Adding debug dump to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto

Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
new file mode 100644
index 0000000000000000000000000000000000000000..f4252449987a341f2ceae6ecc81297f81b2a2923
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -0,0 +1,30 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audio_network_adaptor.debug_dump;
+
+message NetworkMetrics {
+  optional int32 uplink_bandwidth_bps = 1;
+  optional float uplink_packet_loss_fraction = 2;
+  optional int32 target_audio_bitrate_bps = 3;
+  optional int32 rtt_ms = 4;
+}
+
+message EncoderRuntimeConfig {
+  optional int32 bitrate_bps = 1;
+  optional int32 frame_length_ms = 2;
+  optional float uplink_packet_loss_fraction = 3;
+  optional bool enable_fec = 4;
+  optional bool enable_dtx = 5;
+  optional uint32 num_channels = 6;
+}
+
+message Event {
+  enum Type {
+    NETWORK_METRICS = 0;
+    ENCODER_RUNTIME_CONFIG = 1;
+  }
+  required Type type = 1;
+  required uint32 timestamp = 2;
+  optional NetworkMetrics network_metrics = 3;
+  optional EncoderRuntimeConfig encoder_runtime_config = 4;
+}