Wire up transport sequence number / send time callbacks to webrtc via libjingle.

talk/session/media/channel.cc

Index: talk/session/media/channel.cc
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 5a6b7e198f16cb8380790e82d4eda8bc958e4c2a..9e4cc3ff9f37a22228b6b13c5dfa9cff9adf1958 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -191,7 +191,8 @@ BaseChannel::BaseChannel(rtc::Thread* thread,
       has_received_packet_(false),
       dtls_keyed_(false),
       secure_required_(false),
-      rtp_abs_sendtime_extn_id_(-1) {
+      rtp_abs_sendtime_extn_id_(-1),
+      rtp_transport_sequence_number_extn_id_(-1) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   LOG(LS_INFO) << "Created channel for " << content_name;
 }
@@ -343,6 +344,7 @@ void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
 
   tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
   tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
+  tc->SignalSentPacket.connect(this, &BaseChannel::OnPacketSent);
   tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);
 }
 
@@ -477,6 +479,14 @@ void BaseChannel::OnChannelRead(TransportChannel* channel,
   HandlePacket(rtcp, &packet, packet_time);
 }
 
+void BaseChannel::OnPacketSent(TransportChannel* channel,
+                               const rtc::SentPacket& packet_sent) {
+  ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
+  if (channel == transport_channel_) {
+    media_channel_->OnPacketSent(packet_sent);

[pthatcher1, 2015/09/25 23:24:57]

This means that all 3 channels (audio, video, and data) will get called back for
all of the packets for all three.  That seems a bit overkill.  And in the future
when we have N channels, it will be worse.  Can we just call directly into the
Call object somehow?

[stefan-webrtc, 2015/09/28 12:10:49]

That would be nicer. I'll look into it.

+  }
+}
+
 void BaseChannel::OnReadyToSend(TransportChannel* channel) {
   ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
   SetReadyToSend(channel == rtcp_transport_channel_, true);
@@ -561,6 +571,8 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
 #else
       options.packet_time_params.rtp_sendtime_extension_id =
           rtp_abs_sendtime_extn_id_;
+      options.transport_sequence_number =
+          ParseTransportSequenceNumberHeaderExtension(data, len);

[pthatcher1, 2015/09/25 23:24:57]

I really think is the wrong place for this.  It would be much better to change
BaseChannel::SendPacket to take a packet_id as parameter, perhaps by replacing
dscp with packet_options like it is in the lower layers.  Then the layer
creating the packet can just put the ID on the packet rather than having to do
this crazy parsing thing.

[stefan-webrtc, 2015/09/28 12:10:50]

Yes, I definitely agree that it would make more sense to pass the sequence
number with the packet all the way from where it is created. I'll discuss with
others to see if we can change the WebRTC transport interface to allow this.

[stefan-webrtc, 2015/10/02 13:29:12]

Done.

       res = srtp_filter_.ProtectRtp(
           data, len, static_cast<int>(packet->capacity()), &len,
           &options.packet_time_params.srtp_packet_index);
@@ -625,6 +637,88 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
   return true;
 }
 
+int32_t BaseChannel::ParseTransportSequenceNumberHeaderExtension(
+    uint8_t* data,
+    size_t length) const {
+  //  0                   1                   2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |                           timestamp                           |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |           synchronization source (SSRC) identifier            |
+  // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  // |            contributing source (CSRC) identifiers             |
+  // |                             ....                              |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+  // Return if extension bit is not set.
+  if (!(data[0] & 0x10)) {
+    return -1;
+  }
+
+  size_t cc_count = data[0] & 0x0F;
+  const size_t kMinRtpHeaderLength = 12;
+  size_t header_length_without_extension = kMinRtpHeaderLength + 4 * cc_count;
+
+  data += header_length_without_extension;
+
+  // Getting extension profile ID and length.
+  uint16 profile_id = rtc::GetBE16(data);
+  // Length is in 32 bit words.
+  uint16 extension_length_in_32bits = rtc::GetBE16(data + 2);
+  size_t extension_length = extension_length_in_32bits * 4;
+
+  const size_t kRtpExtensionHeaderLength = 4;
+  data += kRtpExtensionHeaderLength;  // Moving past extension header.
+
+  int32_t transport_sequence_number = -1;
+  // The transport sequence number is a one byte header extension.
+  if (profile_id == 0xBEDE) {  // One byte extension header.
+    //  0
+    //  0 1 2 3 4 5 6 7
+    // +-+-+-+-+-+-+-+-+
+    // |  ID   |length |
+    // +-+-+-+-+-+-+-+-+
+
+    //  0                   1                   2                   3
+    //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |       0xBE    |    0xDE       |           length=3            |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |  ID   | L=0   |     data      |  ID   |  L=1  |   data...
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //       ...data   |    0 (pad)    |    0 (pad)    |  ID   | L=3   |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                          data                                 |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    const uint8_t* extension_start = data;
+    const uint8_t* extension_end = extension_start + extension_length;
+
+    while (data < extension_end) {
+      const int id = (*data & 0xF0) >> 4;
+      const size_t length = (*data & 0x0F) + 1;
+      const size_t kOneByteHeaderLength = 1;
+      if (data + kOneByteHeaderLength + length > extension_end) {
+        break;
+      }
+      // The 4-bit length is the number minus one of data bytes of this header
+      // extension element following the one-byte header.
+      if (id == rtp_transport_sequence_number_extn_id_) {
+        transport_sequence_number = rtc::GetBE24(data + kOneByteHeaderLength);
+        break;
+      }
+      data += kOneByteHeaderLength + length;
+      // Counting padding bytes.
+      while ((data < extension_end) && (*data == 0)) {
+        ++data;
+      }
+    }
+  }
+  return transport_sequence_number;
+}
+
 bool BaseChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
   // Protect ourselves against crazy data.
   if (!ValidPacket(rtcp, packet)) {
@@ -1239,12 +1333,17 @@ bool BaseChannel::UpdateRemoteStreams_w(
   return ret;
 }
 
-void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
+void BaseChannel::MaybeCacheRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
   const RtpHeaderExtension* send_time_extension =
       FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
   rtp_abs_sendtime_extn_id_ =
       send_time_extension ? send_time_extension->id : -1;
+
+  const RtpHeaderExtension* transport_sequence_number = FindHeaderExtension(
+      extensions, kRtpTransportSequenceNumberHeaderExtension);
+  rtp_transport_sequence_number_extn_id_ =
+      transport_sequence_number ? transport_sequence_number->id : -1;
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
@@ -1538,7 +1637,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   if (audio->rtp_header_extensions_set()) {
-    MaybeCacheRtpAbsSendTimeHeaderExtension(audio->rtp_header_extensions());
+    MaybeCacheRtpHeaderExtensions(audio->rtp_header_extensions());
   }
 
   set_remote_content_direction(content->direction());
@@ -1864,7 +1963,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   if (video->rtp_header_extensions_set()) {
-    MaybeCacheRtpAbsSendTimeHeaderExtension(video->rtp_header_extensions());
+    MaybeCacheRtpHeaderExtensions(video->rtp_header_extensions());
   }
 
   set_remote_content_direction(content->direction());