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 c9a1adf19676a5776bd75e5fb684f301509fb386..d7bf405940500a48862eda7cccda38453a2ecfe3 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
@@ -20,17 +20,15 @@ |
namespace webrtc { |
RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy( |
- int32_t id, RtpData* data_callback, |
+ RtpData* data_callback, |
RtpAudioFeedback* incoming_messages_callback) { |
- return new RTPReceiverAudio(id, data_callback, incoming_messages_callback); |
+ return new RTPReceiverAudio(data_callback, incoming_messages_callback); |
} |
-RTPReceiverAudio::RTPReceiverAudio(const int32_t id, |
- RtpData* data_callback, |
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback, |
RtpAudioFeedback* incoming_messages_callback) |
: RTPReceiverStrategy(data_callback), |
TelephoneEventHandler(), |
- id_(id), |
last_received_frequency_(8000), |
telephone_event_forward_to_decoder_(false), |
telephone_event_payload_type_(-1), |
@@ -263,16 +261,13 @@ int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const { |
int32_t RTPReceiverAudio::InvokeOnInitializeDecoder( |
RtpFeedback* callback, |
- int32_t id, |
int8_t payload_type, |
const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
const PayloadUnion& specific_payload) const { |
- if (-1 == callback->OnInitializeDecoder(id, |
- payload_type, |
- payload_name, |
- specific_payload.Audio.frequency, |
- specific_payload.Audio.channels, |
- specific_payload.Audio.rate)) { |
+ if (-1 == |
+ callback->OnInitializeDecoder( |
+ payload_type, payload_name, specific_payload.Audio.frequency, |
+ specific_payload.Audio.channels, specific_payload.Audio.rate)) { |
LOG(LS_ERROR) << "Failed to create decoder for payload type: " |
<< payload_name << "/" << static_cast<int>(payload_type); |
return -1; |