modules/rtp_rtcp/source/rtp_receiver_impl.cc - src - 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 "modules/rtp_rtcp/source/rtp_receiver_impl.h"

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

 #include <set>
 #include <vector>

 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/include/module_common_types.h"
 #include "modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "rtc_base/logging.h"

 namespace webrtc {

 namespace {
 bool InOrderPacket(absl::optional<uint16_t> latest_sequence_number,
                    uint16_t current_sequence_number) {
   if (!latest_sequence_number)
     return true;

   // We need to distinguish between a late or retransmitted packet,
   // and a sequence number discontinuity.
   if (IsNewerSequenceNumber(current_sequence_number, *latest_sequence_number)) {
     return true;
   } else {
     // If we have a restart of the remote side this packet is still in order.
     return !IsNewerSequenceNumber(
         current_sequence_number,
         *latest_sequence_number - kDefaultMaxReorderingThreshold);
   }
 }

 }  // namespace

 using RtpUtility::Payload;

 // Only return the sources in the last 10 seconds.
 const int64_t kGetSourcesTimeoutMs = 10000;

 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   RTC_DCHECK(incoming_payload_callback != nullptr);
   return new RtpReceiverImpl(
       clock, rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }

 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   RTC_DCHECK(incoming_payload_callback != nullptr);
   return new RtpReceiverImpl(
       clock, rtp_payload_registry,
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
 }

 int32_t RtpReceiver::RegisterReceivePayload(const CodecInst& audio_codec) {
   return RegisterReceivePayload(audio_codec.pltype,
                                 CodecInstToSdp(audio_codec));
 }

 RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
                                  RTPPayloadRegistry* rtp_payload_registry,
                                  RTPReceiverStrategy* rtp_media_receiver)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1) {

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

 RtpReceiverImpl::~RtpReceiverImpl() {}

 int32_t RtpReceiverImpl::RegisterReceivePayload(
     int payload_type,
     const SdpAudioFormat& audio_format) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);

   // 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_type, audio_format, &created_new_payload);
   if (created_new_payload) {
     if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_type,
                                                      audio_format) != 0) {
       RTC_LOG(LS_ERROR) << "Failed to register payload: " << audio_format.name
                         << "/" << payload_type;
       return -1;
     }
   }
   return result;
 }

 int32_t RtpReceiverImpl::RegisterReceivePayload(const VideoCodec& video_codec) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   return rtp_payload_registry_->RegisterReceivePayload(video_codec);
 }

 int32_t RtpReceiverImpl::DeRegisterReceivePayload(
     const int8_t payload_type) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
 }

 uint32_t RtpReceiverImpl::SSRC() const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   return ssrc_;
 }

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

   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) {
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);

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

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

   auto audio_level =
       rtp_header.extension.hasAudioLevel
           ? absl::optional<uint8_t>(rtp_header.extension.audioLevel)
           : absl::nullopt;
   UpdateSources(audio_level);

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

   if (ret_val < 0) {
     return false;
   }

   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);

     // TODO(nisse): Do not rely on InOrderPacket for recovered packets, when
     // packet is passed as RtpPacketReceived and that information is available.
     // We should ideally never record timestamps for retransmitted or recovered
     // packets.
     if (InOrderPacket(last_received_sequence_number_,
                       rtp_header.sequenceNumber)) {
       last_received_sequence_number_.emplace(rtp_header.sequenceNumber);
       last_received_timestamp_ = rtp_header.timestamp;
       last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
     }
   }

   return true;
 }

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

 std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);

   int64_t now_ms = clock_->TimeInMilliseconds();
   std::vector<RtpSource> sources;

   RTC_DCHECK(std::is_sorted(ssrc_sources_.begin(), ssrc_sources_.end(),
                             [](const RtpSource& lhs, const RtpSource& rhs) {
                               return lhs.timestamp_ms() < rhs.timestamp_ms();
                             }));
   RTC_DCHECK(std::is_sorted(csrc_sources_.begin(), csrc_sources_.end(),
                             [](const RtpSource& lhs, const RtpSource& rhs) {
                               return lhs.timestamp_ms() < rhs.timestamp_ms();
                             }));

   std::set<uint32_t> selected_ssrcs;
   for (auto rit = ssrc_sources_.rbegin(); rit != ssrc_sources_.rend(); ++rit) {
     if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
       break;
     }
     if (selected_ssrcs.insert(rit->source_id()).second) {
       sources.push_back(*rit);
     }
   }

   for (auto rit = csrc_sources_.rbegin(); rit != csrc_sources_.rend(); ++rit) {
     if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
       break;
     }
     sources.push_back(*rit);
   }
   return sources;
 }

 bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
                                           int64_t* receive_time_ms) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!last_received_sequence_number_)
     return false;

   *timestamp = last_received_timestamp_;
   *receive_time_ms = last_received_frame_time_ms_;

   return true;
 }

 // TODO(nisse): Delete.
 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   ssrc_ = rtp_header.ssrc;
 }

 // 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,
                                              PayloadUnion* specific_payload) {
   int8_t payload_type = rtp_header.payloadType;

   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);

     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) {
       bool should_discard_changes = false;

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

       if (should_discard_changes) {
         return 0;
       }

       const auto payload =
           rtp_payload_registry_->PayloadTypeToPayload(payload_type);
       if (!payload) {
         // Not a registered payload type.
         return -1;
       }
       rtp_payload_registry_->set_last_received_payload_type(payload_type);
     }
   }  // End critsect.

   return 0;
 }

 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   const uint8_t num_csrcs = rtp_header.header.numCSRCs;
   if (num_csrcs > kRtpCsrcSize) {
     // Ignore.
     return;
   }
   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);

     // Copy new.
     memcpy(current_remote_csrc_,
            rtp_header.header.arrOfCSRCs,
            num_csrcs * sizeof(uint32_t));

     num_csrcs_ = num_csrcs;
   }  // End critsect.
 }

 void RtpReceiverImpl::UpdateSources(
     const absl::optional<uint8_t>& ssrc_audio_level) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   int64_t now_ms = clock_->TimeInMilliseconds();

   for (size_t i = 0; i < num_csrcs_; ++i) {
     auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
     if (map_it == iterator_by_csrc_.end()) {
       // If it is a new CSRC, append a new object to the end of the list.
       csrc_sources_.emplace_back(now_ms, current_remote_csrc_[i],
                                  RtpSourceType::CSRC);
     } else {
       // If it is an existing CSRC, move the object to the end of the list.
       map_it->second->update_timestamp_ms(now_ms);
       csrc_sources_.splice(csrc_sources_.end(), csrc_sources_, map_it->second);
     }
     // Update the unordered_map.
     iterator_by_csrc_[current_remote_csrc_[i]] = std::prev(csrc_sources_.end());
   }

   // If this is the first packet or the SSRC is changed, insert a new
   // contributing source that uses the SSRC.
   if (ssrc_sources_.empty() || ssrc_sources_.rbegin()->source_id() != ssrc_) {
     ssrc_sources_.emplace_back(now_ms, ssrc_, RtpSourceType::SSRC);
   } else {
     ssrc_sources_.rbegin()->update_timestamp_ms(now_ms);
   }

   ssrc_sources_.back().set_audio_level(ssrc_audio_level);

   RemoveOutdatedSources(now_ms);
 }

 void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {
   std::list<RtpSource>::iterator it;
   for (it = csrc_sources_.begin(); it != csrc_sources_.end(); ++it) {
     if ((now_ms - it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
     iterator_by_csrc_.erase(it->source_id());
   }
   csrc_sources_.erase(csrc_sources_.begin(), it);

   std::vector<RtpSource>::iterator vec_it;
   for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
        ++vec_it) {
     if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
   }
   ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
 }

 }  // 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 "modules/rtp_rtcp/source/rtp_receiver_impl.h"

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

	#include <set>
	#include <vector>

	#include "common_types.h" // NOLINT(build/include)
	#include "modules/audio_coding/codecs/audio_format_conversion.h"
	#include "modules/include/module_common_types.h"
	#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
	#include "rtc_base/logging.h"

	namespace webrtc {

	namespace {
	bool InOrderPacket(absl::optional<uint16_t> latest_sequence_number,
	uint16_t current_sequence_number) {
	if (!latest_sequence_number)
	return true;

	// We need to distinguish between a late or retransmitted packet,
	// and a sequence number discontinuity.
	if (IsNewerSequenceNumber(current_sequence_number, *latest_sequence_number)) {
	return true;
	} else {
	// If we have a restart of the remote side this packet is still in order.
	return !IsNewerSequenceNumber(
	current_sequence_number,
	*latest_sequence_number - kDefaultMaxReorderingThreshold);
	}
	}

	} // namespace

	using RtpUtility::Payload;

	// Only return the sources in the last 10 seconds.
	const int64_t kGetSourcesTimeoutMs = 10000;

	RtpReceiver* RtpReceiver::CreateVideoReceiver(
	Clock* clock,
	RtpData* incoming_payload_callback,
	RTPPayloadRegistry* rtp_payload_registry) {
	RTC_DCHECK(incoming_payload_callback != nullptr);
	return new RtpReceiverImpl(
	clock, rtp_payload_registry,
	RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
	}

	RtpReceiver* RtpReceiver::CreateAudioReceiver(
	Clock* clock,
	RtpData* incoming_payload_callback,
	RTPPayloadRegistry* rtp_payload_registry) {
	RTC_DCHECK(incoming_payload_callback != nullptr);
	return new RtpReceiverImpl(
	clock, rtp_payload_registry,
	RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
	}

	int32_t RtpReceiver::RegisterReceivePayload(const CodecInst& audio_codec) {
	return RegisterReceivePayload(audio_codec.pltype,
	CodecInstToSdp(audio_codec));
	}

	RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
	RTPPayloadRegistry* rtp_payload_registry,
	RTPReceiverStrategy* rtp_media_receiver)
	: clock_(clock),
	rtp_payload_registry_(rtp_payload_registry),
	rtp_media_receiver_(rtp_media_receiver),
	ssrc_(0),
	num_csrcs_(0),
	current_remote_csrc_(),
	last_received_timestamp_(0),
	last_received_frame_time_ms_(-1) {

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

	RtpReceiverImpl::~RtpReceiverImpl() {}

	int32_t RtpReceiverImpl::RegisterReceivePayload(
	int payload_type,
	const SdpAudioFormat& audio_format) {
	rtc::CritScope lock(&critical_section_rtp_receiver_);

	// 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_type, audio_format, &created_new_payload);
	if (created_new_payload) {
	if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_type,
	audio_format) != 0) {
	RTC_LOG(LS_ERROR) << "Failed to register payload: " << audio_format.name
	<< "/" << payload_type;
	return -1;
	}
	}
	return result;
	}

	int32_t RtpReceiverImpl::RegisterReceivePayload(const VideoCodec& video_codec) {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	return rtp_payload_registry_->RegisterReceivePayload(video_codec);
	}

	int32_t RtpReceiverImpl::DeRegisterReceivePayload(
	const int8_t payload_type) {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
	}

	uint32_t RtpReceiverImpl::SSRC() const {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	return ssrc_;
	}

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

	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) {
	// Trigger our callbacks.
	CheckSSRCChanged(rtp_header);

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

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

	auto audio_level =
	rtp_header.extension.hasAudioLevel
	? absl::optional<uint8_t>(rtp_header.extension.audioLevel)
	: absl::nullopt;
	UpdateSources(audio_level);

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

	if (ret_val < 0) {
	return false;
	}

	{
	rtc::CritScope lock(&critical_section_rtp_receiver_);

	// TODO(nisse): Do not rely on InOrderPacket for recovered packets, when
	// packet is passed as RtpPacketReceived and that information is available.
	// We should ideally never record timestamps for retransmitted or recovered
	// packets.
	if (InOrderPacket(last_received_sequence_number_,
	rtp_header.sequenceNumber)) {
	last_received_sequence_number_.emplace(rtp_header.sequenceNumber);
	last_received_timestamp_ = rtp_header.timestamp;
	last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
	}
	}

	return true;
	}

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

	std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
	rtc::CritScope lock(&critical_section_rtp_receiver_);

	int64_t now_ms = clock_->TimeInMilliseconds();
	std::vector<RtpSource> sources;

	RTC_DCHECK(std::is_sorted(ssrc_sources_.begin(), ssrc_sources_.end(),
	[](const RtpSource& lhs, const RtpSource& rhs) {
	return lhs.timestamp_ms() < rhs.timestamp_ms();
	}));
	RTC_DCHECK(std::is_sorted(csrc_sources_.begin(), csrc_sources_.end(),
	[](const RtpSource& lhs, const RtpSource& rhs) {
	return lhs.timestamp_ms() < rhs.timestamp_ms();
	}));

	std::set<uint32_t> selected_ssrcs;
	for (auto rit = ssrc_sources_.rbegin(); rit != ssrc_sources_.rend(); ++rit) {
	if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
	break;
	}
	if (selected_ssrcs.insert(rit->source_id()).second) {
	sources.push_back(*rit);
	}
	}

	for (auto rit = csrc_sources_.rbegin(); rit != csrc_sources_.rend(); ++rit) {
	if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
	break;
	}
	sources.push_back(*rit);
	}
	return sources;
	}

	bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
	int64_t* receive_time_ms) const {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	if (!last_received_sequence_number_)
	return false;

	*timestamp = last_received_timestamp_;
	*receive_time_ms = last_received_frame_time_ms_;

	return true;
	}

	// TODO(nisse): Delete.
	// Implementation note: must not hold critsect when called.
	void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	ssrc_ = rtp_header.ssrc;
	}

	// 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,
	PayloadUnion* specific_payload) {
	int8_t payload_type = rtp_header.payloadType;

	{
	rtc::CritScope lock(&critical_section_rtp_receiver_);

	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) {
	bool should_discard_changes = false;

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

	if (should_discard_changes) {
	return 0;
	}

	const auto payload =
	rtp_payload_registry_->PayloadTypeToPayload(payload_type);
	if (!payload) {
	// Not a registered payload type.
	return -1;
	}
	rtp_payload_registry_->set_last_received_payload_type(payload_type);
	}
	} // End critsect.

	return 0;
	}

	// Implementation note: must not hold critsect when called.
	void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
	const uint8_t num_csrcs = rtp_header.header.numCSRCs;
	if (num_csrcs > kRtpCsrcSize) {
	// Ignore.
	return;
	}
	{
	rtc::CritScope lock(&critical_section_rtp_receiver_);

	// Copy new.
	memcpy(current_remote_csrc_,
	rtp_header.header.arrOfCSRCs,
	num_csrcs * sizeof(uint32_t));

	num_csrcs_ = num_csrcs;
	} // End critsect.
	}

	void RtpReceiverImpl::UpdateSources(
	const absl::optional<uint8_t>& ssrc_audio_level) {
	rtc::CritScope lock(&critical_section_rtp_receiver_);
	int64_t now_ms = clock_->TimeInMilliseconds();

	for (size_t i = 0; i < num_csrcs_; ++i) {
	auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
	if (map_it == iterator_by_csrc_.end()) {
	// If it is a new CSRC, append a new object to the end of the list.
	csrc_sources_.emplace_back(now_ms, current_remote_csrc_[i],
	RtpSourceType::CSRC);
	} else {
	// If it is an existing CSRC, move the object to the end of the list.
	map_it->second->update_timestamp_ms(now_ms);
	csrc_sources_.splice(csrc_sources_.end(), csrc_sources_, map_it->second);
	}
	// Update the unordered_map.
	iterator_by_csrc_[current_remote_csrc_[i]] = std::prev(csrc_sources_.end());
	}

	// If this is the first packet or the SSRC is changed, insert a new
	// contributing source that uses the SSRC.
	if (ssrc_sources_.empty() \|\| ssrc_sources_.rbegin()->source_id() != ssrc_) {
	ssrc_sources_.emplace_back(now_ms, ssrc_, RtpSourceType::SSRC);
	} else {
	ssrc_sources_.rbegin()->update_timestamp_ms(now_ms);
	}

	ssrc_sources_.back().set_audio_level(ssrc_audio_level);

	RemoveOutdatedSources(now_ms);
	}

	void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {
	std::list<RtpSource>::iterator it;
	for (it = csrc_sources_.begin(); it != csrc_sources_.end(); ++it) {
	if ((now_ms - it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
	break;
	}
	iterator_by_csrc_.erase(it->source_id());
	}
	csrc_sources_.erase(csrc_sources_.begin(), it);

	std::vector<RtpSource>::iterator vec_it;
	for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
	++vec_it) {
	if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
	break;
	}
	}
	ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
	}

	} // namespace webrtc