Index: webrtc/modules/audio_coding/neteq/include/neteq.h |
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h |
index 9cd4b57a1f78ee78c914bbb97611a6f196844dd8..4067ba5c71fefa63b60827c69330aa54f7ba98f0 100644 |
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h |
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h |
@@ -272,10 +272,17 @@ class NetEq { |
virtual void PacketBufferStatistics(int* current_num_packets, |
int* max_num_packets) const = 0; |
- // Get sequence number and timestamp of the latest RTP. |
- // This method is to facilitate NACK. |
- virtual int DecodedRtpInfo(int* sequence_number, |
- uint32_t* timestamp) const = 0; |
+ // Enables NACK and sets the maximum size of the NACK list, which should be |
+ // positive and no larger than Nack::kNackListSizeLimit. If NACK is already |
+ // enabled then the maximum NACK list size is modified accordingly. |
+ virtual void EnableNack(size_t max_nack_list_size) = 0; |
+ |
+ virtual void DisableNack() = 0; |
+ |
+ // Returns a list of RTP sequence numbers corresponding to packets to be |
+ // retransmitted, given an estimate of the round-trip time in milliseconds. |
+ virtual std::vector<uint16_t> GetNackList( |
+ int64_t round_trip_time_ms) const = 0; |
protected: |
NetEq() {} |