ReceiveStatisticsProxy::GetCName() const {
 
 void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
                                           const unsigned int framerate,
-                                          const unsigned int bitrate) {
+                                          const unsigned int bitrate_bps) {
   CriticalSectionScoped lock(crit_.get());
   stats_.network_frame_rate = framerate;
-  stats_.bitrate_bps = bitrate;
+  stats_.total_bitrate_bps = bitrate_bps;
 }
 
 void ReceiveStatisticsProxy::StatisticsUpdated(
diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h
index b5fbf860..a1b6735b 100644
--- a/video/receive_statistics_proxy.h
+++ b/video/receive_statistics_proxy.h
@@ -52,7 +52,7 @@ class ReceiveStatisticsProxy : public ViEDecoderObserver,
                                     const VideoCodec& video_codec) OVERRIDE {}
   virtual void IncomingRate(const int video_channel,
                             const unsigned int framerate,
-                            const unsigned int bitrate) OVERRIDE;
+                            const unsigned int bitrate_bps) OVERRIDE;
   virtual void DecoderTiming(int decode_ms,
                              int max_decode_ms,
                              int current_delay_ms,
diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc
index 1b6081d2..f2df0ed2 100644
--- a/video/send_statistics_proxy.cc
+++ b/video/send_statistics_proxy.cc
@@ -29,6 +29,7 @@ void SendStatisticsProxy::OutgoingRate(const int video_channel,
                                        const unsigned int bitrate) {
   CriticalSectionScoped lock(crit_.get());
   stats_.encode_frame_rate = framerate;
+  stats_.media_bitrate_bps = bitrate;
 }
 
 void SendStatisticsProxy::SuspendChange(int video_channel, bool is_suspended) {
@@ -47,8 +48,8 @@ VideoSendStream::Stats SendStatisticsProxy::GetStats() const {
   return stats_;
 }
 
-StreamStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
-  std::map<uint32_t, StreamStats>::iterator it = stats_.substreams.find(ssrc);
+SsrcStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
+  std::map<uint32_t, SsrcStats>::iterator it = stats_.substreams.find(ssrc);
   if (it != stats_.substreams.end())
     return &it->second;
 
@@ -66,7 +67,7 @@ StreamStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
 void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
                                             uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  StreamStats* stats = GetStatsEntry(ssrc);
+  SsrcStats* stats = GetStatsEntry(ssrc);
   if (stats == NULL)
     return;
 
@@ -77,28 +78,30 @@ void SendStatisticsProxy::DataCountersUpdated(
     const StreamDataCounters& counters,
     uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  StreamStats* stats = GetStatsEntry(ssrc);
+  SsrcStats* stats = GetStatsEntry(ssrc);
   if (stats == NULL)
     return;
 
   stats->rtp_stats = counters;
 }
 
-void SendStatisticsProxy::Notify(const BitrateStatistics& bitrate,
+void SendStatisticsProxy::Notify(const BitrateStatistics& total_stats,
+                                 const BitrateStatistics& retransmit_stats,
                                  uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  StreamStats* stats = GetStatsEntry(ssrc);
+  SsrcStats* stats = GetStatsEntry(ssrc);
   if (stats == NULL)
     return;
 
-  stats->bitrate_bps = bitrate.bitrate_bps;
+  stats->total_bitrate_bps = total_stats.bitrate_bps;
+  stats->retransmit_bitrate_bps = retransmit_stats.bitrate_bps;
 }
 
 void SendStatisticsProxy::FrameCountUpdated(FrameType frame_type,
                                             uint32_t frame_count,
                                             const unsigned int ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  StreamStats* stats = GetStatsEntry(ssrc);
+  SsrcStats* stats = GetStatsEntry(ssrc);
   if (stats == NULL)
     return;
 
@@ -120,7 +123,7 @@ void SendStatisticsProxy::SendSideDelayUpdated(int avg_delay_ms,
                                                int max_delay_ms,
                                                uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  StreamStats* stats = GetStatsEntry(ssrc);
+  SsrcStats* stats = GetStatsEntry(ssrc);
   if (stats == NULL)
     return;
   stats->avg_delay_ms = avg_delay_ms;
diff --git a/video/send_statistics_proxy.h b/video/send_statistics_proxy.h
index ef459dab..2f645b1e 100644
--- a/video/send_statistics_proxy.h
+++ b/video/send_statistics_proxy.h
@@ -46,7 +46,9 @@ class SendStatisticsProxy : public RtcpStatisticsCallback,
                                    uint32_t ssrc) OVERRIDE;
 
   // From BitrateStatisticsObserver.
-  virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) OVERRIDE;
+  virtual void Notify(const BitrateStatistics& total_stats,
+                      const BitrateStatistics& retransmit_stats,
+                      uint32_t ssrc) OVERRIDE;
 
   // From FrameCountObserver.
   virtual void FrameCountUpdated(FrameType frame_type,
@@ -75,7 +77,7 @@ class SendStatisticsProxy : public RtcpStatisticsCallback,
                                     uint32_t ssrc) OVERRIDE;
 
  private:
-  StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  SsrcStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   const VideoSendStream::Config config_;
   scoped_ptr<CriticalSectionWrapper> crit_;
diff --git a/video/send_statistics_proxy_unittest.cc b/video/send_statistics_proxy_unittest.cc
index f768452f..d7750f8a 100644
--- a/video/send_statistics_proxy_unittest.cc
+++ b/video/send_statistics_proxy_unittest.cc
@@ -44,22 +44,23 @@ class SendStatisticsProxyTest : public ::testing::Test {
   void ExpectEqual(VideoSendStream::Stats one, VideoSendStream::Stats other) {
     EXPECT_EQ(one.input_frame_rate, other.input_frame_rate);
     EXPECT_EQ(one.encode_frame_rate, other.encode_frame_rate);
+    EXPECT_EQ(one.media_bitrate_bps, other.media_bitrate_bps);
     EXPECT_EQ(one.suspended, other.suspended);
 
     EXPECT_EQ(one.substreams.size(), other.substreams.size());
-    for (std::map<uint32_t, StreamStats>::const_iterator it =
+    for (std::map<uint32_t, SsrcStats>::const_iterator it =
              one.substreams.begin();
          it != one.substreams.end();
          ++it) {
-      std::map<uint32_t, StreamStats>::const_iterator corresponding_it =
+      std::map<uint32_t, SsrcStats>::const_iterator corresponding_it =
           other.substreams.find(it->first);
       ASSERT_TRUE(corresponding_it != other.substreams.end());
-      const StreamStats& a = it->second;
-      const StreamStats& b = corresponding_it->second;
+      const SsrcStats& a = it->second;
+      const SsrcStats& b = corresponding_it->second;
 
       EXPECT_EQ(a.key_frames, b.key_frames);
       EXPECT_EQ(a.delta_frames, b.delta_frames);
-      EXPECT_EQ(a.bitrate_bps, b.bitrate_bps);
+      EXPECT_EQ(a.total_bitrate_bps, b.total_bitrate_bps);
       EXPECT_EQ(a.avg_delay_ms, b.avg_delay_ms);
       EXPECT_EQ(a.max_delay_ms, b.max_delay_ms);
 
@@ -84,7 +85,7 @@ class SendStatisticsProxyTest : public ::testing::Test {
   int avg_delay_ms_;
   int max_delay_ms_;
   VideoSendStream::Stats expected_;
-  typedef std::map<uint32_t, StreamStats>::const_iterator StreamIterator;
+  typedef std::map<uint32_t, SsrcStats>::const_iterator StreamIterator;
 };
 
 TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
@@ -93,7 +94,7 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
        it != config_.rtp.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    StreamStats& ssrc_stats = expected_.substreams[ssrc];
+    SsrcStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
     uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -107,7 +108,7 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
        it != config_.rtp.rtx.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    StreamStats& ssrc_stats = expected_.substreams[ssrc];
+    SsrcStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
     uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -121,17 +122,25 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
   ExpectEqual(expected_, stats);
 }
 
-TEST_F(SendStatisticsProxyTest, FrameRates) {
+TEST_F(SendStatisticsProxyTest, CaptureFramerate) {
   const int capture_fps = 31;
-  const int encode_fps = 29;
 
   ViECaptureObserver* capture_observer = statistics_proxy_.get();
   capture_observer->CapturedFrameRate(0, capture_fps);
-  ViEEncoderObserver* encoder_observer = statistics_proxy_.get();
-  encoder_observer->OutgoingRate(0, encode_fps, 0);
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_EQ(capture_fps, stats.input_frame_rate);
+}
+
+TEST_F(SendStatisticsProxyTest, EncodedBitrateAndFramerate) {
+  const int media_bitrate_bps = 500;
+  const int encode_fps = 29;
+
+  ViEEncoderObserver* encoder_observer = statistics_proxy_.get();
+  encoder_observer->OutgoingRate(0, encode_fps, media_bitrate_bps);
+
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(media_bitrate_bps, stats.media_bitrate_bps);
   EXPECT_EQ(encode_fps, stats.encode_frame_rate);
 }
 
@@ -156,8 +165,8 @@ TEST_F(SendStatisticsProxyTest, FrameCounts) {
        ++it) {
     const uint32_t ssrc = *it;
     // Add statistics with some arbitrary, but unique, numbers.
-    StreamStats& stats = expected_.substreams[ssrc];
-    uint32_t offset = ssrc * sizeof(StreamStats);
+    SsrcStats& stats = expected_.substreams[ssrc];
+    uint32_t offset = ssrc * sizeof(SsrcStats);
     stats.key_frames = offset;
     stats.delta_frames = offset + 1;
     observer->FrameCountUpdated(kVideoFrameKey, stats.key_frames, ssrc);
@@ -168,8 +177,8 @@ TEST_F(SendStatisticsProxyTest, FrameCounts) {
        ++it) {
     const uint32_t ssrc = *it;
     // Add statistics with some arbitrary, but unique, numbers.
-    StreamStats& stats = expected_.substreams[ssrc];
-    uint32_t offset = ssrc * sizeof(StreamStats);
+    SsrcStats& stats = expected_.substreams[ssrc];
+    uint32_t offset = ssrc * sizeof(SsrcStats);
     stats.key_frames = offset;
     stats.delta_frames = offset + 1;
     observer->FrameCountUpdated(kVideoFrameKey, stats.key_frames, ssrc);
@@ -223,21 +232,27 @@ TEST_F(SendStatisticsProxyTest, Bitrate) {
        it != config_.rtp.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    BitrateStatistics bitrate;
+    BitrateStatistics total;
+    BitrateStatistics retransmit;
     // Use ssrc as bitrate_bps to get a unique value for each stream.
-    bitrate.bitrate_bps = ssrc;
-    observer->Notify(bitrate, ssrc);
-    expected_.substreams[ssrc].bitrate_bps = ssrc;
+    total.bitrate_bps = ssrc;
+    retransmit.bitrate_bps = ssrc + 1;
+    observer->Notify(total, retransmit, ssrc);
+    expected_.substreams[ssrc].total_bitrate_bps = total.bitrate_bps;
+    expected_.substreams[ssrc].retransmit_bitrate_bps = retransmit.bitrate_bps;
   }
   for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
        it != config_.rtp.rtx.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    BitrateStatistics bitrate;
+    BitrateStatistics total;
+    BitrateStatistics retransmit;
     // Use ssrc as bitrate_bps to get a unique value for each stream.
-    bitrate.bitrate_bps = ssrc;
-    observer->Notify(bitrate, ssrc);
-    expected_.substreams[ssrc].bitrate_bps = ssrc;
+    total.bitrate_bps = ssrc;
+    retransmit.bitrate_bps = ssrc + 1;
+    observer->Notify(total, retransmit, ssrc);
+    expected_.substreams[ssrc].total_bitrate_bps = total.bitrate_bps;
+    expected_.substreams[ssrc].retransmit_bitrate_bps = retransmit.bitrate_bps;
   }
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
@@ -292,9 +307,10 @@ TEST_F(SendStatisticsProxyTest, NoSubstreams) {
   rtp_callback->DataCountersUpdated(rtp_stats, exluded_ssrc);
 
   // From BitrateStatisticsObserver.
-  BitrateStatistics bitrate;
+  BitrateStatistics total;
+  BitrateStatistics retransmit;
   BitrateStatisticsObserver* bitrate_observer = statistics_proxy_.get();
-  bitrate_observer->Notify(bitrate, exluded_ssrc);
+  bitrate_observer->Notify(total, retransmit, exluded_ssrc);
 
   // From FrameCountObserver.
   FrameCountObserver* fps_observer = statistics_proxy_.get();
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 28231b00..489cd140 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -492,5 +492,12 @@ void VideoSendStream::SignalNetworkState(Call::NetworkState state) {
     rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
 }
 
+int VideoSendStream::GetPacerQueuingDelayMs() const {
+  int pacer_delay_ms = 0;
+  if (rtp_rtcp_->GetPacerQueuingDelayMs(channel_, &pacer_delay_ms) != 0) {
+    return 0;
+  }
+  return pacer_delay_ms;
+}
 }  // namespace internal
 }  // namespace webrtc
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index f7874308..873785dc 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -74,6 +74,8 @@ class VideoSendStream : public webrtc::VideoSendStream,
 
   void SignalNetworkState(Call::NetworkState state);
 
+  int GetPacerQueuingDelayMs() const;
+
  private:
   void ConfigureSsrcs();
   TransportAdapter transport_adapter_;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index b863957b..3ab11274 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -962,8 +962,8 @@ TEST_F(VideoSendStreamTest, ProducesStats) {
                 config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
         // Check for data populated by various sources. RTCP excluded as this
         // data is received from remote side. Tested in call tests instead.
-        const StreamStats& entry = stats.substreams[ssrc];
-        if (entry.key_frames > 0u && entry.bitrate_bps > 0 &&
+        const SsrcStats& entry = stats.substreams[ssrc];
+        if (entry.key_frames > 0u && entry.total_bitrate_bps > 0 &&
             entry.rtp_stats.packets > 0u && entry.avg_delay_ms > 0 &&
             entry.max_delay_ms > 0) {
           return true;
@@ -1045,20 +1045,20 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
       VideoSendStream::Stats stats = stream_->GetStats();
       if (!stats.substreams.empty()) {
         EXPECT_EQ(1u, stats.substreams.size());
-        int bitrate_bps = stats.substreams.begin()->second.bitrate_bps;
-        test::PrintResult(
-            "bitrate_stats_",
-            "min_transmit_bitrate_low_remb",
-            "bitrate_bps",
-            static_cast<size_t>(bitrate_bps),
-            "bps",
-            false);
-        if (bitrate_bps > kHighBitrateBps) {
+        int total_bitrate_bps =
+            stats.substreams.begin()->second.total_bitrate_bps;
+        test::PrintResult("bitrate_stats_",
+                          "min_transmit_bitrate_low_remb",
+                          "bitrate_bps",
+                          static_cast<size_t>(total_bitrate_bps),
+                          "bps",
+                          false);
+        if (total_bitrate_bps > kHighBitrateBps) {
           rtp_rtcp_->SetREMBData(kRembBitrateBps, 1, &header.ssrc);
           rtp_rtcp_->Process();
           bitrate_capped_ = true;
         } else if (bitrate_capped_ &&
-                   bitrate_bps < kRembRespectedBitrateBps) {
+                   total_bitrate_bps < kRembRespectedBitrateBps) {
           observation_complete_->Set();
         }
       }
diff --git a/video_decoder.h b/video_decoder.h
index 03a564e3..941c0ac1 100644
--- a/video_decoder.h
+++ b/video_decoder.h
@@ -40,6 +40,7 @@ class VideoDecoder {
  public:
   enum DecoderType {
     kVp8,
+    kVp9
   };
 
   static VideoDecoder* Create(DecoderType codec_type);
diff --git a/video_engine/vie_base_impl.cc b/video_engine/vie_base_impl.cc
index cf64eada..81c748af 100644
--- a/video_engine/vie_base_impl.cc
+++ b/video_engine/vie_base_impl.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/video_engine/vie_base_impl.h"
 
-#include <sstream>
 #include <string>
 #include <utility>
 
@@ -329,23 +328,8 @@ int ViEBaseImpl::StopReceive(const int video_channel) {
 }
 
 int ViEBaseImpl::GetVersion(char version[1024]) {
-  assert(kViEVersionMaxMessageSize == 1024);
-  if (!version) {
-    shared_data_.SetLastError(kViEBaseInvalidArgument);
-    return -1;
-  }
-
-  // Add WebRTC Version.
-  std::stringstream version_stream;
-  version_stream << "VideoEngine 39" << std::endl;
-
-  // Add build info.
-  version_stream << "Build: " << BUILDINFO << std::endl;
-
-  int version_length = version_stream.tellp();
-  assert(version_length < 1024);
-  memcpy(version, version_stream.str().c_str(), version_length);
-  version[version_length] = '\0';
+  assert(version != NULL);
+  strcpy(version, "VideoEngine 39");
   return 0;
 }
 
diff --git a/video_engine/vie_base_impl.h b/video_engine/vie_base_impl.h
index 20fd6159..0ae7818d 100644
--- a/video_engine/vie_base_impl.h
+++ b/video_engine/vie_base_impl.h
@@ -64,11 +64,6 @@ class ViEBaseImpl
   ViESharedData* shared_data() { return &shared_data_; }
 
  private:
-  // Version functions.
-  int32_t AddViEVersion(char* str) const;
-  int32_t AddBuildInfo(char* str) const;
-  int32_t AddExternalTransportBuild(char* str) const;
-
   int CreateChannel(int& video_channel, int original_channel,  // NOLINT
                     bool sender);
 
diff --git a/video_engine/vie_channel.h b/video_engine/vie_channel.h
index 03279069..3b8d96a2 100644
--- a/video_engine/vie_channel.h
+++ b/video_engine/vie_channel.h
@@ -415,10 +415,12 @@ class ViEChannel
 
   class RegisterableBitrateStatisticsObserver:
     public RegisterableCallback<BitrateStatisticsObserver> {
-    virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) {
+    virtual void Notify(const BitrateStatistics& total_stats,
+                        const BitrateStatistics& retransmit_stats,
+                        uint32_t ssrc) {
       CriticalSectionScoped cs(critsect_.get());
       if (callback_)
-        callback_->Notify(stats, ssrc);
+        callback_->Notify(total_stats, retransmit_stats, ssrc);
     }
   }
   send_bitrate_observer_;
diff --git a/video_engine/vie_defines.h b/video_engine/vie_defines.h
index 7bfed465..74b5e1a7 100644
--- a/video_engine/vie_defines.h
+++ b/video_engine/vie_defines.h
@@ -35,8 +35,6 @@ enum { kViEMinKeyRequestIntervalMs = 300 };
 
 // ViEBase
 enum { kViEMaxNumberOfChannels = 64 };
-enum { kViEVersionMaxMessageSize = 1024 };
-enum { kViEMaxModuleVersionSize = 960 };
 
 // ViECapture
 enum { kViEMaxCaptureDevices = 256 };
@@ -101,21 +99,6 @@ inline int ChannelId(const int moduleId) {
   return static_cast<int>(moduleId & 0xffff);
 }
 
-//  Build information macros
-#if defined(_DEBUG) || defined(DEBUG)
-#define BUILDMODE "d"
-#elif defined(NDEBUG)
-#define BUILDMODE "r"
-#else
-#define BUILDMODE "?"
-#endif
-
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-
-// Example: "Oct 10 2002 12:05:30 r".
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
 // Windows specific.
 #if defined(_WIN32)
   #define RENDER_MODULE_TYPE kRenderWindows
diff --git a/video_receive_stream.h b/video_receive_stream.h
index 5ab898c7..a8620d98 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -63,7 +63,7 @@ class VideoReceiveStream {
     int expected_delay_ms;
   };
 
-  struct Stats : public StreamStats {
+  struct Stats : public SsrcStats {
     Stats()
         : network_frame_rate(0),
           decode_frame_rate(0),
diff --git a/video_send_stream.h b/video_send_stream.h
index aa5033af..a9aba945 100644
--- a/video_send_stream.h
+++ b/video_send_stream.h
@@ -41,11 +41,13 @@ class VideoSendStream {
     Stats()
         : input_frame_rate(0),
           encode_frame_rate(0),
+          media_bitrate_bps(0),
           suspended(false) {}
     int input_frame_rate;
     int encode_frame_rate;
+    int media_bitrate_bps;
     bool suspended;
-    std::map<uint32_t, StreamStats> substreams;
+    std::map<uint32_t, SsrcStats> substreams;
   };
 
   struct Config {
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index ad6314a7..17830950 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -777,15 +777,6 @@ int VoEBaseImpl::GetVersion(char version[1024])
     accLen += len;
     assert(accLen < kVoiceEngineVersionMaxMessageSize);
 
-    len = AddBuildInfo(versionPtr);
-    if (len == -1)
-    {
-        return -1;
-    }
-    versionPtr += len;
-    accLen += len;
-    assert(accLen < kVoiceEngineVersionMaxMessageSize);
-
 #ifdef WEBRTC_EXTERNAL_TRANSPORT
     len = AddExternalTransportBuild(versionPtr);
     if (len == -1)
@@ -828,11 +819,6 @@ int VoEBaseImpl::GetVersion(char version[1024])
     return 0;
 }
 
-int32_t VoEBaseImpl::AddBuildInfo(char* str) const
-{
-    return sprintf(str, "Build: %s\n", BUILDINFO);
-}
-
 int32_t VoEBaseImpl::AddVoEVersion(char* str) const
 {
     return sprintf(str, "VoiceEngine 4.1.0\n");
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index 985ef5d8..2f377362 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -146,7 +146,6 @@ private:
                         int64_t* elapsed_time_ms,
                         int64_t* ntp_time_ms);
 
-    int32_t AddBuildInfo(char* str) const;
     int32_t AddVoEVersion(char* str) const;
 
     // Initialize channel by setting Engine Information then initializing
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index b3cba7c3..cde9470a 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -128,26 +128,6 @@ enum { kVoiceEngineMaxRtpExtensionId = 14 };
 }  // namespace webrtc
 
 // ----------------------------------------------------------------------------
-//  Build information macros
-// ----------------------------------------------------------------------------
-
-#if defined(_DEBUG)
-#define BUILDMODE "d"
-#elif defined(DEBUG)
-#define BUILDMODE "d"
-#elif defined(NDEBUG)
-#define BUILDMODE "r"
-#else
-#define BUILDMODE "?"
-#endif
-
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-
-// Example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-// ----------------------------------------------------------------------------
 //  Macros
 // ----------------------------------------------------------------------------