| 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";
|
|
|