modules/rtp_rtcp/source/rtp_receiver_impl.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"

 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>

 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/system_wrappers/interface/logging.h"

 namespace webrtc {

 using RtpUtility::GetCurrentRTP;
 using RtpUtility::Payload;
 using RtpUtility::StringCompare;

 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     int id, Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
       rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }

 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     int id, Clock* clock,
     RtpAudioFeedback* incoming_audio_feedback,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_audio_feedback)
     incoming_audio_feedback = NullObjectRtpAudioFeedback();
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       id, clock, incoming_audio_feedback, incoming_messages_callback,
       rtp_payload_registry,
       RTPReceiverStrategy::CreateAudioStrategy(id, 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)
     : 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()),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1),
       last_received_sequence_number_(0),
       nack_method_(kNackOff) {
   assert(incoming_audio_messages_callback);
   assert(incoming_messages_callback);

   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
 }

 RtpReceiverImpl::~RtpReceiverImpl() {
   for (int i = 0; i < num_csrcs_; ++i) {
     cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
                                             false);
   }
 }

 int32_t RtpReceiverImpl::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
     const uint8_t channels,
     const uint32_t rate) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

   // TODO(phoglund): Try to streamline handling of the RED codec and some other
   // cases which makes it necessary to keep track of whether we created a
   // payload or not.
   bool created_new_payload = false;
   int32_t result = rtp_payload_registry_->RegisterReceivePayload(
       payload_name, payload_type, frequency, channels, rate,
       &created_new_payload);
   if (created_new_payload) {
     if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
                                                      frequency) != 0) {
       LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
                  << payload_type;
       return -1;
     }
   }
   return result;
 }

 int32_t RtpReceiverImpl::DeRegisterReceivePayload(
     const int8_t payload_type) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
 }

 NACKMethod RtpReceiverImpl::NACK() const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   return nack_method_;
 }

 // Turn negative acknowledgment requests on/off.
 void RtpReceiverImpl::SetNACKStatus(const NACKMethod method) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   nack_method_ = method;
 }

 uint32_t RtpReceiverImpl::SSRC() const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   return ssrc_;
 }

 // Get remote CSRC.
 int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

   assert(num_csrcs_ <= kRtpCsrcSize);

   if (num_csrcs_ > 0) {
     memcpy(array_of_csrcs, current_remote_csrc_, sizeof(uint32_t)*num_csrcs_);
   }
   return num_csrcs_;
 }

 int32_t RtpReceiverImpl::Energy(
     uint8_t array_of_energy[kRtpCsrcSize]) const {
   return rtp_media_receiver_->Energy(array_of_energy);
 }

 bool RtpReceiverImpl::IncomingRtpPacket(
   const RTPHeader& rtp_header,
   const uint8_t* payload,
   size_t payload_length,
   PayloadUnion payload_specific,
   bool in_order) {
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);

   int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
   bool should_reset_statistics = false;

   if (CheckPayloadChanged(rtp_header,
                           first_payload_byte,
                           is_red,
                           &payload_specific,
                           &should_reset_statistics) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
       return true;
     }
     LOG(LS_WARNING) << "Receiving invalid payload type.";
     return false;
   }

   if (should_reset_statistics) {
     cb_rtp_feedback_->ResetStatistics(ssrc_);
   }

   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);

   size_t payload_data_length = payload_length - rtp_header.paddingLength;

   bool is_first_packet_in_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
     if (HaveReceivedFrame()) {
       is_first_packet_in_frame =
           last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
           last_received_timestamp_ != rtp_header.timestamp;
     } else {
       is_first_packet_in_frame = true;
     }
   }

   int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
       &webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
       clock_->TimeInMilliseconds(), is_first_packet_in_frame);

   if (ret_val < 0) {
     return false;
   }

   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

     last_receive_time_ = clock_->TimeInMilliseconds();
     last_received_payload_length_ = payload_data_length;

     if (in_order) {
       if (last_received_timestamp_ != rtp_header.timestamp) {
         last_received_timestamp_ = rtp_header.timestamp;
         last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
       }
       last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
   return true;
 }

 TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }

 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   if (!HaveReceivedFrame())
     return false;
   *timestamp = last_received_timestamp_;
   return true;
 }

 bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   if (!HaveReceivedFrame())
     return false;
   *receive_time_ms = last_received_frame_time_ms_;
   return true;
 }

 bool RtpReceiverImpl::HaveReceivedFrame() const {
   return last_received_frame_time_ms_ >= 0;
 }

 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   bool new_ssrc = false;
   bool re_initialize_decoder = false;
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
   uint8_t channels = 1;
   uint32_t rate = 0;

   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
     if (ssrc_ != rtp_header.ssrc ||
         (last_received_payload_type == -1 && ssrc_ == 0)) {
       // We need the payload_type_ to make the call if the remote SSRC is 0.
       new_ssrc = true;

       cb_rtp_feedback_->ResetStatistics(ssrc_);

       last_received_timestamp_ = 0;
       last_received_sequence_number_ = 0;
       last_received_frame_time_ms_ = -1;

       // Do we have a SSRC? Then the stream is restarted.
       if (ssrc_ != 0) {
         // Do we have the same codec? Then re-initialize coder.
         if (rtp_header.payloadType == last_received_payload_type) {
           re_initialize_decoder = true;

           Payload* payload;
           if (!rtp_payload_registry_->PayloadTypeToPayload(
               rtp_header.payloadType, payload)) {
             return;
           }
           assert(payload);
           payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
           strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
           if (payload->audio) {
             channels = payload->typeSpecific.Audio.channels;
             rate = payload->typeSpecific.Audio.rate;
           }
         }
       }
       ssrc_ = rtp_header.ssrc;
     }
   }

   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);
   }

   if (re_initialize_decoder) {
     if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
         id_, 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: "
                     << rtp_header.payloadType;
     }
   }
 }

 // Implementation note: must not hold critsect when called.
 // TODO(phoglund): Move as much as possible of this code path into the media
 // specific receivers. Basically this method goes through a lot of trouble to
 // compute something which is only used by the media specific parts later. If
 // this code path moves we can get rid of some of the rtp_receiver ->
 // media_specific interface (such as CheckPayloadChange, possibly get/set
 // last known payload).
 int32_t RtpReceiverImpl::CheckPayloadChanged(
   const RTPHeader& rtp_header,
   const int8_t first_payload_byte,
   bool& is_red,
   PayloadUnion* specific_payload,
   bool* should_reset_statistics) {
   bool re_initialize_decoder = false;

   char payload_name[RTP_PAYLOAD_NAME_SIZE];
   int8_t payload_type = rtp_header.payloadType;

   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
     // TODO(holmer): Remove this code when RED parsing has been broken out from
     // RtpReceiverAudio.
     if (payload_type != last_received_payload_type) {
       if (rtp_payload_registry_->red_payload_type() == payload_type) {
         // Get the real codec payload type.
         payload_type = first_payload_byte & 0x7f;
         is_red = true;

         if (rtp_payload_registry_->red_payload_type() == payload_type) {
           // Invalid payload type, traced by caller. If we proceeded here,
           // this would be set as |_last_received_payload_type|, and we would no
           // longer catch corrupt packets at this level.
           return -1;
         }

         // When we receive RED we need to check the real payload type.
         if (payload_type == last_received_payload_type) {
           rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
           return 0;
         }
       }
       *should_reset_statistics = false;
       bool should_discard_changes = false;

       rtp_media_receiver_->CheckPayloadChanged(
         payload_type, specific_payload, should_reset_statistics,
         &should_discard_changes);

       if (should_discard_changes) {
         is_red = false;
         return 0;
       }

       Payload* payload;
       if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
         // Not a registered payload type.
         return -1;
       }
       assert(payload);
       payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
       strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);

       rtp_payload_registry_->set_last_received_payload_type(payload_type);

       re_initialize_decoder = true;

       rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);

       if (!payload->audio) {
         bool media_type_unchanged =
             rtp_payload_registry_->ReportMediaPayloadType(payload_type);
         if (media_type_unchanged) {
           // Only reset the decoder if the media codec type has changed.
           re_initialize_decoder = false;
         }
       }
       if (re_initialize_decoder) {
         *should_reset_statistics = true;
       }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
       is_red = false;
     }
   }  // End critsect.

   if (re_initialize_decoder) {
     if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
         cb_rtp_feedback_, id_, payload_type, payload_name,
         *specific_payload)) {
       return -1;  // Wrong payload type.
     }
   }
   return 0;
 }

 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   int32_t num_csrcs_diff = 0;
   uint32_t old_remote_csrc[kRtpCsrcSize];
   uint8_t old_num_csrcs = 0;

   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

     if (!rtp_media_receiver_->ShouldReportCsrcChanges(
         rtp_header.header.payloadType)) {
       return;
     }
     old_num_csrcs  = num_csrcs_;
     if (old_num_csrcs > 0) {
       // Make a copy of old.
       memcpy(old_remote_csrc, current_remote_csrc_,
              num_csrcs_ * sizeof(uint32_t));
     }
     const uint8_t num_csrcs = rtp_header.header.numCSRCs;
     if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
       // Copy new.
       memcpy(current_remote_csrc_,
              rtp_header.header.arrOfCSRCs,
              num_csrcs * sizeof(uint32_t));
     }
     if (num_csrcs > 0 || old_num_csrcs > 0) {
       num_csrcs_diff = num_csrcs - old_num_csrcs;
       num_csrcs_ = num_csrcs;  // Update stored CSRCs.
     } else {
       // No change.
       return;
     }
   }  // End critsect.

   bool have_called_callback = false;
   // Search for new CSRC in old array.
   for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
     const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];

     bool found_match = false;
     for (uint8_t j = 0; j < old_num_csrcs; ++j) {
       if (csrc == old_remote_csrc[j]) {  // old list
         found_match = true;
         break;
       }
     }
     if (!found_match && csrc) {
       // Didn't find it, report it as new.
       have_called_callback = true;
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true);
     }
   }
   // Search for old CSRC in new array.
   for (uint8_t i = 0; i < old_num_csrcs; ++i) {
     const uint32_t csrc = old_remote_csrc[i];

     bool found_match = false;
     for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
       if (csrc == rtp_header.header.arrOfCSRCs[j]) {
         found_match = true;
         break;
       }
     }
     if (!found_match && csrc) {
       // Did not find it, report as removed.
       have_called_callback = true;
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false);
     }
   }
   if (!have_called_callback) {
     // If the CSRC list contain non-unique entries we will end up here.
     // 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);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
     }
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"

	#include <assert.h>
	#include <math.h>
	#include <stdlib.h>
	#include <string.h>

	#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
	#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
	#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
	#include "webrtc/system_wrappers/interface/logging.h"

	namespace webrtc {

	using RtpUtility::GetCurrentRTP;
	using RtpUtility::Payload;
	using RtpUtility::StringCompare;

	RtpReceiver* RtpReceiver::CreateVideoReceiver(
	int id, Clock* clock,
	RtpData* incoming_payload_callback,
	RtpFeedback* incoming_messages_callback,
	RTPPayloadRegistry* rtp_payload_registry) {
	if (!incoming_payload_callback)
	incoming_payload_callback = NullObjectRtpData();
	if (!incoming_messages_callback)
	incoming_messages_callback = NullObjectRtpFeedback();
	return new RtpReceiverImpl(
	id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
	rtp_payload_registry,
	RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
	}

	RtpReceiver* RtpReceiver::CreateAudioReceiver(
	int id, Clock* clock,
	RtpAudioFeedback* incoming_audio_feedback,
	RtpData* incoming_payload_callback,
	RtpFeedback* incoming_messages_callback,
	RTPPayloadRegistry* rtp_payload_registry) {
	if (!incoming_audio_feedback)
	incoming_audio_feedback = NullObjectRtpAudioFeedback();
	if (!incoming_payload_callback)
	incoming_payload_callback = NullObjectRtpData();
	if (!incoming_messages_callback)
	incoming_messages_callback = NullObjectRtpFeedback();
	return new RtpReceiverImpl(
	id, clock, incoming_audio_feedback, incoming_messages_callback,
	rtp_payload_registry,
	RTPReceiverStrategy::CreateAudioStrategy(id, 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)
	: 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()),
	last_receive_time_(0),
	last_received_payload_length_(0),
	ssrc_(0),
	num_csrcs_(0),
	current_remote_csrc_(),
	last_received_timestamp_(0),
	last_received_frame_time_ms_(-1),
	last_received_sequence_number_(0),
	nack_method_(kNackOff) {
	assert(incoming_audio_messages_callback);
	assert(incoming_messages_callback);

	memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
	}

	RtpReceiverImpl::~RtpReceiverImpl() {
	for (int i = 0; i < num_csrcs_; ++i) {
	cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
	false);
	}
	}

	int32_t RtpReceiverImpl::RegisterReceivePayload(
	const char payload_name[RTP_PAYLOAD_NAME_SIZE],
	const int8_t payload_type,
	const uint32_t frequency,
	const uint8_t channels,
	const uint32_t rate) {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	// TODO(phoglund): Try to streamline handling of the RED codec and some other
	// cases which makes it necessary to keep track of whether we created a
	// payload or not.
	bool created_new_payload = false;
	int32_t result = rtp_payload_registry_->RegisterReceivePayload(
	payload_name, payload_type, frequency, channels, rate,
	&created_new_payload);
	if (created_new_payload) {
	if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
	frequency) != 0) {
	LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
	<< payload_type;
	return -1;
	}
	}
	return result;
	}

	int32_t RtpReceiverImpl::DeRegisterReceivePayload(
	const int8_t payload_type) {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
	}

	NACKMethod RtpReceiverImpl::NACK() const {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	return nack_method_;
	}

	// Turn negative acknowledgment requests on/off.
	void RtpReceiverImpl::SetNACKStatus(const NACKMethod method) {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	nack_method_ = method;
	}

	uint32_t RtpReceiverImpl::SSRC() const {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	return ssrc_;
	}

	// Get remote CSRC.
	int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	assert(num_csrcs_ <= kRtpCsrcSize);

	if (num_csrcs_ > 0) {
	memcpy(array_of_csrcs, current_remote_csrc_, sizeof(uint32_t)*num_csrcs_);
	}
	return num_csrcs_;
	}

	int32_t RtpReceiverImpl::Energy(
	uint8_t array_of_energy[kRtpCsrcSize]) const {
	return rtp_media_receiver_->Energy(array_of_energy);
	}

	bool RtpReceiverImpl::IncomingRtpPacket(
	const RTPHeader& rtp_header,
	const uint8_t* payload,
	size_t payload_length,
	PayloadUnion payload_specific,
	bool in_order) {
	// Trigger our callbacks.
	CheckSSRCChanged(rtp_header);

	int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
	bool is_red = false;
	bool should_reset_statistics = false;

	if (CheckPayloadChanged(rtp_header,
	first_payload_byte,
	is_red,
	&payload_specific,
	&should_reset_statistics) == -1) {
	if (payload_length == 0) {
	// OK, keep-alive packet.
	return true;
	}
	LOG(LS_WARNING) << "Receiving invalid payload type.";
	return false;
	}

	if (should_reset_statistics) {
	cb_rtp_feedback_->ResetStatistics(ssrc_);
	}

	WebRtcRTPHeader webrtc_rtp_header;
	memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
	webrtc_rtp_header.header = rtp_header;
	CheckCSRC(webrtc_rtp_header);

	size_t payload_data_length = payload_length - rtp_header.paddingLength;

	bool is_first_packet_in_frame = false;
	{
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	if (HaveReceivedFrame()) {
	is_first_packet_in_frame =
	last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
	last_received_timestamp_ != rtp_header.timestamp;
	} else {
	is_first_packet_in_frame = true;
	}
	}

	int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
	&webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
	clock_->TimeInMilliseconds(), is_first_packet_in_frame);

	if (ret_val < 0) {
	return false;
	}

	{
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	last_receive_time_ = clock_->TimeInMilliseconds();
	last_received_payload_length_ = payload_data_length;

	if (in_order) {
	if (last_received_timestamp_ != rtp_header.timestamp) {
	last_received_timestamp_ = rtp_header.timestamp;
	last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
	}
	last_received_sequence_number_ = rtp_header.sequenceNumber;
	}
	}
	return true;
	}

	TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
	return rtp_media_receiver_->GetTelephoneEventHandler();
	}

	bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	if (!HaveReceivedFrame())
	return false;
	*timestamp = last_received_timestamp_;
	return true;
	}

	bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
	if (!HaveReceivedFrame())
	return false;
	*receive_time_ms = last_received_frame_time_ms_;
	return true;
	}

	bool RtpReceiverImpl::HaveReceivedFrame() const {
	return last_received_frame_time_ms_ >= 0;
	}

	// Implementation note: must not hold critsect when called.
	void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
	bool new_ssrc = false;
	bool re_initialize_decoder = false;
	char payload_name[RTP_PAYLOAD_NAME_SIZE];
	uint8_t channels = 1;
	uint32_t rate = 0;

	{
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	int8_t last_received_payload_type =
	rtp_payload_registry_->last_received_payload_type();
	if (ssrc_ != rtp_header.ssrc \|\|
	(last_received_payload_type == -1 && ssrc_ == 0)) {
	// We need the payload_type_ to make the call if the remote SSRC is 0.
	new_ssrc = true;

	cb_rtp_feedback_->ResetStatistics(ssrc_);

	last_received_timestamp_ = 0;
	last_received_sequence_number_ = 0;
	last_received_frame_time_ms_ = -1;

	// Do we have a SSRC? Then the stream is restarted.
	if (ssrc_ != 0) {
	// Do we have the same codec? Then re-initialize coder.
	if (rtp_header.payloadType == last_received_payload_type) {
	re_initialize_decoder = true;

	Payload* payload;
	if (!rtp_payload_registry_->PayloadTypeToPayload(
	rtp_header.payloadType, payload)) {
	return;
	}
	assert(payload);
	payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
	strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
	if (payload->audio) {
	channels = payload->typeSpecific.Audio.channels;
	rate = payload->typeSpecific.Audio.rate;
	}
	}
	}
	ssrc_ = rtp_header.ssrc;
	}
	}

	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);
	}

	if (re_initialize_decoder) {
	if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
	id_, 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: "
	<< rtp_header.payloadType;
	}
	}
	}

	// Implementation note: must not hold critsect when called.
	// TODO(phoglund): Move as much as possible of this code path into the media
	// specific receivers. Basically this method goes through a lot of trouble to
	// compute something which is only used by the media specific parts later. If
	// this code path moves we can get rid of some of the rtp_receiver ->
	// media_specific interface (such as CheckPayloadChange, possibly get/set
	// last known payload).
	int32_t RtpReceiverImpl::CheckPayloadChanged(
	const RTPHeader& rtp_header,
	const int8_t first_payload_byte,
	bool& is_red,
	PayloadUnion* specific_payload,
	bool* should_reset_statistics) {
	bool re_initialize_decoder = false;

	char payload_name[RTP_PAYLOAD_NAME_SIZE];
	int8_t payload_type = rtp_header.payloadType;

	{
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	int8_t last_received_payload_type =
	rtp_payload_registry_->last_received_payload_type();
	// TODO(holmer): Remove this code when RED parsing has been broken out from
	// RtpReceiverAudio.
	if (payload_type != last_received_payload_type) {
	if (rtp_payload_registry_->red_payload_type() == payload_type) {
	// Get the real codec payload type.
	payload_type = first_payload_byte & 0x7f;
	is_red = true;

	if (rtp_payload_registry_->red_payload_type() == payload_type) {
	// Invalid payload type, traced by caller. If we proceeded here,
	// this would be set as \|_last_received_payload_type\|, and we would no
	// longer catch corrupt packets at this level.
	return -1;
	}

	// When we receive RED we need to check the real payload type.
	if (payload_type == last_received_payload_type) {
	rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
	return 0;
	}
	}
	*should_reset_statistics = false;
	bool should_discard_changes = false;

	rtp_media_receiver_->CheckPayloadChanged(
	payload_type, specific_payload, should_reset_statistics,
	&should_discard_changes);

	if (should_discard_changes) {
	is_red = false;
	return 0;
	}

	Payload* payload;
	if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
	// Not a registered payload type.
	return -1;
	}
	assert(payload);
	payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
	strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);

	rtp_payload_registry_->set_last_received_payload_type(payload_type);

	re_initialize_decoder = true;

	rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
	rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);

	if (!payload->audio) {
	bool media_type_unchanged =
	rtp_payload_registry_->ReportMediaPayloadType(payload_type);
	if (media_type_unchanged) {
	// Only reset the decoder if the media codec type has changed.
	re_initialize_decoder = false;
	}
	}
	if (re_initialize_decoder) {
	*should_reset_statistics = true;
	}
	} else {
	rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
	is_red = false;
	}
	} // End critsect.

	if (re_initialize_decoder) {
	if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
	cb_rtp_feedback_, id_, payload_type, payload_name,
	*specific_payload)) {
	return -1; // Wrong payload type.
	}
	}
	return 0;
	}

	// Implementation note: must not hold critsect when called.
	void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
	int32_t num_csrcs_diff = 0;
	uint32_t old_remote_csrc[kRtpCsrcSize];
	uint8_t old_num_csrcs = 0;

	{
	CriticalSectionScoped lock(critical_section_rtp_receiver_.get());

	if (!rtp_media_receiver_->ShouldReportCsrcChanges(
	rtp_header.header.payloadType)) {
	return;
	}
	old_num_csrcs = num_csrcs_;
	if (old_num_csrcs > 0) {
	// Make a copy of old.
	memcpy(old_remote_csrc, current_remote_csrc_,
	num_csrcs_ * sizeof(uint32_t));
	}
	const uint8_t num_csrcs = rtp_header.header.numCSRCs;
	if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
	// Copy new.
	memcpy(current_remote_csrc_,
	rtp_header.header.arrOfCSRCs,
	num_csrcs * sizeof(uint32_t));
	}
	if (num_csrcs > 0 \|\| old_num_csrcs > 0) {
	num_csrcs_diff = num_csrcs - old_num_csrcs;
	num_csrcs_ = num_csrcs; // Update stored CSRCs.
	} else {
	// No change.
	return;
	}
	} // End critsect.

	bool have_called_callback = false;
	// Search for new CSRC in old array.
	for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
	const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];

	bool found_match = false;
	for (uint8_t j = 0; j < old_num_csrcs; ++j) {
	if (csrc == old_remote_csrc[j]) { // old list
	found_match = true;
	break;
	}
	}
	if (!found_match && csrc) {
	// Didn't find it, report it as new.
	have_called_callback = true;
	cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true);
	}
	}
	// Search for old CSRC in new array.
	for (uint8_t i = 0; i < old_num_csrcs; ++i) {
	const uint32_t csrc = old_remote_csrc[i];

	bool found_match = false;
	for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
	if (csrc == rtp_header.header.arrOfCSRCs[j]) {
	found_match = true;
	break;
	}
	}
	if (!found_match && csrc) {
	// Did not find it, report as removed.
	have_called_callback = true;
	cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false);
	}
	}
	if (!have_called_callback) {
	// If the CSRC list contain non-unique entries we will end up here.
	// 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);
	} else if (num_csrcs_diff < 0) {
	cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
	}
	}
	}

	} // namespace webrtc