| 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);
|
| }
|
| }
|
| }
|
|
|