Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
index 171448808e94298bd38916eab473413aa0407a92..08af919e21d60d727bb06874af336ea14e63b842 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
@@ -159,7 +159,7 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, |
} |
int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio, |
- size_t* samples_per_channel, int* num_channels, |
+ size_t* samples_per_channel, size_t* num_channels, |
NetEqOutputType* type) { |
CriticalSectionScoped lock(crit_sect_.get()); |
LOG(LS_VERBOSE) << "GetAudio"; |
@@ -709,7 +709,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, |
int NetEqImpl::GetAudioInternal(size_t max_length, |
int16_t* output, |
size_t* samples_per_channel, |
- int* num_channels) { |
+ size_t* num_channels) { |
PacketList packet_list; |
DtmfEvent dtmf_event; |
Operations operation; |
@@ -837,7 +837,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, |
size_t samples_from_sync = |
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel, |
output); |
- *num_channels = static_cast<int>(sync_buffer_->Channels()); |
+ *num_channels = sync_buffer_->Channels(); |
LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" << |
" insert " << algorithm_buffer_->Size() << " samples, extract " << |
samples_from_sync << " samples"; |