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Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics().

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R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4644 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2013-08-30 15:36:53 +00:00
parent 9ded07e3a4
commit 3170b5750f
1 changed files with 9 additions and 8 deletions
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17
webrtc/modules/audio_coding/neteq4/neteq_impl.cc
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@ -354,6 +354,15 @@ void NetEqImpl::FlushBuffers() {
first_packet_ = true;
}
void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
int* max_num_packets,
int* current_memory_size_bytes,
int* max_memory_size_bytes) const {
CriticalSectionScoped lock(crit_sect_);
packet_buffer_->BufferStat(current_num_packets, max_num_packets,
current_memory_size_bytes, max_memory_size_bytes);
}
int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) {
CriticalSectionScoped lock(crit_sect_);
if (decoded_packet_sequence_number_ < 0)
@ -1808,12 +1817,4 @@ NetEqOutputType NetEqImpl::LastOutputType() {
}
}
void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
int* max_num_packets,
int* current_memory_size_bytes,
int* max_memory_size_bytes) const {
packet_buffer_->BufferStat(current_num_packets, max_num_packets,
current_memory_size_bytes, max_memory_size_bytes);
}
} // namespace webrtc