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 431e0f122c55f4fb6fd97ab51dc6751dd65e4a28..c89de12318b990ed4b3df4344cac2c0ef5e5f8a4 100644 |
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.cc |
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.cc |
@@ -181,7 +181,7 @@ const RTPHeader* PacketBuffer::NextRtpHeader() const { |
return const_cast<const RTPHeader*>(&(buffer_.front()->header)); |
} |
-Packet* PacketBuffer::GetNextPacket(int* discard_count) { |
+Packet* PacketBuffer::GetNextPacket(size_t* discard_count) { |
if (Empty()) { |
// Buffer is empty. |
return NULL; |
@@ -194,7 +194,7 @@ Packet* PacketBuffer::GetNextPacket(int* discard_count) { |
// Discard other packets with the same timestamp. These are duplicates or |
// redundant payloads that should not be used. |
- int discards = 0; |
+ size_t discards = 0; |
while (!Empty() && |
buffer_.front()->header.timestamp == packet->header.timestamp) { |
@@ -240,15 +240,15 @@ int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) { |
return DiscardOldPackets(timestamp_limit, 0); |
} |
-int PacketBuffer::NumPacketsInBuffer() const { |
- return static_cast<int>(buffer_.size()); |
+size_t PacketBuffer::NumPacketsInBuffer() const { |
+ return buffer_.size(); |
} |
-int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database, |
- int last_decoded_length) const { |
+size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database, |
+ size_t last_decoded_length) const { |
PacketList::const_iterator it; |
- int num_samples = 0; |
- int last_duration = last_decoded_length; |
+ size_t num_samples = 0; |
+ size_t last_duration = last_decoded_length; |
for (it = buffer_.begin(); it != buffer_.end(); ++it) { |
Packet* packet = (*it); |
AudioDecoder* decoder = |