Index: webrtc/modules/audio_coding/acm2/acm_receiver.cc |
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc |
index 335c2d6bac93a16616545c208d42175b3ceb107b..f45d5d34149acea5c693c1aa36cc73dc06fad8c3 100644 |
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc |
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc |
@@ -213,7 +213,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, |
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { |
enum NetEqOutputType type; |
size_t samples_per_channel; |
- int num_channels; |
+ size_t num_channels; |
// Accessing members, take the lock. |
CriticalSectionScoped lock(crit_sect_.get()); |
@@ -301,7 +301,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { |
int32_t AcmReceiver::AddCodec(int acm_codec_id, |
uint8_t payload_type, |
- int channels, |
+ size_t channels, |
int sample_rate_hz, |
AudioDecoder* audio_decoder, |
const std::string& name) { |