| 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 =
|
|
|