diff --git a/webrtc/modules/audio_coding/main/acm2/dump.proto b/webrtc/modules/audio_coding/main/acm2/dump.proto
index 416bb7a61b..232faec428 100644
--- a/webrtc/modules/audio_coding/main/acm2/dump.proto
+++ b/webrtc/modules/audio_coding/main/acm2/dump.proto
@@ -10,6 +10,7 @@ message ACMDumpEventStream {
   repeated ACMDumpEvent stream = 1;
 }
 
+
 message ACMDumpEvent {
   // required - Elapsed wallclock time in us since the start of the log.
   optional int64 timestamp_us = 1;
@@ -21,6 +22,7 @@ message ACMDumpEvent {
     UNKNOWN_EVENT = 0;
     RTP_EVENT = 1;
     DEBUG_EVENT = 2;
+    CONFIG_EVENT = 3;
   }
 
   // required - Indicates the type of this event
@@ -31,8 +33,12 @@ message ACMDumpEvent {
 
   // optional - but required if type == DEBUG_EVENT
   optional ACMDumpDebugEvent debug_event = 4;
+
+  // optional - but required if type == CONFIG_EVENT
+  optional ACMDumpConfigEvent config = 5;
 }
 
+
 message ACMDumpRTPPacket {
   // Indicates if the packet is incoming or outgoing with respect to the user
   // that is logging the data.
@@ -58,6 +64,7 @@ message ACMDumpRTPPacket {
   optional bytes RTP_data = 3;
 }
 
+
 message ACMDumpDebugEvent {
   // Indicates the type of the debug event.
   // LOG_START and LOG_END indicate the start and end of the log respectively.
@@ -75,4 +82,88 @@ message ACMDumpDebugEvent {
   // An optional message that can be used to store additional information about
   // the debug event.
   optional string message = 2;
-}
\ No newline at end of file
+}
+
+
+// TODO(terelius): Video and audio streams could in principle share SSRC,
+// so identifying a stream based only on SSRC might not work.
+// It might be better to use a combination of SSRC and media type
+// or SSRC and port number, but for now we will rely on SSRC only.
+message ACMDumpConfigEvent {
+  // Synchronization source (stream identifier) to be received.
+  optional uint32 remote_ssrc = 1;
+
+  // RTX settings for incoming video payloads that may be received. RTX is
+  // disabled if there's no config present.
+  optional RtcpConfig rtcp_config = 3;
+
+  // Map from video RTP payload type -> RTX config.
+  repeated RtxMap rtx_map = 4;
+
+  // RTP header extensions used for the received stream.
+  repeated RtpHeaderExtension header_extensions = 5;
+
+  // List of decoders associated with the stream.
+  repeated DecoderConfig decoders = 6;
+}
+
+
+// Maps decoder names to payload types.
+message DecoderConfig {
+  // required
+  optional string name = 1;
+
+  // required
+  optional sint32 payload_type = 2;
+}
+
+
+// Maps RTP header extension names to numerical ids.
+message RtpHeaderExtension {
+  // required
+  optional string name = 1;
+
+  // required
+  optional sint32 id = 2;
+}
+
+
+// RTX settings for incoming video payloads that may be received.
+// RTX is disabled if there's no config present.
+message RtxConfig {
+  // required - SSRCs to use for the RTX streams.
+  optional uint32 ssrc = 1;
+
+  // required - Payload type to use for the RTX stream.
+  optional sint32 payload_type = 2;
+}
+
+
+message RtxMap {
+  // required
+  optional sint32 payload_type = 1;
+
+  // required
+  optional RtxConfig config = 2;
+}
+
+
+// Configuration information for RTCP.
+// For bandwidth estimation purposes it is more interesting to log the
+// RTCP messages that the sender receives, but we will support logging
+// at the receiver side too.
+message RtcpConfig {
+  // Sender SSRC used for sending RTCP (such as receiver reports).
+  optional uint32 local_ssrc = 1;
+
+  // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+  // RTCP mode is described by RFC 5506.
+  enum RtcpMode {RTCP_COMPOUND = 1; RTCP_REDUCEDSIZE = 2;}
+  optional RtcpMode rtcp_mode = 2;
+
+  // Extended RTCP settings.
+  optional bool receiver_reference_time_report = 3;
+
+  // Receiver estimated maximum bandwidth.
+  optional bool remb = 4;
+}