(Auto)update libjingle 62691533-> 62713454

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/media/base/constants.cc

@@ -96,7 +96,16 @@ const char kGoogleSctpDataCodecName[] = "google-sctp-data";
 
 const char kComfortNoiseCodecName[] = "CN";
 
-const char kRtpAbsoluteSendTimeHeaderExtension[] =
+const int kRtpAudioLevelHeaderExtensionDefaultId = 1;
+const char kRtpAudioLevelHeaderExtension[] =
+    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+
+const int kRtpTimestampOffsetHeaderExtensionDefaultId = 2;
+const char kRtpTimestampOffsetHeaderExtension[] =
+    "urn:ietf:params:rtp-hdrext:toffset";
+
+const int kRtpAbsoluteSenderTimeHeaderExtensionDefaultId = 3;
+const char kRtpAbsoluteSenderTimeHeaderExtension[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 
 const int kNumDefaultUnsignalledVideoRecvStreams = 0;

talk/media/base/constants.h

@@ -116,9 +116,20 @@ extern const char kGoogleSctpDataCodecName[];
 
 extern const char kComfortNoiseCodecName[];
 
+// Extension header for audio levels, as defined in
+// http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
+extern const int kRtpAudioLevelHeaderExtensionDefaultId;
+extern const char kRtpAudioLevelHeaderExtension[];
+
+// Extension header for RTP timestamp offset, see RFC 5450 for details:
+// http://tools.ietf.org/html/rfc5450
+extern const int kRtpTimestampOffsetHeaderExtensionDefaultId;
+extern const char kRtpTimestampOffsetHeaderExtension[];
+
 // Extension header for absolute send time, see url for details:
 // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-extern const char kRtpAbsoluteSendTimeHeaderExtension[];
+extern const int kRtpAbsoluteSenderTimeHeaderExtensionDefaultId;
+extern const char kRtpAbsoluteSenderTimeHeaderExtension[];
 
 extern const int kNumDefaultUnsignalledVideoRecvStreams;
 }  // namespace cricket

talk/media/base/fakemediaengine.h

@@ -737,6 +737,10 @@ class FakeBaseEngine {
   const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
     return rtp_header_extensions_;
   }
+  void set_rtp_header_extensions(
+      const std::vector<RtpHeaderExtension>& extensions) {
+    rtp_header_extensions_ = extensions;
+  }
 
  protected:
   int loglevel_;
@@ -959,18 +963,25 @@ class FakeMediaEngine :
   }
   virtual ~FakeMediaEngine() {}
 
-  virtual void SetAudioCodecs(const std::vector<AudioCodec> codecs) {
+  void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
     voice_.SetCodecs(codecs);
   }
-
+  void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
-  virtual void SetVideoCodecs(const std::vector<VideoCodec> codecs) {
     video_.SetCodecs(codecs);
   }
 
+  void SetAudioRtpHeaderExtensions(
+      const std::vector<RtpHeaderExtension>& extensions) {
+    voice_.set_rtp_header_extensions(extensions);
+  }
+  void SetVideoRtpHeaderExtensions(
+      const std::vector<RtpHeaderExtension>& extensions) {
+    video_.set_rtp_header_extensions(extensions);
+  }
+
   FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
     return voice_.GetChannel(index);
   }
-
   FakeVideoMediaChannel* GetVideoChannel(size_t index) {
     return video_.GetChannel(index);
   }

talk/media/webrtc/fakewebrtcvoiceengine.h

@@ -60,6 +60,15 @@ static const int kFakeDeviceId = 1;
 #endif
 
 
+// Verify the header extension ID, if enabled, is within the bounds specified in
+// [RFC5285]: 1-14 inclusive.
+#define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
+  do { \
+    if (enable && (id < 1 || id > 14)) { \
+      return -1; \
+    } \
+  } while (0);
+
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
@@ -98,7 +107,9 @@ class FakeWebRtcVoiceEngine
           fec_type(117),
           nack_max_packets(0),
           send_ssrc(0),
-          level_header_ext_(-1),
+          send_audio_level_ext_(-1),
+          send_absolute_sender_time_ext_(-1),
+          receive_absolute_sender_time_ext_(-1),
           using_experimental_acm(use_experimental_acm) {
       memset(&send_codec, 0, sizeof(send_codec));
       memset(&rx_agc_config, 0, sizeof(rx_agc_config));
@@ -123,7 +134,9 @@ class FakeWebRtcVoiceEngine
     int fec_type;
     int nack_max_packets;
     uint32 send_ssrc;
-    int level_header_ext_;
+    int send_audio_level_ext_;
+    int send_absolute_sender_time_ext_;
+    int receive_absolute_sender_time_ext_;
     DtmfInfo dtmf_info;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
@@ -274,6 +287,15 @@ class FakeWebRtcVoiceEngine
     channels_[++last_channel_] = ch;
     return last_channel_;
   }
+  int GetSendAudioLevelId(int channel) {
+    return channels_[channel]->send_audio_level_ext_;
+  }
+  int GetSendAbsoluteSenderTimeId(int channel) {
+    return channels_[channel]->send_absolute_sender_time_ext_;
+  }
+  int GetReceiveAbsoluteSenderTimeId(int channel) {
+    return channels_[channel]->receive_absolute_sender_time_ext_;
+  }
 
   WEBRTC_STUB(Release, ());
 
@@ -489,7 +511,6 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(SetDtmfPlayoutStatus, (int channel, bool enable));
   WEBRTC_STUB(GetDtmfPlayoutStatus, (int channel, bool& enabled));
 
-
   WEBRTC_FUNC(PlayDtmfTone,
       (int event_code, int length_ms = 200, int attenuation_db = 10)) {
     dtmf_info_.dtmf_event_code = event_code;
@@ -685,24 +706,41 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
   WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
+#ifndef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetRTPAudioLevelIndicationStatus, (int channel, bool enable,
       unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    if (enable && (id < 1 || id > 14)) {
+    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
-      // [RFC5285] The 4-bit ID is the local identifier of this element in
+    channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
-      // the range 1-14 inclusive.
-      return -1;
-    }
-    channels_[channel]->level_header_ext_ = (enable) ? id : -1;
     return 0;
   }
-  WEBRTC_FUNC(GetRTPAudioLevelIndicationStatus, (int channel, bool& enabled,
+  WEBRTC_STUB(GetRTPAudioLevelIndicationStatus, (int channel, bool& enable,
-      unsigned char& id)) {
+      unsigned char& id));
+#endif
+#ifdef USE_WEBRTC_DEV_BRANCH
+  WEBRTC_FUNC(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
+      unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    enabled = (channels_[channel]->level_header_ext_ != -1);
+    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
-    id = channels_[channel]->level_header_ext_;
+    channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
     return 0;
   }
+  WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
+      unsigned char id)) {
+    WEBRTC_CHECK_CHANNEL(channel);
+    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
+    channels_[channel]->send_absolute_sender_time_ext_ = (enable) ? id : -1;
+    return 0;
+  }
+  WEBRTC_FUNC(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
+      unsigned char id)) {
+    WEBRTC_CHECK_CHANNEL(channel);
+    WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
+    channels_[channel]->receive_absolute_sender_time_ext_ = (enable) ? id : -1;
+    return 0;
+  }
+#endif
+
   WEBRTC_STUB(GetRemoteCSRCs, (int channel, unsigned int arrCSRC[15]));
   WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
   WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
@@ -1068,6 +1106,8 @@ class FakeWebRtcVoiceEngine
   webrtc::VoEMediaProcess* media_processor_;
 };
 
+#undef WEBRTC_CHECK_HEADER_EXTENSION_ID
+
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_

talk/media/webrtc/webrtcvideoengine.cc

@@ -146,14 +146,6 @@ static const int kCpuMonitorPeriodMs = 2000;  // 2 seconds.
 
 static const bool kNotSending = false;
 
-// Extension header for RTP timestamp offset, see RFC 5450 for details:
-// http://tools.ietf.org/html/rfc5450
-static const char kRtpTimestampOffsetHeaderExtension[] =
-    "urn:ietf:params:rtp-hdrext:toffset";
-static const int kRtpTimeOffsetExtensionId = 2;
-
-// Extension header ID for absolute send time. Url defined in constants.cc
-static const int kRtpAbsoluteSendTimeExtensionId = 3;
 // Default video dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details
 // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
@@ -916,10 +908,10 @@ void WebRtcVideoEngine::Construct(ViEWrapper* vie_wrapper,
   // Load our RTP Header extensions.
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
-                         kRtpTimeOffsetExtensionId));
+                         kRtpTimestampOffsetHeaderExtensionDefaultId));
   rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAbsoluteSendTimeHeaderExtension,
+      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
-                         kRtpAbsoluteSendTimeExtensionId));
+                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
 }
 
 WebRtcVideoEngine::~WebRtcVideoEngine() {
@@ -2684,7 +2676,7 @@ bool WebRtcVideoMediaChannel::SetRecvRtpHeaderExtensions(
   const RtpHeaderExtension* offset_extension =
       FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
   const RtpHeaderExtension* send_time_extension =
-      FindHeaderExtension(extensions, kRtpAbsoluteSendTimeHeaderExtension);
+      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
 
   // Loop through all receive channels and enable/disable the extensions.
   for (RecvChannelMap::iterator channel_it = recv_channels_.begin();
@@ -2711,7 +2703,7 @@ bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
   const RtpHeaderExtension* offset_extension =
       FindHeaderExtension(extensions, kRtpTimestampOffsetHeaderExtension);
   const RtpHeaderExtension* send_time_extension =
-      FindHeaderExtension(extensions, kRtpAbsoluteSendTimeHeaderExtension);
+      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
 
   // Loop through all send channels and enable/disable the extensions.
   for (SendChannelMap::iterator channel_it = send_channels_.begin();
@@ -2742,7 +2734,7 @@ bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
 
 int WebRtcVideoMediaChannel::GetRtpSendTimeExtnId() const {
   const RtpHeaderExtension* send_time_extension = FindHeaderExtension(
-      send_extensions_, kRtpAbsoluteSendTimeHeaderExtension);
+      send_extensions_, kRtpAbsoluteSenderTimeHeaderExtension);
   if (send_time_extension) {
     return send_time_extension->id;
   }
@@ -3271,10 +3263,9 @@ bool WebRtcVideoMediaChannel::ConfigureReceiving(int channel_id,
       channel_id, receive_extensions_, kRtpTimestampOffsetHeaderExtension)) {
     return false;
   }
-
   if (!SetHeaderExtension(
       &webrtc::ViERTP_RTCP::SetReceiveAbsoluteSendTimeStatus, channel_id,
-      receive_extensions_, kRtpAbsoluteSendTimeHeaderExtension)) {
+      receive_extensions_, kRtpAbsoluteSenderTimeHeaderExtension)) {
     return false;
   }
 
@@ -3389,7 +3380,7 @@ bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id,
   }
 
   if (!SetHeaderExtension(&webrtc::ViERTP_RTCP::SetSendAbsoluteSendTimeStatus,
-      channel_id, send_extensions_, kRtpAbsoluteSendTimeHeaderExtension)) {
+      channel_id, send_extensions_, kRtpAbsoluteSenderTimeHeaderExtension)) {
     return false;
   }
 

talk/media/webrtc/webrtcvoiceengine.cc

@@ -107,12 +107,6 @@ static const int kDefaultSoundclipDeviceId = -2;
 static const int kDefaultAudioDeviceId = 0;
 #endif
 
-// extension header for audio levels, as defined in
-// http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
-static const char kRtpAudioLevelHeaderExtension[] =
-    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
-static const int kRtpAudioLevelHeaderExtensionId = 1;
-
 static const char kIsacCodecName[] = "ISAC";
 static const char kL16CodecName[] = "L16";
 // Codec parameters for Opus.
@@ -391,7 +385,12 @@ void WebRtcVoiceEngine::Construct() {
   // Load our RTP Header extensions.
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
-                         kRtpAudioLevelHeaderExtensionId));
+                         kRtpAudioLevelHeaderExtensionDefaultId));
+#ifdef USE_WEBRTC_DEV_BRANCH
+  rtp_header_extensions_.push_back(
+      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+#endif
   options_ = GetDefaultEngineOptions();
 
   // Initialize the VoE Configuration to the default ACM.
@@ -2218,43 +2217,74 @@ bool WebRtcVoiceMediaChannel::SetSendCodec(
 
 bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
-  // We don't support any incoming extensions headers right now.
+#ifdef USE_WEBRTC_DEV_BRANCH
+  const RtpHeaderExtension* send_time_extension =
+      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
+
+  // Loop through all receive channels and enable/disable the extensions.
+  for (ChannelMap::const_iterator channel_it = receive_channels_.begin();
+       channel_it != receive_channels_.end(); ++channel_it) {
+    int channel_id = channel_it->second->channel();
+    if (!SetHeaderExtension(
+        &webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
+        send_time_extension)) {
+      return false;
+    }
+  }
+#endif
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
-  // Enable the audio level extension header if requested.
+  const RtpHeaderExtension* audio_level_extension =
-  std::vector<RtpHeaderExtension>::const_iterator it;
+      FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
-  for (it = extensions.begin(); it != extensions.end(); ++it) {
+#ifdef USE_WEBRTC_DEV_BRANCH
-    if (it->uri == kRtpAudioLevelHeaderExtension) {
+  const RtpHeaderExtension* send_time_extension =
-      break;
+      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
-    }
+#endif
-  }
 
-  bool enable = (it != extensions.end());
+#ifndef USE_WEBRTC_DEV_BRANCH
-  int id = 0;
+  if (!SetHeaderExtension(
-
+      &webrtc::VoERTP_RTCP::SetRTPAudioLevelIndicationStatus, voe_channel(),
-  if (enable) {
+      audio_level_extension)) {
-    id = it->id;
-    if (id < kMinRtpHeaderExtensionId ||
-        id > kMaxRtpHeaderExtensionId) {
-      LOG(LS_WARNING) << "Invalid RTP header extension id " << id;
     return false;
   }
-  }
+#else
-
+  if (!SetHeaderExtension(
-  LOG(LS_INFO) << "Enabling audio level header extension with ID " << id;
+      &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, voe_channel(),
-  for (ChannelMap::const_iterator iter = send_channels_.begin();
+      audio_level_extension)) {
-       iter != send_channels_.end(); ++iter) {
-    if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus(
-            iter->second->channel(), enable, id) == -1) {
-      LOG_RTCERR3(SetRTPAudioLevelIndicationStatus,
-                  iter->second->channel(), enable, id);
     return false;
   }
+  if (!SetHeaderExtension(
+      &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, voe_channel(),
+      send_time_extension)) {
+    return false;
   }
+#endif
 
+  for (ChannelMap::const_iterator channel_it = send_channels_.begin();
+       channel_it != send_channels_.end(); ++channel_it) {
+    int channel_id = channel_it->second->channel();
+#ifndef USE_WEBRTC_DEV_BRANCH
+    if (!SetHeaderExtension(
+        &webrtc::VoERTP_RTCP::SetRTPAudioLevelIndicationStatus, channel_id,
+        audio_level_extension)) {
+      return false;
+    }
+#else
+    if (!SetHeaderExtension(
+        &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
+        audio_level_extension)) {
+      return false;
+    }
+    if (!SetHeaderExtension(
+        &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
+        send_time_extension)) {
+      return false;
+    }
+#endif
+  }
   return true;
 }
 
@@ -3485,6 +3515,21 @@ VoiceMediaChannel::Error
   }
 }
 
+bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
+    int channel_id, const RtpHeaderExtension* extension) {
+  bool enable = false;
+  unsigned char id = 0;
+  if (extension) {
+    enable = true;
+    id = extension->id;
+  }
+  if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
+    LOG_RTCERR4(*setter, extension->uri, channel_id, enable, id);
+    return false;
+  }
+  return true;
+}
+
 int WebRtcSoundclipStream::Read(void *buf, int len) {
   size_t res = 0;
   mem_.Read(buf, len, &res, NULL);

talk/media/webrtc/webrtcvoiceengine.h

@@ -397,6 +397,8 @@ class WebRtcVoiceMediaChannel
   // WebRtcVoiceChannelRenderer will be created for every new stream and
   // will be destroyed when the stream goes away.
   typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap;
+  typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
+      unsigned char);
 
   void SetNack(int channel, bool nack_enabled);
   void SetNack(const ChannelMap& channels, bool nack_enabled);
@@ -417,6 +419,9 @@ class WebRtcVoiceMediaChannel
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBandwidthInternal(int bps);
 
+  bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
+                          const RtpHeaderExtension* extension);
+
   talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
   std::set<int> ringback_channels_;  // channels playing ringback
   std::vector<AudioCodec> recv_codecs_;

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -1,6 +1,29 @@
-// Copyright 2008 Google Inc.
+/*
-//
+ * libjingle
-// Author: Justin Uberti (juberti@google.com)
+ * Copyright 2008 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
 
 #ifdef WIN32
 #include "talk/base/win32.h"
@@ -226,43 +249,95 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
   void TestSetSendRtpHeaderExtensions(int channel_id) {
     std::vector<cricket::RtpHeaderExtension> extensions;
-    bool enable = false;
-    unsigned char id = 0;
 
-    // Ensure audio levels are off by default.
+    // Ensure extensions are off by default.
-    EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+    EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-        channel_id, enable, id));
+#ifdef USE_WEBRTC_DEV_BRANCH
-    EXPECT_FALSE(enable);
+    EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
+#endif
 
     // Ensure unknown extensions won't cause an error.
     extensions.push_back(cricket::RtpHeaderExtension(
-        "urn:ietf:params:unknowextention", 1));
+        "urn:ietf:params:unknownextention", 1));
     EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
-    EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+    EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-        channel_id, enable, id));
+#ifdef USE_WEBRTC_DEV_BRANCH
-    EXPECT_FALSE(enable);
+    EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
+#endif
 
-    // Ensure audio levels stay off with an empty list of headers.
+    // Ensure extensions stay off with an empty list of headers.
+    extensions.clear();
     EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
-    EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+    EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-        channel_id, enable, id));
+#ifdef USE_WEBRTC_DEV_BRANCH
-    EXPECT_FALSE(enable);
+    EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
+#endif
 
-    // Ensure audio levels are enabled if the audio-level header is specified.
+    // Ensure audio levels are enabled if the audio-level header is specified
+    // (but AST is still off).
     extensions.push_back(cricket::RtpHeaderExtension(
         "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
     EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
-    EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+    EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
-        channel_id, enable, id));
+#ifdef USE_WEBRTC_DEV_BRANCH
-    EXPECT_TRUE(enable);
+    EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
-    EXPECT_EQ(8, id);
+#endif
 
-    // Ensure audio levels go back off with an empty list.
+#ifdef USE_WEBRTC_DEV_BRANCH
+    // Ensure audio level and AST are enabled if the extensions are specified.
+    extensions.push_back(cricket::RtpHeaderExtension(
+        "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 12));
+    EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+    EXPECT_EQ(8, voe_.GetSendAudioLevelId(channel_id));
+    EXPECT_EQ(12, voe_.GetSendAbsoluteSenderTimeId(channel_id));
+#endif
+
+    // Ensure all extensions go back off with an empty list.
     extensions.clear();
     EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
-    EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
+    EXPECT_EQ(-1, voe_.GetSendAudioLevelId(channel_id));
-        channel_id, enable, id));
+#ifdef USE_WEBRTC_DEV_BRANCH
-    EXPECT_FALSE(enable);
+    EXPECT_EQ(-1, voe_.GetSendAbsoluteSenderTimeId(channel_id));
+#endif
+  }
+
+  void TestSetRecvRtpHeaderExtensions(int channel_id) {
+    std::vector<cricket::RtpHeaderExtension> extensions;
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+    // Ensure extensions are off by default.
+    EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
+#endif
+
+    // Ensure unknown extensions won't cause an error.
+    extensions.push_back(cricket::RtpHeaderExtension(
+        "urn:ietf:params:unknownextention", 1));
+    EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+#ifdef USE_WEBRTC_DEV_BRANCH
+    EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
+#endif
+
+    // An empty list shouldn't cause any headers to be enabled.
+    extensions.clear();
+    EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+#ifdef USE_WEBRTC_DEV_BRANCH
+    EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
+#endif
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+    // Nor should indicating we can receive the absolute sender time header.
+    extensions.push_back(cricket::RtpHeaderExtension(
+        "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 11));
+    EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+    EXPECT_EQ(11, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
+#endif
+
+    // Resetting to an empty list shouldn't cause any headers to be enabled.
+    extensions.clear();
+    EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+#ifdef USE_WEBRTC_DEV_BRANCH
+    EXPECT_EQ(-1, voe_.GetReceiveAbsoluteSenderTimeId(channel_id));
+#endif
   }
 
  protected:
@@ -1488,35 +1563,17 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
   EXPECT_FALSE(voe_.GetFEC(channel_num));
 }
 
-// Test that we support setting an empty list of recv header extensions.
-TEST_F(WebRtcVoiceEngineTestFake, SetRecvRtpHeaderExtensions) {
-  EXPECT_TRUE(SetupEngine());
-  std::vector<cricket::RtpHeaderExtension> extensions;
-  int channel_num = voe_.GetLastChannel();
-  bool enable = false;
-  unsigned char id = 0;
-
-  // An empty list shouldn't cause audio-level headers to be enabled.
-  EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-  EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
-      channel_num, enable, id));
-  EXPECT_FALSE(enable);
-
-  // Nor should indicating we can receive the audio-level header.
-  extensions.push_back(cricket::RtpHeaderExtension(
-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8));
-  EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-  EXPECT_EQ(0, voe_.GetRTPAudioLevelIndicationStatus(
-      channel_num, enable, id));
-  EXPECT_FALSE(enable);
-}
-
 // Test that we support setting certain send header extensions.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendRtpHeaderExtensions) {
   EXPECT_TRUE(SetupEngine());
-  std::vector<cricket::RtpHeaderExtension> extensions;
+  TestSetSendRtpHeaderExtensions(voe_.GetLastChannel());
-  int channel_num = voe_.GetLastChannel();
+}
-  TestSetSendRtpHeaderExtensions(channel_num);
+
+// Test that we support setting recv header extensions.
+TEST_F(WebRtcVoiceEngineTestFake, SetRecvRtpHeaderExtensions) {
+  EXPECT_TRUE(SetupEngine());
+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
+  TestSetRecvRtpHeaderExtensions(voe_.GetLastChannel());
 }
 
 // Test that we can create a channel and start sending/playing out on it.
@@ -1676,26 +1733,41 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
   EXPECT_EQ(1u, info.receivers.size());
 }
 
-// Test that we support setting certain send header extensions on multiple
+// Test that we support setting header extensions on multiple send streams.
-// send streams.
 TEST_F(WebRtcVoiceEngineTestFake,
-       SetSendRtpHeaderExtensionsWithMultpleSendStreams) {
+       SetSendRtpHeaderExtensionsWithMultipleSendStreams) {
   SetupForMultiSendStream();
 
-  static const uint32 kSsrcs4[] = {1, 2, 3, 4};
+  static const uint32 kSsrcs[] = {1, 2, 3, 4};
-  // Create send streams.
+  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
-  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
     EXPECT_TRUE(channel_->AddSendStream(
-        cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
+        cricket::StreamParams::CreateLegacy(kSsrcs[i])));
   }
 
-  // Test SendRtpHeaderExtensions on each send channel.
+  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
-  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
+    int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs[i]);
-    int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
     TestSetSendRtpHeaderExtensions(channel_num);
   }
 }
 
+// Test that we support setting header extensions on multiple receive streams.
+TEST_F(WebRtcVoiceEngineTestFake,
+       SetRecvRtpHeaderExtensionsWithMultipleRecvStreams) {
+  EXPECT_TRUE(SetupEngine());
+
+  static const uint32 kSsrcs[] = {1, 2, 3, 4};
+  int channel_ids[ARRAY_SIZE(kSsrcs)] = {0};
+  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
+    EXPECT_TRUE(channel_->AddRecvStream(
+        cricket::StreamParams::CreateLegacy(kSsrcs[i])));
+    channel_ids[i] = voe_.GetLastChannel();
+  }
+
+  for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
+    TestSetRecvRtpHeaderExtensions(channel_ids[i]);
+  }
+}
+
 // Test that we can add and remove receive streams, and do proper send/playout.
 // We can receive on multiple streams while sending one stream.
 TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {

talk/session/media/channel.cc

@@ -1216,7 +1216,7 @@ bool BaseChannel::SetBaseRemoteContent_w(const MediaContentDescription* content,
 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
     const std::vector<RtpHeaderExtension>& extensions) {
   const RtpHeaderExtension* send_time_extension =
-      FindHeaderExtension(extensions, kRtpAbsoluteSendTimeHeaderExtension);
+      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
   rtp_abs_sendtime_extn_id_ =
       send_time_extension ? send_time_extension->id : -1;
 }

talk/session/media/mediasession.cc

@@ -398,9 +398,9 @@ class UsedRtpHeaderExtensionIds : public UsedIds<RtpHeaderExtension> {
   }
 
  private:
-  // Min and Max local identifier as specified by RFC5285.
+  // Min and Max local identifier for one-byte header extensions, per RFC5285.
   static const int kLocalIdMin = 1;
-  static const int kLocalIdMax = 255;
+  static const int kLocalIdMax = 14;
 };
 
 static bool IsSctp(const MediaContentDescription* desc) {
@@ -854,12 +854,16 @@ static bool FindByUri(const RtpHeaderExtensions& extensions,
 static void FindAndSetRtpHdrExtUsed(
   const RtpHeaderExtensions& reference_extensions,
   RtpHeaderExtensions* offered_extensions,
+  const RtpHeaderExtensions& other_extensions,
   UsedRtpHeaderExtensionIds* used_extensions) {
   for (RtpHeaderExtensions::const_iterator it = reference_extensions.begin();
       it != reference_extensions.end(); ++it) {
     if (!FindByUri(*offered_extensions, *it, NULL)) {
-      RtpHeaderExtension ext = *it;
+      RtpHeaderExtension ext;
+      if (!FindByUri(other_extensions, *it, &ext)) {
+        ext = *it;
         used_extensions->FindAndSetIdUsed(&ext);
+      }
       offered_extensions->push_back(ext);
     }
   }
@@ -1509,6 +1513,8 @@ void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
     const SessionDescription* current_description,
     RtpHeaderExtensions* audio_extensions,
     RtpHeaderExtensions* video_extensions) const {
+  // All header extensions allocated from the same range to avoid potential
+  // issues when using BUNDLE.
   UsedRtpHeaderExtensionIds used_ids;
   audio_extensions->clear();
   video_extensions->clear();
@@ -1535,9 +1541,9 @@ void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
   // Add our default RTP header extensions that are not in
   // |current_description|.
   FindAndSetRtpHdrExtUsed(audio_rtp_header_extensions(), audio_extensions,
-                          &used_ids);
+                          *video_extensions, &used_ids);
   FindAndSetRtpHdrExtUsed(video_rtp_header_extensions(), video_extensions,
-                          &used_ids);
+                          *audio_extensions, &used_ids);
 }
 
 bool MediaSessionDescriptionFactory::AddTransportOffer(

talk/session/media/mediasession_unittest.cc

@@ -143,6 +143,7 @@ static const RtpHeaderExtension kAudioRtpExtension1[] = {
 static const RtpHeaderExtension kAudioRtpExtension2[] = {
   RtpHeaderExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 2),
   RtpHeaderExtension("http://google.com/testing/audio_something_else", 8),
+  RtpHeaderExtension("http://google.com/testing/both_audio_and_video", 7),
 };
 
 static const RtpHeaderExtension kAudioRtpExtensionAnswer[] = {
@@ -151,12 +152,13 @@ static const RtpHeaderExtension kAudioRtpExtensionAnswer[] = {
 
 static const RtpHeaderExtension kVideoRtpExtension1[] = {
   RtpHeaderExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
-  RtpHeaderExtension("http://google.com/testing/video_something", 15),
+  RtpHeaderExtension("http://google.com/testing/video_something", 13),
 };
 
 static const RtpHeaderExtension kVideoRtpExtension2[] = {
   RtpHeaderExtension("urn:ietf:params:rtp-hdrext:toffset", 2),
   RtpHeaderExtension("http://google.com/testing/video_something_else", 14),
+  RtpHeaderExtension("http://google.com/testing/both_audio_and_video", 7),
 };
 
 static const RtpHeaderExtension kVideoRtpExtensionAnswer[] = {
@@ -1616,17 +1618,19 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // extensions from the first offer/answer exchange plus the extensions only
   // |f2_| offer.
   // Since the default local extension id |f2_| uses has already been used by
-  // |f1_| for another extensions, it is changed to 255.
+  // |f1_| for another extensions, it is changed to 13.
   const RtpHeaderExtension kUpdatedAudioRtpExtensions[] = {
     kAudioRtpExtensionAnswer[0],
-    RtpHeaderExtension(kAudioRtpExtension2[1].uri, 255),
+    RtpHeaderExtension(kAudioRtpExtension2[1].uri, 13),
+    kAudioRtpExtension2[2],
   };
 
   // Since the default local extension id |f2_| uses has already been used by
-  // |f1_| for another extensions, is is changed to 254.
+  // |f1_| for another extensions, is is changed to 12.
   const RtpHeaderExtension kUpdatedVideoRtpExtensions[] = {
     kVideoRtpExtensionAnswer[0],
-    RtpHeaderExtension(kVideoRtpExtension2[1].uri, 254),
+    RtpHeaderExtension(kVideoRtpExtension2[1].uri, 12),
+    kVideoRtpExtension2[2],
   };
 
   const AudioContentDescription* updated_acd =