Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
index 7a3b6fd829002b8a1d30891a02ae5e9fd71a5461..c4c7dbb4cd3eafaf811573115f302af8bafb84b4 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
@@ -20,11 +20,13 @@ |
namespace webrtc { |
RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( |
- RtpData* data_callback) { |
- return new RTPReceiverAudio(data_callback); |
-} |
- |
-RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback) |
+ RtpData* data_callback, |
+ RtpAudioFeedback* incoming_messages_callback) { |
+ return new RTPReceiverAudio(data_callback, incoming_messages_callback); |
+} |
+ |
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback, |
+ RtpAudioFeedback* incoming_messages_callback) |
: RTPReceiverStrategy(data_callback), |
TelephoneEventHandler(), |
last_received_frequency_(8000), |
@@ -38,7 +40,8 @@ |
g722_payload_type_(-1), |
last_received_g722_(false), |
num_energy_(0), |
- current_remote_energy_() { |
+ current_remote_energy_(), |
+ cb_audio_feedback_(incoming_messages_callback) { |
last_payload_.Audio.channels = 1; |
memset(current_remote_energy_, 0, sizeof(current_remote_energy_)); |
} |