modules/rtp_rtcp/source/rtp_payload_registry.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_payload_registry.h"

 #include "webrtc/system_wrappers/interface/trace.h"

 namespace webrtc {

 RTPPayloadRegistry::RTPPayloadRegistry(
     const WebRtc_Word32 id)
     : id_(id),
       rtp_media_receiver_(NULL),
       red_payload_type_(-1),
       last_received_payload_type_(-1),
       last_received_media_payload_type_(-1) {
 }

 RTPPayloadRegistry::~RTPPayloadRegistry() {
   while (!payload_type_map_.empty()) {
     ModuleRTPUtility::PayloadTypeMap::iterator it = payload_type_map_.begin();
     delete it->second;
     payload_type_map_.erase(it);
   }
 }

 WebRtc_Word32 RTPPayloadRegistry::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const WebRtc_Word8 payload_type,
     const WebRtc_UWord32 frequency,
     const WebRtc_UWord8 channels,
     const WebRtc_UWord32 rate) {
   assert(rtp_media_receiver_);
   assert(payload_name);

   // Sanity check.
   switch (payload_type) {
     // Reserved payload types to avoid RTCP conflicts when marker bit is set.
     case 64:        //  192 Full INTRA-frame request.
     case 72:        //  200 Sender report.
     case 73:        //  201 Receiver report.
     case 74:        //  202 Source description.
     case 75:        //  203 Goodbye.
     case 76:        //  204 Application-defined.
     case 77:        //  205 Transport layer FB message.
     case 78:        //  206 Payload-specific FB message.
     case 79:        //  207 Extended report.
       WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
                    "%s invalid payloadtype:%d",
                    __FUNCTION__, payload_type);
       return -1;
     default:
       break;
   }
   size_t payload_name_length = strlen(payload_name);

   ModuleRTPUtility::PayloadTypeMap::iterator it =
     payload_type_map_.find(payload_type);

   if (it != payload_type_map_.end()) {
     // We already use this payload type.
     ModuleRTPUtility::Payload* payload = it->second;
     assert(payload);

     size_t name_length = strlen(payload->name);

     // Check if it's the same as we already have.
     // If same, ignore sending an error.
     if (payload_name_length == name_length &&
         ModuleRTPUtility::StringCompare(
             payload->name, payload_name, payload_name_length)) {
       if (rtp_media_receiver_->PayloadIsCompatible(*payload, frequency,
                                                    channels, rate)) {
         rtp_media_receiver_->UpdatePayloadRate(payload, rate);
         return 0;
       }
     }
     WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
                  "%s invalid argument payload_type:%d already registered",
                  __FUNCTION__, payload_type);
     return -1;
   }

   rtp_media_receiver_->PossiblyRemoveExistingPayloadType(
     &payload_type_map_, payload_name, payload_name_length, frequency, channels,
     rate);

   ModuleRTPUtility::Payload* payload = NULL;

   // Save the RED payload type. Used in both audio and video.
   if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) {
     red_payload_type_ = payload_type;
     payload = new ModuleRTPUtility::Payload;
     payload->audio = false;
     payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
     strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
   } else {
     payload = rtp_media_receiver_->CreatePayloadType(
         payload_name, payload_type, frequency, channels, rate);
   }
   if (payload == NULL) {
     WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
                  "%s failed to register payload",
                  __FUNCTION__);
     return -1;
   }
   payload_type_map_[payload_type] = payload;

   // Successful set of payload type, clear the value of last received payload
   // type since it might mean something else.
   last_received_payload_type_ = -1;
   last_received_media_payload_type_ = -1;
   return 0;
 }

 WebRtc_Word32 RTPPayloadRegistry::DeRegisterReceivePayload(
     const WebRtc_Word8 payload_type) {
   assert(rtp_media_receiver_);
   ModuleRTPUtility::PayloadTypeMap::iterator it =
     payload_type_map_.find(payload_type);

   if (it == payload_type_map_.end()) {
     WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
                  "%s failed to find payload_type:%d",
                  __FUNCTION__, payload_type);
     return -1;
   }
   delete it->second;
   payload_type_map_.erase(it);
   return 0;
 }

 WebRtc_Word32 RTPPayloadRegistry::ReceivePayloadType(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const WebRtc_UWord32 frequency,
     const WebRtc_UWord8 channels,
     const WebRtc_UWord32 rate,
     WebRtc_Word8* payload_type) const {
   if (payload_type == NULL) {
     WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
                  "%s invalid argument", __FUNCTION__);
     return -1;
   }
   size_t payload_name_length = strlen(payload_name);

   ModuleRTPUtility::PayloadTypeMap::const_iterator it =
       payload_type_map_.begin();

   while (it != payload_type_map_.end()) {
     ModuleRTPUtility::Payload* payload = it->second;
     assert(payload);

     size_t name_length = strlen(payload->name);
     if (payload_name_length == name_length &&
         ModuleRTPUtility::StringCompare(
             payload->name, payload_name, payload_name_length)) {
       // Name matches.
       if (payload->audio) {
         if (rate == 0) {
           // [default] audio, check freq and channels.
           if (payload->typeSpecific.Audio.frequency == frequency &&
               payload->typeSpecific.Audio.channels == channels) {
             *payload_type = it->first;
             return 0;
           }
         } else {
           // Non-default audio, check freq, channels and rate.
           if (payload->typeSpecific.Audio.frequency == frequency &&
               payload->typeSpecific.Audio.channels == channels &&
               payload->typeSpecific.Audio.rate == rate) {
             // extra rate condition added
             *payload_type = it->first;
             return 0;
           }
         }
       } else {
         // Video.
         *payload_type = it->first;
         return 0;
       }
     }
     it++;
   }
   return -1;
 }

 WebRtc_Word32 RTPPayloadRegistry::ReceivePayload(
     const WebRtc_Word8 payload_type,
     char payload_name[RTP_PAYLOAD_NAME_SIZE],
     WebRtc_UWord32* frequency,
     WebRtc_UWord8* channels,
     WebRtc_UWord32* rate) const {
   assert(rtp_media_receiver_);
   ModuleRTPUtility::PayloadTypeMap::const_iterator it =
     payload_type_map_.find(payload_type);

   if (it == payload_type_map_.end()) {
     return -1;
   }
   ModuleRTPUtility::Payload* payload = it->second;
   assert(payload);

   if (frequency) {
     if (payload->audio) {
       *frequency = payload->typeSpecific.Audio.frequency;
     } else {
       *frequency = kDefaultVideoFrequency;
     }
   }
   if (channels) {
     if (payload->audio) {
       *channels = payload->typeSpecific.Audio.channels;
     } else {
       *channels = 1;
     }
   }
   if (rate) {
     if (payload->audio) {
       *rate = payload->typeSpecific.Audio.rate;
     } else {
       assert(false);
       *rate = 0;
     }
   }
   if (payload_name) {
     payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
     strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
   }
   return 0;
 }

 WebRtc_UWord32 RTPPayloadRegistry::PayloadTypeToPayload(
   const WebRtc_UWord8 payload_type,
   ModuleRTPUtility::Payload*& payload) const {
   assert(rtp_media_receiver_);

   ModuleRTPUtility::PayloadTypeMap::const_iterator it =
     payload_type_map_.find(payload_type);

   // Check that this is a registered payload type.
   if (it == payload_type_map_.end()) {
     return -1;
   }
   payload = it->second;
   return 0;
 }

 bool RTPPayloadRegistry::ReportMediaPayloadType(
     WebRtc_UWord8 media_payload_type) {
   if (last_received_media_payload_type_ == media_payload_type) {
     // Media type unchanged.
     return true;
   }
   last_received_media_payload_type_ = media_payload_type;
   return 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_payload_registry.h"

	#include "webrtc/system_wrappers/interface/trace.h"

	namespace webrtc {

	RTPPayloadRegistry::RTPPayloadRegistry(
	const WebRtc_Word32 id)
	: id_(id),
	rtp_media_receiver_(NULL),
	red_payload_type_(-1),
	last_received_payload_type_(-1),
	last_received_media_payload_type_(-1) {
	}

	RTPPayloadRegistry::~RTPPayloadRegistry() {
	while (!payload_type_map_.empty()) {
	ModuleRTPUtility::PayloadTypeMap::iterator it = payload_type_map_.begin();
	delete it->second;
	payload_type_map_.erase(it);
	}
	}

	WebRtc_Word32 RTPPayloadRegistry::RegisterReceivePayload(
	const char payload_name[RTP_PAYLOAD_NAME_SIZE],
	const WebRtc_Word8 payload_type,
	const WebRtc_UWord32 frequency,
	const WebRtc_UWord8 channels,
	const WebRtc_UWord32 rate) {
	assert(rtp_media_receiver_);
	assert(payload_name);

	// Sanity check.
	switch (payload_type) {
	// Reserved payload types to avoid RTCP conflicts when marker bit is set.
	case 64: // 192 Full INTRA-frame request.
	case 72: // 200 Sender report.
	case 73: // 201 Receiver report.
	case 74: // 202 Source description.
	case 75: // 203 Goodbye.
	case 76: // 204 Application-defined.
	case 77: // 205 Transport layer FB message.
	case 78: // 206 Payload-specific FB message.
	case 79: // 207 Extended report.
	WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
	"%s invalid payloadtype:%d",
	__FUNCTION__, payload_type);
	return -1;
	default:
	break;
	}
	size_t payload_name_length = strlen(payload_name);

	ModuleRTPUtility::PayloadTypeMap::iterator it =
	payload_type_map_.find(payload_type);

	if (it != payload_type_map_.end()) {
	// We already use this payload type.
	ModuleRTPUtility::Payload* payload = it->second;
	assert(payload);

	size_t name_length = strlen(payload->name);

	// Check if it's the same as we already have.
	// If same, ignore sending an error.
	if (payload_name_length == name_length &&
	ModuleRTPUtility::StringCompare(
	payload->name, payload_name, payload_name_length)) {
	if (rtp_media_receiver_->PayloadIsCompatible(*payload, frequency,
	channels, rate)) {
	rtp_media_receiver_->UpdatePayloadRate(payload, rate);
	return 0;
	}
	}
	WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
	"%s invalid argument payload_type:%d already registered",
	__FUNCTION__, payload_type);
	return -1;
	}

	rtp_media_receiver_->PossiblyRemoveExistingPayloadType(
	&payload_type_map_, payload_name, payload_name_length, frequency, channels,
	rate);

	ModuleRTPUtility::Payload* payload = NULL;

	// Save the RED payload type. Used in both audio and video.
	if (ModuleRTPUtility::StringCompare(payload_name, "red", 3)) {
	red_payload_type_ = payload_type;
	payload = new ModuleRTPUtility::Payload;
	payload->audio = false;
	payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
	strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
	} else {
	payload = rtp_media_receiver_->CreatePayloadType(
	payload_name, payload_type, frequency, channels, rate);
	}
	if (payload == NULL) {
	WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
	"%s failed to register payload",
	__FUNCTION__);
	return -1;
	}
	payload_type_map_[payload_type] = payload;

	// Successful set of payload type, clear the value of last received payload
	// type since it might mean something else.
	last_received_payload_type_ = -1;
	last_received_media_payload_type_ = -1;
	return 0;
	}

	WebRtc_Word32 RTPPayloadRegistry::DeRegisterReceivePayload(
	const WebRtc_Word8 payload_type) {
	assert(rtp_media_receiver_);
	ModuleRTPUtility::PayloadTypeMap::iterator it =
	payload_type_map_.find(payload_type);

	if (it == payload_type_map_.end()) {
	WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
	"%s failed to find payload_type:%d",
	__FUNCTION__, payload_type);
	return -1;
	}
	delete it->second;
	payload_type_map_.erase(it);
	return 0;
	}

	WebRtc_Word32 RTPPayloadRegistry::ReceivePayloadType(
	const char payload_name[RTP_PAYLOAD_NAME_SIZE],
	const WebRtc_UWord32 frequency,
	const WebRtc_UWord8 channels,
	const WebRtc_UWord32 rate,
	WebRtc_Word8* payload_type) const {
	if (payload_type == NULL) {
	WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
	"%s invalid argument", __FUNCTION__);
	return -1;
	}
	size_t payload_name_length = strlen(payload_name);

	ModuleRTPUtility::PayloadTypeMap::const_iterator it =
	payload_type_map_.begin();

	while (it != payload_type_map_.end()) {
	ModuleRTPUtility::Payload* payload = it->second;
	assert(payload);

	size_t name_length = strlen(payload->name);
	if (payload_name_length == name_length &&
	ModuleRTPUtility::StringCompare(
	payload->name, payload_name, payload_name_length)) {
	// Name matches.
	if (payload->audio) {
	if (rate == 0) {
	// [default] audio, check freq and channels.
	if (payload->typeSpecific.Audio.frequency == frequency &&
	payload->typeSpecific.Audio.channels == channels) {
	*payload_type = it->first;
	return 0;
	}
	} else {
	// Non-default audio, check freq, channels and rate.
	if (payload->typeSpecific.Audio.frequency == frequency &&
	payload->typeSpecific.Audio.channels == channels &&
	payload->typeSpecific.Audio.rate == rate) {
	// extra rate condition added
	*payload_type = it->first;
	return 0;
	}
	}
	} else {
	// Video.
	*payload_type = it->first;
	return 0;
	}
	}
	it++;
	}
	return -1;
	}

	WebRtc_Word32 RTPPayloadRegistry::ReceivePayload(
	const WebRtc_Word8 payload_type,
	char payload_name[RTP_PAYLOAD_NAME_SIZE],
	WebRtc_UWord32* frequency,
	WebRtc_UWord8* channels,
	WebRtc_UWord32* rate) const {
	assert(rtp_media_receiver_);
	ModuleRTPUtility::PayloadTypeMap::const_iterator it =
	payload_type_map_.find(payload_type);

	if (it == payload_type_map_.end()) {
	return -1;
	}
	ModuleRTPUtility::Payload* payload = it->second;
	assert(payload);

	if (frequency) {
	if (payload->audio) {
	*frequency = payload->typeSpecific.Audio.frequency;
	} else {
	*frequency = kDefaultVideoFrequency;
	}
	}
	if (channels) {
	if (payload->audio) {
	*channels = payload->typeSpecific.Audio.channels;
	} else {
	*channels = 1;
	}
	}
	if (rate) {
	if (payload->audio) {
	*rate = payload->typeSpecific.Audio.rate;
	} else {
	assert(false);
	*rate = 0;
	}
	}
	if (payload_name) {
	payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
	strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
	}
	return 0;
	}

	WebRtc_UWord32 RTPPayloadRegistry::PayloadTypeToPayload(
	const WebRtc_UWord8 payload_type,
	ModuleRTPUtility::Payload*& payload) const {
	assert(rtp_media_receiver_);

	ModuleRTPUtility::PayloadTypeMap::const_iterator it =
	payload_type_map_.find(payload_type);

	// Check that this is a registered payload type.
	if (it == payload_type_map_.end()) {
	return -1;
	}
	payload = it->second;
	return 0;
	}

	bool RTPPayloadRegistry::ReportMediaPayloadType(
	WebRtc_UWord8 media_payload_type) {
	if (last_received_media_payload_type_ == media_payload_type) {
	// Media type unchanged.
	return true;
	}
	last_received_media_payload_type_ = media_payload_type;
	return false;
	}

	} // namespace webrtc