Index: webrtc/modules/audio_coding/neteq/packet_buffer.h |
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.h b/webrtc/modules/audio_coding/neteq/packet_buffer.h |
index 6867b4cb37e03fe0b5c3432dc35f9301da5c5a4e..be2ecebaa379f871bcddd4da97fc3714cbdb43a3 100644 |
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.h |
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.h |
@@ -12,6 +12,7 @@ |
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ |
#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/optional.h" |
#include "webrtc/modules/audio_coding/neteq/packet.h" |
#include "webrtc/typedefs.h" |
@@ -59,10 +60,11 @@ class PacketBuffer { |
// The last three parameters are included for legacy compatibility. |
// TODO(hlundin): Redesign to not use current_*_payload_type and |
// decoder_database. |
- virtual int InsertPacketList(PacketList* packet_list, |
- const DecoderDatabase& decoder_database, |
- uint8_t* current_rtp_payload_type, |
- uint8_t* current_cng_rtp_payload_type); |
+ virtual int InsertPacketList( |
+ PacketList* packet_list, |
+ const DecoderDatabase& decoder_database, |
+ rtc::Optional<uint8_t>* current_rtp_payload_type, |
+ rtc::Optional<uint8_t>* current_cng_rtp_payload_type); |
// Gets the timestamp for the first packet in the buffer and writes it to the |
// output variable |next_timestamp|. |