| 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_));
|
| }
|
|
|