Index: webrtc/modules/audio_coding/neteq/packet_buffer.cc |
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/webrtc/modules/audio_coding/neteq/packet_buffer.cc |
index c89de12318b990ed4b3df4344cac2c0ef5e5f8a4..849e48c0064eedff2aa1a0a9c3f39591a181f1fb 100644 |
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc |
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc |
@@ -19,6 +19,7 @@ |
#include "webrtc/base/logging.h" |
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
#include "webrtc/modules/audio_coding/neteq/decoder_database.h" |
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h" |
namespace webrtc { |
@@ -37,8 +38,9 @@ class NewTimestampIsLarger { |
const Packet* new_packet_; |
}; |
-PacketBuffer::PacketBuffer(size_t max_number_of_packets) |
- : max_number_of_packets_(max_number_of_packets) {} |
+PacketBuffer::PacketBuffer(size_t max_number_of_packets, |
+ const TickTimer& tick_timer) |
+ : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {} |
// Destructor. All packets in the buffer will be destroyed. |
PacketBuffer::~PacketBuffer() { |
@@ -65,6 +67,8 @@ int PacketBuffer::InsertPacket(Packet* packet) { |
int return_val = kOK; |
+ packet->waiting_time = tick_timer_.GetNewStopwatch(); |
+ |
if (buffer_.size() >= max_number_of_packets_) { |
// Buffer is full. Flush it. |
Flush(); |
@@ -268,13 +272,6 @@ size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database, |
return num_samples; |
} |
-void PacketBuffer::IncrementWaitingTimes(int inc) { |
- PacketList::iterator it; |
- for (it = buffer_.begin(); it != buffer_.end(); ++it) { |
- (*it)->waiting_time += inc; |
- } |
-} |
- |
bool PacketBuffer::DeleteFirstPacket(PacketList* packet_list) { |
if (packet_list->empty()) { |
return false; |