Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
index 6be0c5a8273615ec33f29a3effadbf0b1f709fa2..40612a6ddfeb2fb84b5ba175b1adf6bd061729a2 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
@@ -27,7 +27,7 @@ using RtpUtility::Payload; |
using RtpUtility::StringCompare; |
RtpReceiver* RtpReceiver::CreateVideoReceiver( |
- int id, Clock* clock, |
+ Clock* clock, |
RtpData* incoming_payload_callback, |
RtpFeedback* incoming_messages_callback, |
RTPPayloadRegistry* rtp_payload_registry) { |
@@ -36,13 +36,13 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver( |
if (!incoming_messages_callback) |
incoming_messages_callback = NullObjectRtpFeedback(); |
return new RtpReceiverImpl( |
- id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback, |
+ clock, NullObjectRtpAudioFeedback(), incoming_messages_callback, |
rtp_payload_registry, |
RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback)); |
} |
RtpReceiver* RtpReceiver::CreateAudioReceiver( |
- int id, Clock* clock, |
+ Clock* clock, |
RtpAudioFeedback* incoming_audio_feedback, |
RtpData* incoming_payload_callback, |
RtpFeedback* incoming_messages_callback, |
@@ -54,25 +54,24 @@ RtpReceiver* RtpReceiver::CreateAudioReceiver( |
if (!incoming_messages_callback) |
incoming_messages_callback = NullObjectRtpFeedback(); |
return new RtpReceiverImpl( |
- id, clock, incoming_audio_feedback, incoming_messages_callback, |
+ clock, incoming_audio_feedback, incoming_messages_callback, |
rtp_payload_registry, |
- RTPReceiverStrategy::CreateAudioStrategy(id, incoming_payload_callback, |
+ RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback, |
incoming_audio_feedback)); |
} |
-RtpReceiverImpl::RtpReceiverImpl(int32_t id, |
- Clock* clock, |
- RtpAudioFeedback* incoming_audio_messages_callback, |
- RtpFeedback* incoming_messages_callback, |
- RTPPayloadRegistry* rtp_payload_registry, |
- RTPReceiverStrategy* rtp_media_receiver) |
+RtpReceiverImpl::RtpReceiverImpl( |
+ Clock* clock, |
+ RtpAudioFeedback* incoming_audio_messages_callback, |
+ RtpFeedback* incoming_messages_callback, |
+ RTPPayloadRegistry* rtp_payload_registry, |
+ RTPReceiverStrategy* rtp_media_receiver) |
: clock_(clock), |
rtp_payload_registry_(rtp_payload_registry), |
rtp_media_receiver_(rtp_media_receiver), |
- id_(id), |
cb_rtp_feedback_(incoming_messages_callback), |
critical_section_rtp_receiver_( |
- CriticalSectionWrapper::CreateCriticalSection()), |
+ CriticalSectionWrapper::CreateCriticalSection()), |
last_receive_time_(0), |
last_received_payload_length_(0), |
ssrc_(0), |
@@ -90,8 +89,7 @@ RtpReceiverImpl::RtpReceiverImpl(int32_t id, |
RtpReceiverImpl::~RtpReceiverImpl() { |
for (int i = 0; i < num_csrcs_; ++i) { |
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i], |
- false); |
+ cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false); |
} |
} |
@@ -299,13 +297,14 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { |
if (new_ssrc) { |
// We need to get this to our RTCP sender and receiver. |
// We need to do this outside critical section. |
- cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header.ssrc); |
+ cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc); |
} |
if (re_initialize_decoder) { |
- if (-1 == cb_rtp_feedback_->OnInitializeDecoder( |
- id_, rtp_header.payloadType, payload_name, |
- rtp_header.payload_type_frequency, channels, rate)) { |
+ if (-1 == |
+ cb_rtp_feedback_->OnInitializeDecoder( |
+ rtp_header.payloadType, payload_name, |
+ rtp_header.payload_type_frequency, channels, rate)) { |
// New stream, same codec. |
LOG(LS_ERROR) << "Failed to create decoder for payload type: " |
<< static_cast<int>(rtp_header.payloadType); |
@@ -397,9 +396,9 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header, |
} // End critsect. |
if (re_initialize_decoder) { |
- if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder( |
- cb_rtp_feedback_, id_, payload_type, payload_name, |
- *specific_payload)) { |
+ if (-1 == |
+ rtp_media_receiver_->InvokeOnInitializeDecoder( |
+ cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) { |
return -1; // Wrong payload type. |
} |
} |
@@ -456,7 +455,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) { |
if (!found_match && csrc) { |
// Didn't find it, report it as new. |
have_called_callback = true; |
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true); |
+ cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true); |
} |
} |
// Search for old CSRC in new array. |
@@ -473,7 +472,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) { |
if (!found_match && csrc) { |
// Did not find it, report as removed. |
have_called_callback = true; |
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false); |
+ cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false); |
} |
} |
if (!have_called_callback) { |
@@ -481,9 +480,9 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) { |
// Using CSRC 0 to signal this event, not interop safe, other |
// implementations might have CSRC 0 as a valid value. |
if (num_csrcs_diff > 0) { |
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true); |
+ cb_rtp_feedback_->OnIncomingCSRCChanged(0, true); |
} else if (num_csrcs_diff < 0) { |
- cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false); |
+ cb_rtp_feedback_->OnIncomingCSRCChanged(0, false); |
} |
} |
} |