| /* |
| * Copyright 2004 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 "pc/mediasession.h" |
| |
| #include <algorithm> // For std::find_if, std::sort. |
| #include <functional> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <unordered_map> |
| #include <utility> |
| |
| #include "api/cryptoparams.h" |
| #include "api/optional.h" |
| #include "common_types.h" // NOLINT(build/include) |
| #include "media/base/h264_profile_level_id.h" |
| #include "media/base/mediaconstants.h" |
| #include "p2p/base/p2pconstants.h" |
| #include "pc/channelmanager.h" |
| #include "pc/rtpmediautils.h" |
| #include "pc/srtpfilter.h" |
| #include "rtc_base/base64.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/helpers.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/stringutils.h" |
| |
| namespace { |
| |
| using webrtc::RtpTransceiverDirection; |
| |
| const char kInline[] = "inline:"; |
| |
| void GetSupportedSdesCryptoSuiteNames(void (*func)(const rtc::CryptoOptions&, |
| std::vector<int>*), |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* names) { |
| std::vector<int> crypto_suites; |
| func(crypto_options, &crypto_suites); |
| for (const auto crypto : crypto_suites) { |
| names->push_back(rtc::SrtpCryptoSuiteToName(crypto)); |
| } |
| } |
| } // namespace |
| |
| namespace cricket { |
| |
| // RTP Profile names |
| // http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml |
| // RFC4585 |
| const char kMediaProtocolAvpf[] = "RTP/AVPF"; |
| // RFC5124 |
| const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF"; |
| |
| // We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP, |
| // but we tolerate "RTP/SAVPF" in offers we receive, for compatibility. |
| const char kMediaProtocolSavpf[] = "RTP/SAVPF"; |
| |
| const char kMediaProtocolRtpPrefix[] = "RTP/"; |
| |
| const char kMediaProtocolSctp[] = "SCTP"; |
| const char kMediaProtocolDtlsSctp[] = "DTLS/SCTP"; |
| const char kMediaProtocolUdpDtlsSctp[] = "UDP/DTLS/SCTP"; |
| const char kMediaProtocolTcpDtlsSctp[] = "TCP/DTLS/SCTP"; |
| |
| // Note that the below functions support some protocol strings purely for |
| // legacy compatibility, as required by JSEP in Section 5.1.2, Profile Names |
| // and Interoperability. |
| |
| static bool IsDtlsRtp(const std::string& protocol) { |
| // Most-likely values first. |
| return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" || |
| protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP"; |
| } |
| |
| static bool IsPlainRtp(const std::string& protocol) { |
| // Most-likely values first. |
| return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" || |
| protocol == "RTP/SAVP" || protocol == "RTP/AVP"; |
| } |
| |
| static bool IsDtlsSctp(const std::string& protocol) { |
| return protocol == kMediaProtocolDtlsSctp || |
| protocol == kMediaProtocolUdpDtlsSctp || |
| protocol == kMediaProtocolTcpDtlsSctp; |
| } |
| |
| static bool IsPlainSctp(const std::string& protocol) { |
| return protocol == kMediaProtocolSctp; |
| } |
| |
| static bool IsSctp(const std::string& protocol) { |
| return IsPlainSctp(protocol) || IsDtlsSctp(protocol); |
| } |
| |
| static RtpTransceiverDirection NegotiateRtpTransceiverDirection( |
| RtpTransceiverDirection offer, |
| RtpTransceiverDirection wants) { |
| bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer); |
| bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer); |
| bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants); |
| bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants); |
| return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send, |
| offer_send && wants_recv); |
| } |
| |
| static bool IsMediaContentOfType(const ContentInfo* content, |
| MediaType media_type) { |
| if (!content || !content->media_description()) { |
| return false; |
| } |
| return content->media_description()->type() == media_type; |
| } |
| |
| static bool CreateCryptoParams(int tag, const std::string& cipher, |
| CryptoParams *out) { |
| int key_len; |
| int salt_len; |
| if (!rtc::GetSrtpKeyAndSaltLengths( |
| rtc::SrtpCryptoSuiteFromName(cipher), &key_len, &salt_len)) { |
| return false; |
| } |
| |
| int master_key_len = key_len + salt_len; |
| std::string master_key; |
| if (!rtc::CreateRandomData(master_key_len, &master_key)) { |
| return false; |
| } |
| |
| RTC_CHECK_EQ(master_key_len, master_key.size()); |
| std::string key = rtc::Base64::Encode(master_key); |
| |
| out->tag = tag; |
| out->cipher_suite = cipher; |
| out->key_params = kInline; |
| out->key_params += key; |
| return true; |
| } |
| |
| static bool AddCryptoParams(const std::string& cipher_suite, |
| CryptoParamsVec *out) { |
| int size = static_cast<int>(out->size()); |
| |
| out->resize(size + 1); |
| return CreateCryptoParams(size, cipher_suite, &out->at(size)); |
| } |
| |
| void AddMediaCryptos(const CryptoParamsVec& cryptos, |
| MediaContentDescription* media) { |
| for (CryptoParamsVec::const_iterator crypto = cryptos.begin(); |
| crypto != cryptos.end(); ++crypto) { |
| media->AddCrypto(*crypto); |
| } |
| } |
| |
| bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites, |
| MediaContentDescription* media) { |
| CryptoParamsVec cryptos; |
| for (std::vector<std::string>::const_iterator it = crypto_suites.begin(); |
| it != crypto_suites.end(); ++it) { |
| if (!AddCryptoParams(*it, &cryptos)) { |
| return false; |
| } |
| } |
| AddMediaCryptos(cryptos, media); |
| return true; |
| } |
| |
| const CryptoParamsVec* GetCryptos(const ContentInfo* content) { |
| if (!content || !content->media_description()) { |
| return nullptr; |
| } |
| return &content->media_description()->cryptos(); |
| } |
| |
| bool FindMatchingCrypto(const CryptoParamsVec& cryptos, |
| const CryptoParams& crypto, |
| CryptoParams* out) { |
| for (CryptoParamsVec::const_iterator it = cryptos.begin(); |
| it != cryptos.end(); ++it) { |
| if (crypto.Matches(*it)) { |
| *out = *it; |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the |
| // low overhead. |
| void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| if (crypto_options.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| if (crypto_options.enable_aes128_sha1_32_crypto_cipher) { |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32); |
| } |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| } |
| |
| void GetSupportedAudioSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| if (crypto_options.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| } |
| |
| void GetSupportedVideoSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options, |
| std::vector<int>* crypto_suites) { |
| if (crypto_options.enable_gcm_crypto_suites) { |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM); |
| crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM); |
| } |
| crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80); |
| } |
| |
| void GetSupportedDataSdesCryptoSuiteNames( |
| const rtc::CryptoOptions& crypto_options, |
| std::vector<std::string>* crypto_suite_names) { |
| GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites, |
| crypto_options, crypto_suite_names); |
| } |
| |
| // Support any GCM cipher (if enabled through options). For video support only |
| // 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless |
| // bundle is enabled because it is low overhead. |
| // Pick the crypto in the list that is supported. |
| static bool SelectCrypto(const MediaContentDescription* offer, |
| bool bundle, |
| const rtc::CryptoOptions& crypto_options, |
| CryptoParams *crypto) { |
| bool audio = offer->type() == MEDIA_TYPE_AUDIO; |
| const CryptoParamsVec& cryptos = offer->cryptos(); |
| |
| for (CryptoParamsVec::const_iterator i = cryptos.begin(); |
| i != cryptos.end(); ++i) { |
| if ((crypto_options.enable_gcm_crypto_suites && |
| rtc::IsGcmCryptoSuiteName(i->cipher_suite)) || |
| rtc::CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite || |
| (rtc::CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio && |
| !bundle && crypto_options.enable_aes128_sha1_32_crypto_cipher)) { |
| return CreateCryptoParams(i->tag, i->cipher_suite, crypto); |
| } |
| } |
| return false; |
| } |
| |
| // Generate random SSRC values that are not already present in |params_vec|. |
| // The generated values are added to |ssrcs|. |
| // |num_ssrcs| is the number of the SSRC will be generated. |
| static void GenerateSsrcs(const StreamParamsVec& params_vec, |
| int num_ssrcs, |
| std::vector<uint32_t>* ssrcs) { |
| for (int i = 0; i < num_ssrcs; i++) { |
| uint32_t candidate; |
| do { |
| candidate = rtc::CreateRandomNonZeroId(); |
| } while (GetStreamBySsrc(params_vec, candidate) || |
| std::count(ssrcs->begin(), ssrcs->end(), candidate) > 0); |
| ssrcs->push_back(candidate); |
| } |
| } |
| |
| // Finds all StreamParams of all media types and attach them to stream_params. |
| static void GetCurrentStreamParams(const SessionDescription* sdesc, |
| StreamParamsVec* stream_params) { |
| RTC_DCHECK(stream_params); |
| if (!sdesc) { |
| return; |
| } |
| for (const ContentInfo& content : sdesc->contents()) { |
| if (!content.media_description()) { |
| continue; |
| } |
| for (const StreamParams& params : content.media_description()->streams()) { |
| stream_params->push_back(params); |
| } |
| } |
| } |
| |
| // Filters the data codecs for the data channel type. |
| void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) { |
| // Filter RTP codec for SCTP and vice versa. |
| const char* codec_name = |
| sctp ? kGoogleRtpDataCodecName : kGoogleSctpDataCodecName; |
| for (std::vector<DataCodec>::iterator iter = codecs->begin(); |
| iter != codecs->end();) { |
| if (CodecNamesEq(iter->name, codec_name)) { |
| iter = codecs->erase(iter); |
| } else { |
| ++iter; |
| } |
| } |
| } |
| |
| template <typename IdStruct> |
| class UsedIds { |
| public: |
| UsedIds(int min_allowed_id, int max_allowed_id) |
| : min_allowed_id_(min_allowed_id), |
| max_allowed_id_(max_allowed_id), |
| next_id_(max_allowed_id) { |
| } |
| |
| // Loops through all Id in |ids| and changes its id if it is |
| // already in use by another IdStruct. Call this methods with all Id |
| // in a session description to make sure no duplicate ids exists. |
| // Note that typename Id must be a type of IdStruct. |
| template <typename Id> |
| void FindAndSetIdUsed(std::vector<Id>* ids) { |
| for (typename std::vector<Id>::iterator it = ids->begin(); |
| it != ids->end(); ++it) { |
| FindAndSetIdUsed(&*it); |
| } |
| } |
| |
| // Finds and sets an unused id if the |idstruct| id is already in use. |
| void FindAndSetIdUsed(IdStruct* idstruct) { |
| const int original_id = idstruct->id; |
| int new_id = idstruct->id; |
| |
| if (original_id > max_allowed_id_ || original_id < min_allowed_id_) { |
| // If the original id is not in range - this is an id that can't be |
| // dynamically changed. |
| return; |
| } |
| |
| if (IsIdUsed(original_id)) { |
| new_id = FindUnusedId(); |
| RTC_LOG(LS_WARNING) << "Duplicate id found. Reassigning from " |
| << original_id << " to " << new_id; |
| idstruct->id = new_id; |
| } |
| SetIdUsed(new_id); |
| } |
| |
| private: |
| // Returns the first unused id in reverse order. |
| // This hopefully reduce the risk of more collisions. We want to change the |
| // default ids as little as possible. |
| int FindUnusedId() { |
| while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) { |
| --next_id_; |
| } |
| RTC_DCHECK(next_id_ >= min_allowed_id_); |
| return next_id_; |
| } |
| |
| bool IsIdUsed(int new_id) { |
| return id_set_.find(new_id) != id_set_.end(); |
| } |
| |
| void SetIdUsed(int new_id) { |
| id_set_.insert(new_id); |
| } |
| |
| const int min_allowed_id_; |
| const int max_allowed_id_; |
| int next_id_; |
| std::set<int> id_set_; |
| }; |
| |
| // Helper class used for finding duplicate RTP payload types among audio, video |
| // and data codecs. When bundle is used the payload types may not collide. |
| class UsedPayloadTypes : public UsedIds<Codec> { |
| public: |
| UsedPayloadTypes() |
| : UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) { |
| } |
| |
| |
| private: |
| static const int kDynamicPayloadTypeMin = 96; |
| static const int kDynamicPayloadTypeMax = 127; |
| }; |
| |
| // Helper class used for finding duplicate RTP Header extension ids among |
| // audio and video extensions. |
| class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> { |
| public: |
| UsedRtpHeaderExtensionIds() |
| : UsedIds<webrtc::RtpExtension>(webrtc::RtpExtension::kMinId, |
| webrtc::RtpExtension::kMaxId) {} |
| |
| private: |
| }; |
| |
| // Adds a StreamParams for each SenderOptions in |sender_options| to |
| // content_description. |
| // |current_params| - All currently known StreamParams of any media type. |
| template <class C> |
| static bool AddStreamParams( |
| const std::vector<SenderOptions>& sender_options, |
| const std::string& rtcp_cname, |
| StreamParamsVec* current_streams, |
| MediaContentDescriptionImpl<C>* content_description) { |
| // SCTP streams are not negotiated using SDP/ContentDescriptions. |
| if (IsSctp(content_description->protocol())) { |
| return true; |
| } |
| |
| const bool include_rtx_streams = |
| ContainsRtxCodec(content_description->codecs()); |
| |
| |
| const bool include_flexfec_stream = |
| ContainsFlexfecCodec(content_description->codecs()); |
| |
| for (const SenderOptions& sender : sender_options) { |
| // groupid is empty for StreamParams generated using |
| // MediaSessionDescriptionFactory. |
| StreamParams* param = |
| GetStreamByIds(*current_streams, "" /*group_id*/, sender.track_id); |
| if (!param) { |
| // This is a new sender. |
| std::vector<uint32_t> ssrcs; |
| GenerateSsrcs(*current_streams, sender.num_sim_layers, &ssrcs); |
| StreamParams stream_param; |
| stream_param.id = sender.track_id; |
| // Add the generated ssrc. |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| stream_param.ssrcs.push_back(ssrcs[i]); |
| } |
| if (sender.num_sim_layers > 1) { |
| SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs); |
| stream_param.ssrc_groups.push_back(group); |
| } |
| // Generate extra ssrcs for include_rtx_streams case. |
| if (include_rtx_streams) { |
| // Generate an RTX ssrc for every ssrc in the group. |
| std::vector<uint32_t> rtx_ssrcs; |
| GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()), |
| &rtx_ssrcs); |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]); |
| } |
| } |
| // Generate extra ssrc for include_flexfec_stream case. |
| if (include_flexfec_stream) { |
| // TODO(brandtr): Update when we support multistream protection. |
| if (ssrcs.size() == 1) { |
| std::vector<uint32_t> flexfec_ssrcs; |
| GenerateSsrcs(*current_streams, 1, &flexfec_ssrcs); |
| stream_param.AddFecFrSsrc(ssrcs[0], flexfec_ssrcs[0]); |
| } else if (!ssrcs.empty()) { |
| RTC_LOG(LS_WARNING) |
| << "Our FlexFEC implementation only supports protecting " |
| "a single media streams. This session has multiple " |
| "media streams however, so no FlexFEC SSRC will be generated."; |
| } |
| } |
| stream_param.cname = rtcp_cname; |
| stream_param.set_stream_ids(sender.stream_ids); |
| content_description->AddStream(stream_param); |
| |
| // Store the new StreamParams in current_streams. |
| // This is necessary so that we can use the CNAME for other media types. |
| current_streams->push_back(stream_param); |
| } else { |
| // Use existing generated SSRCs/groups, but update the sync_label if |
| // necessary. This may be needed if a MediaStreamTrack was moved from one |
| // MediaStream to another. |
| param->set_stream_ids(sender.stream_ids); |
| content_description->AddStream(*param); |
| } |
| } |
| return true; |
| } |
| |
| // Updates the transport infos of the |sdesc| according to the given |
| // |bundle_group|. The transport infos of the content names within the |
| // |bundle_group| should be updated to use the ufrag, pwd and DTLS role of the |
| // first content within the |bundle_group|. |
| static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group, |
| SessionDescription* sdesc) { |
| // The bundle should not be empty. |
| if (!sdesc || !bundle_group.FirstContentName()) { |
| return false; |
| } |
| |
| // We should definitely have a transport for the first content. |
| const std::string& selected_content_name = *bundle_group.FirstContentName(); |
| const TransportInfo* selected_transport_info = |
| sdesc->GetTransportInfoByName(selected_content_name); |
| if (!selected_transport_info) { |
| return false; |
| } |
| |
| // Set the other contents to use the same ICE credentials. |
| const std::string& selected_ufrag = |
| selected_transport_info->description.ice_ufrag; |
| const std::string& selected_pwd = |
| selected_transport_info->description.ice_pwd; |
| ConnectionRole selected_connection_role = |
| selected_transport_info->description.connection_role; |
| for (TransportInfos::iterator it = |
| sdesc->transport_infos().begin(); |
| it != sdesc->transport_infos().end(); ++it) { |
| if (bundle_group.HasContentName(it->content_name) && |
| it->content_name != selected_content_name) { |
| it->description.ice_ufrag = selected_ufrag; |
| it->description.ice_pwd = selected_pwd; |
| it->description.connection_role = selected_connection_role; |
| } |
| } |
| return true; |
| } |
| |
| // Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and |
| // sets it to |cryptos|. |
| static bool GetCryptosByName(const SessionDescription* sdesc, |
| const std::string& content_name, |
| CryptoParamsVec* cryptos) { |
| if (!sdesc || !cryptos) { |
| return false; |
| } |
| const ContentInfo* content = sdesc->GetContentByName(content_name); |
| if (!content || !content->media_description()) { |
| return false; |
| } |
| *cryptos = content->media_description()->cryptos(); |
| return true; |
| } |
| |
| // Predicate function used by the remove_if. |
| // Returns true if the |crypto|'s cipher_suite is not found in |filter|. |
| static bool CryptoNotFound(const CryptoParams crypto, |
| const CryptoParamsVec* filter) { |
| if (filter == NULL) { |
| return true; |
| } |
| for (CryptoParamsVec::const_iterator it = filter->begin(); |
| it != filter->end(); ++it) { |
| if (it->cipher_suite == crypto.cipher_suite) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| // Prunes the |target_cryptos| by removing the crypto params (cipher_suite) |
| // which are not available in |filter|. |
| static void PruneCryptos(const CryptoParamsVec& filter, |
| CryptoParamsVec* target_cryptos) { |
| if (!target_cryptos) { |
| return; |
| } |
| target_cryptos->erase(std::remove_if(target_cryptos->begin(), |
| target_cryptos->end(), |
| bind2nd(ptr_fun(CryptoNotFound), |
| &filter)), |
| target_cryptos->end()); |
| } |
| |
| bool IsRtpProtocol(const std::string& protocol) { |
| return protocol.empty() || |
| (protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos); |
| } |
| |
| static bool IsRtpContent(SessionDescription* sdesc, |
| const std::string& content_name) { |
| bool is_rtp = false; |
| ContentInfo* content = sdesc->GetContentByName(content_name); |
| if (content && content->media_description()) { |
| is_rtp = IsRtpProtocol(content->media_description()->protocol()); |
| } |
| return is_rtp; |
| } |
| |
| // Updates the crypto parameters of the |sdesc| according to the given |
| // |bundle_group|. The crypto parameters of all the contents within the |
| // |bundle_group| should be updated to use the common subset of the |
| // available cryptos. |
| static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group, |
| SessionDescription* sdesc) { |
| // The bundle should not be empty. |
| if (!sdesc || !bundle_group.FirstContentName()) { |
| return false; |
| } |
| |
| bool common_cryptos_needed = false; |
| // Get the common cryptos. |
| const ContentNames& content_names = bundle_group.content_names(); |
| CryptoParamsVec common_cryptos; |
| for (ContentNames::const_iterator it = content_names.begin(); |
| it != content_names.end(); ++it) { |
| if (!IsRtpContent(sdesc, *it)) { |
| continue; |
| } |
| // The common cryptos are needed if any of the content does not have DTLS |
| // enabled. |
| if (!sdesc->GetTransportInfoByName(*it)->description.secure()) { |
| common_cryptos_needed = true; |
| } |
| if (it == content_names.begin()) { |
| // Initial the common_cryptos with the first content in the bundle group. |
| if (!GetCryptosByName(sdesc, *it, &common_cryptos)) { |
| return false; |
| } |
| if (common_cryptos.empty()) { |
| // If there's no crypto params, we should just return. |
| return true; |
| } |
| } else { |
| CryptoParamsVec cryptos; |
| if (!GetCryptosByName(sdesc, *it, &cryptos)) { |
| return false; |
| } |
| PruneCryptos(cryptos, &common_cryptos); |
| } |
| } |
| |
| if (common_cryptos.empty() && common_cryptos_needed) { |
| return false; |
| } |
| |
| // Update to use the common cryptos. |
| for (ContentNames::const_iterator it = content_names.begin(); |
| it != content_names.end(); ++it) { |
| if (!IsRtpContent(sdesc, *it)) { |
| continue; |
| } |
| ContentInfo* content = sdesc->GetContentByName(*it); |
| if (IsMediaContent(content)) { |
| MediaContentDescription* media_desc = content->media_description(); |
| if (!media_desc) { |
| return false; |
| } |
| media_desc->set_cryptos(common_cryptos); |
| } |
| } |
| return true; |
| } |
| |
| template <class C> |
| static bool ContainsRtxCodec(const std::vector<C>& codecs) { |
| for (const auto& codec : codecs) { |
| if (IsRtxCodec(codec)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| template <class C> |
| static bool IsRtxCodec(const C& codec) { |
| return STR_CASE_CMP(codec.name.c_str(), kRtxCodecName) == 0; |
| } |
| |
| template <class C> |
| static bool ContainsFlexfecCodec(const std::vector<C>& codecs) { |
| for (const auto& codec : codecs) { |
| if (IsFlexfecCodec(codec)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| template <class C> |
| static bool IsFlexfecCodec(const C& codec) { |
| return STR_CASE_CMP(codec.name.c_str(), kFlexfecCodecName) == 0; |
| } |
| |
| // Create a media content to be offered for the given |sender_options|, |
| // according to the given options.rtcp_mux, session_options.is_muc, codecs, |
| // secure_transport, crypto, and current_streams. If we don't currently have |
| // crypto (in current_cryptos) and it is enabled (in secure_policy), crypto is |
| // created (according to crypto_suites). The created content is added to the |
| // offer. |
| template <class C> |
| static bool CreateMediaContentOffer( |
| const std::vector<SenderOptions>& sender_options, |
| const MediaSessionOptions& session_options, |
| const std::vector<C>& codecs, |
| const SecurePolicy& secure_policy, |
| const CryptoParamsVec* current_cryptos, |
| const std::vector<std::string>& crypto_suites, |
| const RtpHeaderExtensions& rtp_extensions, |
| StreamParamsVec* current_streams, |
| MediaContentDescriptionImpl<C>* offer) { |
| offer->AddCodecs(codecs); |
| |
| offer->set_rtcp_mux(session_options.rtcp_mux_enabled); |
| if (offer->type() == cricket::MEDIA_TYPE_VIDEO) { |
| offer->set_rtcp_reduced_size(true); |
| } |
| offer->set_rtp_header_extensions(rtp_extensions); |
| |
| if (!AddStreamParams(sender_options, session_options.rtcp_cname, |
| current_streams, offer)) { |
| return false; |
| } |
| |
| if (secure_policy != SEC_DISABLED) { |
| if (current_cryptos) { |
| AddMediaCryptos(*current_cryptos, offer); |
| } |
| if (offer->cryptos().empty()) { |
| if (!CreateMediaCryptos(crypto_suites, offer)) { |
| return false; |
| } |
| } |
| } |
| |
| if (secure_policy == SEC_REQUIRED && offer->cryptos().empty()) { |
| return false; |
| } |
| return true; |
| } |
| |
| template <class C> |
| static bool ReferencedCodecsMatch(const std::vector<C>& codecs1, |
| const int codec1_id, |
| const std::vector<C>& codecs2, |
| const int codec2_id) { |
| const C* codec1 = FindCodecById(codecs1, codec1_id); |
| const C* codec2 = FindCodecById(codecs2, codec2_id); |
| return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2); |
| } |
| |
| template <class C> |
| static void NegotiateCodecs(const std::vector<C>& local_codecs, |
| const std::vector<C>& offered_codecs, |
| std::vector<C>* negotiated_codecs) { |
| for (const C& ours : local_codecs) { |
| C theirs; |
| // Note that we intentionally only find one matching codec for each of our |
| // local codecs, in case the remote offer contains duplicate codecs. |
| if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) { |
| C negotiated = ours; |
| negotiated.IntersectFeedbackParams(theirs); |
| if (IsRtxCodec(negotiated)) { |
| const auto apt_it = |
| theirs.params.find(kCodecParamAssociatedPayloadType); |
| // FindMatchingCodec shouldn't return something with no apt value. |
| RTC_DCHECK(apt_it != theirs.params.end()); |
| negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second); |
| } |
| if (CodecNamesEq(ours.name.c_str(), kH264CodecName)) { |
| webrtc::H264::GenerateProfileLevelIdForAnswer( |
| ours.params, theirs.params, &negotiated.params); |
| } |
| negotiated.id = theirs.id; |
| negotiated.name = theirs.name; |
| negotiated_codecs->push_back(std::move(negotiated)); |
| } |
| } |
| // RFC3264: Although the answerer MAY list the formats in their desired |
| // order of preference, it is RECOMMENDED that unless there is a |
| // specific reason, the answerer list formats in the same relative order |
| // they were present in the offer. |
| std::unordered_map<int, int> payload_type_preferences; |
| int preference = static_cast<int>(offered_codecs.size() + 1); |
| for (const C& codec : offered_codecs) { |
| payload_type_preferences[codec.id] = preference--; |
| } |
| std::sort(negotiated_codecs->begin(), negotiated_codecs->end(), |
| [&payload_type_preferences](const C& a, const C& b) { |
| return payload_type_preferences[a.id] > |
| payload_type_preferences[b.id]; |
| }); |
| } |
| |
| // Finds a codec in |codecs2| that matches |codec_to_match|, which is |
| // a member of |codecs1|. If |codec_to_match| is an RTX codec, both |
| // the codecs themselves and their associated codecs must match. |
| template <class C> |
| static bool FindMatchingCodec(const std::vector<C>& codecs1, |
| const std::vector<C>& codecs2, |
| const C& codec_to_match, |
| C* found_codec) { |
| // |codec_to_match| should be a member of |codecs1|, in order to look up RTX |
| // codecs' associated codecs correctly. If not, that's a programming error. |
| RTC_DCHECK(std::find_if(codecs1.begin(), codecs1.end(), |
| [&codec_to_match](const C& codec) { |
| return &codec == &codec_to_match; |
| }) != codecs1.end()); |
| for (const C& potential_match : codecs2) { |
| if (potential_match.Matches(codec_to_match)) { |
| if (IsRtxCodec(codec_to_match)) { |
| int apt_value_1 = 0; |
| int apt_value_2 = 0; |
| if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType, |
| &apt_value_1) || |
| !potential_match.GetParam(kCodecParamAssociatedPayloadType, |
| &apt_value_2)) { |
| RTC_LOG(LS_WARNING) << "RTX missing associated payload type."; |
| continue; |
| } |
| if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2, |
| apt_value_2)) { |
| continue; |
| } |
| } |
| if (found_codec) { |
| *found_codec = potential_match; |
| } |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Find the codec in |codec_list| that |rtx_codec| is associated with. |
| template <class C> |
| static const C* GetAssociatedCodec(const std::vector<C>& codec_list, |
| const C& rtx_codec) { |
| std::string associated_pt_str; |
| if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType, |
| &associated_pt_str)) { |
| RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.name |
| << " is missing an associated payload type."; |
| return nullptr; |
| } |
| |
| int associated_pt; |
| if (!rtc::FromString(associated_pt_str, &associated_pt)) { |
| RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str |
| << " of RTX codec " << rtx_codec.name |
| << " to an integer."; |
| return nullptr; |
| } |
| |
| // Find the associated reference codec for the reference RTX codec. |
| const C* associated_codec = FindCodecById(codec_list, associated_pt); |
| if (!associated_codec) { |
| RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type " |
| << associated_pt << " for RTX codec " << rtx_codec.name |
| << "."; |
| } |
| return associated_codec; |
| } |
| |
| // Adds all codecs from |reference_codecs| to |offered_codecs| that don't |
| // already exist in |offered_codecs| and ensure the payload types don't |
| // collide. |
| template <class C> |
| static void MergeCodecs(const std::vector<C>& reference_codecs, |
| std::vector<C>* offered_codecs, |
| UsedPayloadTypes* used_pltypes) { |
| // Add all new codecs that are not RTX codecs. |
| for (const C& reference_codec : reference_codecs) { |
| if (!IsRtxCodec(reference_codec) && |
| !FindMatchingCodec<C>(reference_codecs, *offered_codecs, |
| reference_codec, nullptr)) { |
| C codec = reference_codec; |
| used_pltypes->FindAndSetIdUsed(&codec); |
| offered_codecs->push_back(codec); |
| } |
| } |
| |
| // Add all new RTX codecs. |
| for (const C& reference_codec : reference_codecs) { |
| if (IsRtxCodec(reference_codec) && |
| !FindMatchingCodec<C>(reference_codecs, *offered_codecs, |
| reference_codec, nullptr)) { |
| C rtx_codec = reference_codec; |
| const C* associated_codec = |
| GetAssociatedCodec(reference_codecs, rtx_codec); |
| if (!associated_codec) { |
| continue; |
| } |
| // Find a codec in the offered list that matches the reference codec. |
| // Its payload type may be different than the reference codec. |
| C matching_codec; |
| if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs, |
| *associated_codec, &matching_codec)) { |
| RTC_LOG(LS_WARNING) |
| << "Couldn't find matching " << associated_codec->name << " codec."; |
| continue; |
| } |
| |
| rtx_codec.params[kCodecParamAssociatedPayloadType] = |
| rtc::ToString(matching_codec.id); |
| used_pltypes->FindAndSetIdUsed(&rtx_codec); |
| offered_codecs->push_back(rtx_codec); |
| } |
| } |
| } |
| |
| static bool FindByUriAndEncryption(const RtpHeaderExtensions& extensions, |
| const webrtc::RtpExtension& ext_to_match, |
| webrtc::RtpExtension* found_extension) { |
| for (RtpHeaderExtensions::const_iterator it = extensions.begin(); |
| it != extensions.end(); ++it) { |
| // We assume that all URIs are given in a canonical format. |
| if (it->uri == ext_to_match.uri && it->encrypt == ext_to_match.encrypt) { |
| if (found_extension) { |
| *found_extension = *it; |
| } |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| static bool FindByUri(const RtpHeaderExtensions& extensions, |
| const webrtc::RtpExtension& ext_to_match, |
| webrtc::RtpExtension* found_extension) { |
| // We assume that all URIs are given in a canonical format. |
| const webrtc::RtpExtension* found = |
| webrtc::RtpExtension::FindHeaderExtensionByUri(extensions, |
| ext_to_match.uri); |
| if (!found) { |
| return false; |
| } |
| if (found_extension) { |
| *found_extension = *found; |
| } |
| return true; |
| } |
| |
| static bool FindByUriWithEncryptionPreference( |
| const RtpHeaderExtensions& extensions, |
| const webrtc::RtpExtension& ext_to_match, bool encryption_preference, |
| webrtc::RtpExtension* found_extension) { |
| const webrtc::RtpExtension* unencrypted_extension = nullptr; |
| for (RtpHeaderExtensions::const_iterator it = extensions.begin(); |
| it != extensions.end(); ++it) { |
| // We assume that all URIs are given in a canonical format. |
| if (it->uri == ext_to_match.uri) { |
| if (!encryption_preference || it->encrypt) { |
| if (found_extension) { |
| *found_extension = *it; |
| } |
| return true; |
| } |
| unencrypted_extension = &(*it); |
| } |
| } |
| if (unencrypted_extension) { |
| if (found_extension) { |
| *found_extension = *unencrypted_extension; |
| } |
| return true; |
| } |
| return false; |
| } |
| |
| // Adds all extensions from |reference_extensions| to |offered_extensions| that |
| // don't already exist in |offered_extensions| and ensure the IDs don't |
| // collide. If an extension is added, it's also added to |regular_extensions| or |
| // |encrypted_extensions|, and if the extension is in |regular_extensions| or |
| // |encrypted_extensions|, its ID is marked as used in |used_ids|. |
| // |offered_extensions| is for either audio or video while |regular_extensions| |
| // and |encrypted_extensions| are used for both audio and video. There could be |
| // overlap between audio extensions and video extensions. |
| static void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions, |
| RtpHeaderExtensions* offered_extensions, |
| RtpHeaderExtensions* regular_extensions, |
| RtpHeaderExtensions* encrypted_extensions, |
| UsedRtpHeaderExtensionIds* used_ids) { |
| for (auto reference_extension : reference_extensions) { |
| if (!FindByUriAndEncryption(*offered_extensions, reference_extension, |
| nullptr)) { |
| webrtc::RtpExtension existing; |
| if (reference_extension.encrypt) { |
| if (FindByUriAndEncryption(*encrypted_extensions, reference_extension, |
| &existing)) { |
| offered_extensions->push_back(existing); |
| } else { |
| used_ids->FindAndSetIdUsed(&reference_extension); |
| encrypted_extensions->push_back(reference_extension); |
| offered_extensions->push_back(reference_extension); |
| } |
| } else { |
| if (FindByUriAndEncryption(*regular_extensions, reference_extension, |
| &existing)) { |
| offered_extensions->push_back(existing); |
| } else { |
| used_ids->FindAndSetIdUsed(&reference_extension); |
| regular_extensions->push_back(reference_extension); |
| offered_extensions->push_back(reference_extension); |
| } |
| } |
| } |
| } |
| } |
| |
| static void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* extensions, |
| RtpHeaderExtensions* all_extensions, |
| UsedRtpHeaderExtensionIds* used_ids) { |
| RtpHeaderExtensions encrypted_extensions; |
| for (const webrtc::RtpExtension& extension : *extensions) { |
| webrtc::RtpExtension existing; |
| // Don't add encrypted extensions again that were already included in a |
| // previous offer or regular extensions that are also included as encrypted |
| // extensions. |
| if (extension.encrypt || |
| !webrtc::RtpExtension::IsEncryptionSupported(extension.uri) || |
| (FindByUriWithEncryptionPreference(*extensions, extension, true, |
| &existing) && existing.encrypt)) { |
| continue; |
| } |
| |
| if (FindByUri(*all_extensions, extension, &existing)) { |
| encrypted_extensions.push_back(existing); |
| } else { |
| webrtc::RtpExtension encrypted(extension); |
| encrypted.encrypt = true; |
| used_ids->FindAndSetIdUsed(&encrypted); |
| all_extensions->push_back(encrypted); |
| encrypted_extensions.push_back(encrypted); |
| } |
| } |
| extensions->insert(extensions->end(), encrypted_extensions.begin(), |
| encrypted_extensions.end()); |
| } |
| |
| static void NegotiateRtpHeaderExtensions( |
| const RtpHeaderExtensions& local_extensions, |
| const RtpHeaderExtensions& offered_extensions, |
| bool enable_encrypted_rtp_header_extensions, |
| RtpHeaderExtensions* negotiated_extenstions) { |
| RtpHeaderExtensions::const_iterator ours; |
| for (ours = local_extensions.begin(); |
| ours != local_extensions.end(); ++ours) { |
| webrtc::RtpExtension theirs; |
| if (FindByUriWithEncryptionPreference(offered_extensions, *ours, |
| enable_encrypted_rtp_header_extensions, &theirs)) { |
| // We respond with their RTP header extension id. |
| negotiated_extenstions->push_back(theirs); |
| } |
| } |
| } |
| |
| static void StripCNCodecs(AudioCodecs* audio_codecs) { |
| AudioCodecs::iterator iter = audio_codecs->begin(); |
| while (iter != audio_codecs->end()) { |
| if (STR_CASE_CMP(iter->name.c_str(), kComfortNoiseCodecName) == 0) { |
| iter = audio_codecs->erase(iter); |
| } else { |
| ++iter; |
| } |
| } |
| } |
| |
| // Create a media content to be answered for the given |sender_options| |
| // according to the given session_options.rtcp_mux, session_options.streams, |
| // codecs, crypto, and current_streams. If we don't currently have crypto (in |
| // current_cryptos) and it is enabled (in secure_policy), crypto is created |
| // (according to crypto_suites). The codecs, rtcp_mux, and crypto are all |
| // negotiated with the offer. If the negotiation fails, this method returns |
| // false. The created content is added to the offer. |
| template <class C> |
| static bool CreateMediaContentAnswer( |
| const MediaContentDescriptionImpl<C>* offer, |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const std::vector<C>& local_codecs, |
| const SecurePolicy& sdes_policy, |
| const CryptoParamsVec* current_cryptos, |
| const RtpHeaderExtensions& local_rtp_extenstions, |
| bool enable_encrypted_rtp_header_extensions, |
| StreamParamsVec* current_streams, |
| bool bundle_enabled, |
| MediaContentDescriptionImpl<C>* answer) { |
| std::vector<C> negotiated_codecs; |
| NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs); |
| answer->AddCodecs(negotiated_codecs); |
| answer->set_protocol(offer->protocol()); |
| RtpHeaderExtensions negotiated_rtp_extensions; |
| NegotiateRtpHeaderExtensions(local_rtp_extenstions, |
| offer->rtp_header_extensions(), |
| enable_encrypted_rtp_header_extensions, |
| &negotiated_rtp_extensions); |
| answer->set_rtp_header_extensions(negotiated_rtp_extensions); |
| |
| answer->set_rtcp_mux(session_options.rtcp_mux_enabled && offer->rtcp_mux()); |
| if (answer->type() == cricket::MEDIA_TYPE_VIDEO) { |
| answer->set_rtcp_reduced_size(offer->rtcp_reduced_size()); |
| } |
| |
| if (sdes_policy != SEC_DISABLED) { |
| CryptoParams crypto; |
| if (SelectCrypto(offer, bundle_enabled, session_options.crypto_options, |
| &crypto)) { |
| if (current_cryptos) { |
| FindMatchingCrypto(*current_cryptos, crypto, &crypto); |
| } |
| answer->AddCrypto(crypto); |
| } |
| } |
| |
| if (answer->cryptos().empty() && sdes_policy == SEC_REQUIRED) { |
| return false; |
| } |
| |
| if (!AddStreamParams(media_description_options.sender_options, |
| session_options.rtcp_cname, current_streams, answer)) { |
| return false; // Something went seriously wrong. |
| } |
| |
| answer->set_direction(NegotiateRtpTransceiverDirection( |
| offer->direction(), media_description_options.direction)); |
| return true; |
| } |
| |
| static bool IsMediaProtocolSupported(MediaType type, |
| const std::string& protocol, |
| bool secure_transport) { |
| // Since not all applications serialize and deserialize the media protocol, |
| // we will have to accept |protocol| to be empty. |
| if (protocol.empty()) { |
| return true; |
| } |
| |
| if (type == MEDIA_TYPE_DATA) { |
| // Check for SCTP, but also for RTP for RTP-based data channels. |
| // TODO(pthatcher): Remove RTP once RTP-based data channels are gone. |
| if (secure_transport) { |
| // Most likely scenarios first. |
| return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) || |
| IsPlainRtp(protocol); |
| } else { |
| return IsPlainSctp(protocol) || IsPlainRtp(protocol); |
| } |
| } |
| |
| // Allow for non-DTLS RTP protocol even when using DTLS because that's what |
| // JSEP specifies. |
| if (secure_transport) { |
| // Most likely scenarios first. |
| return IsDtlsRtp(protocol) || IsPlainRtp(protocol); |
| } else { |
| return IsPlainRtp(protocol); |
| } |
| } |
| |
| static void SetMediaProtocol(bool secure_transport, |
| MediaContentDescription* desc) { |
| if (!desc->cryptos().empty()) |
| desc->set_protocol(kMediaProtocolSavpf); |
| else if (secure_transport) |
| desc->set_protocol(kMediaProtocolDtlsSavpf); |
| else |
| desc->set_protocol(kMediaProtocolAvpf); |
| } |
| |
| // Gets the TransportInfo of the given |content_name| from the |
| // |current_description|. If doesn't exist, returns a new one. |
| static const TransportDescription* GetTransportDescription( |
| const std::string& content_name, |
| const SessionDescription* current_description) { |
| const TransportDescription* desc = NULL; |
| if (current_description) { |
| const TransportInfo* info = |
| current_description->GetTransportInfoByName(content_name); |
| if (info) { |
| desc = &info->description; |
| } |
| } |
| return desc; |
| } |
| |
| // Gets the current DTLS state from the transport description. |
| static bool IsDtlsActive(const ContentInfo* content, |
| const SessionDescription* current_description) { |
| if (!content) { |
| return false; |
| } |
| |
| size_t msection_index = content - ¤t_description->contents()[0]; |
| |
| if (current_description->transport_infos().size() <= msection_index) { |
| return false; |
| } |
| |
| return current_description->transport_infos()[msection_index] |
| .description.secure(); |
| } |
| |
| void MediaDescriptionOptions::AddAudioSender( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids) { |
| RTC_DCHECK(type == MEDIA_TYPE_AUDIO); |
| AddSenderInternal(track_id, stream_ids, 1); |
| } |
| |
| void MediaDescriptionOptions::AddVideoSender( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| int num_sim_layers) { |
| RTC_DCHECK(type == MEDIA_TYPE_VIDEO); |
| AddSenderInternal(track_id, stream_ids, num_sim_layers); |
| } |
| |
| void MediaDescriptionOptions::AddRtpDataChannel(const std::string& track_id, |
| const std::string& stream_id) { |
| RTC_DCHECK(type == MEDIA_TYPE_DATA); |
| // TODO(steveanton): Is it the case that RtpDataChannel will never have more |
| // than one stream? |
| AddSenderInternal(track_id, {stream_id}, 1); |
| } |
| |
| void MediaDescriptionOptions::AddSenderInternal( |
| const std::string& track_id, |
| const std::vector<std::string>& stream_ids, |
| int num_sim_layers) { |
| // TODO(steveanton): Support any number of stream ids. |
| RTC_CHECK(stream_ids.size() == 1U); |
| sender_options.push_back(SenderOptions{track_id, stream_ids, num_sim_layers}); |
| } |
| |
| bool MediaSessionOptions::HasMediaDescription(MediaType type) const { |
| return std::find_if(media_description_options.begin(), |
| media_description_options.end(), |
| [type](const MediaDescriptionOptions& t) { |
| return t.type == type; |
| }) != media_description_options.end(); |
| } |
| |
| MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| const TransportDescriptionFactory* transport_desc_factory) |
| : transport_desc_factory_(transport_desc_factory) {} |
| |
| MediaSessionDescriptionFactory::MediaSessionDescriptionFactory( |
| ChannelManager* channel_manager, |
| const TransportDescriptionFactory* transport_desc_factory) |
| : transport_desc_factory_(transport_desc_factory) { |
| channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_); |
| channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_); |
| channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_); |
| channel_manager->GetSupportedVideoCodecs(&video_codecs_); |
| channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_); |
| channel_manager->GetSupportedDataCodecs(&data_codecs_); |
| ComputeAudioCodecsIntersectionAndUnion(); |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs() |
| const { |
| return audio_sendrecv_codecs_; |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::audio_send_codecs() const { |
| return audio_send_codecs_; |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const { |
| return audio_recv_codecs_; |
| } |
| |
| void MediaSessionDescriptionFactory::set_audio_codecs( |
| const AudioCodecs& send_codecs, const AudioCodecs& recv_codecs) { |
| audio_send_codecs_ = send_codecs; |
| audio_recv_codecs_ = recv_codecs; |
| ComputeAudioCodecsIntersectionAndUnion(); |
| } |
| |
| SessionDescription* MediaSessionDescriptionFactory::CreateOffer( |
| const MediaSessionOptions& session_options, |
| const SessionDescription* current_description) const { |
| std::unique_ptr<SessionDescription> offer(new SessionDescription()); |
| |
| StreamParamsVec current_streams; |
| GetCurrentStreamParams(current_description, ¤t_streams); |
| |
| AudioCodecs offer_audio_codecs; |
| VideoCodecs offer_video_codecs; |
| DataCodecs offer_data_codecs; |
| GetCodecsForOffer(current_description, &offer_audio_codecs, |
| &offer_video_codecs, &offer_data_codecs); |
| |
| if (!session_options.vad_enabled) { |
| // If application doesn't want CN codecs in offer. |
| StripCNCodecs(&offer_audio_codecs); |
| } |
| FilterDataCodecs(&offer_data_codecs, |
| session_options.data_channel_type == DCT_SCTP); |
| |
| RtpHeaderExtensions audio_rtp_extensions; |
| RtpHeaderExtensions video_rtp_extensions; |
| GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions, |
| &video_rtp_extensions); |
| |
| // Must have options for each existing section. |
| if (current_description) { |
| RTC_DCHECK(current_description->contents().size() <= |
| session_options.media_description_options.size()); |
| } |
| |
| // Iterate through the media description options, matching with existing media |
| // descriptions in |current_description|. |
| size_t msection_index = 0; |
| for (const MediaDescriptionOptions& media_description_options : |
| session_options.media_description_options) { |
| const ContentInfo* current_content = nullptr; |
| if (current_description && |
| msection_index < current_description->contents().size()) { |
| current_content = ¤t_description->contents()[msection_index]; |
| // Media type must match unless this media section is being recycled. |
| RTC_DCHECK(current_content->rejected || |
| IsMediaContentOfType(current_content, |
| media_description_options.type)); |
| } |
| switch (media_description_options.type) { |
| case MEDIA_TYPE_AUDIO: |
| if (!AddAudioContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| audio_rtp_extensions, offer_audio_codecs, |
| ¤t_streams, offer.get())) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_VIDEO: |
| if (!AddVideoContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| video_rtp_extensions, offer_video_codecs, |
| ¤t_streams, offer.get())) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_DATA: |
| if (!AddDataContentForOffer(media_description_options, session_options, |
| current_content, current_description, |
| offer_data_codecs, ¤t_streams, |
| offer.get())) { |
| return nullptr; |
| } |
| break; |
| default: |
| RTC_NOTREACHED(); |
| } |
| ++msection_index; |
| } |
| |
| // Bundle the contents together, if we've been asked to do so, and update any |
| // parameters that need to be tweaked for BUNDLE. |
| if (session_options.bundle_enabled && offer->contents().size() > 0u) { |
| ContentGroup offer_bundle(GROUP_TYPE_BUNDLE); |
| for (const ContentInfo& content : offer->contents()) { |
| // TODO(deadbeef): There are conditions that make bundling two media |
| // descriptions together illegal. For example, they use the same payload |
| // type to represent different codecs, or same IDs for different header |
| // extensions. We need to detect this and not try to bundle those media |
| // descriptions together. |
| offer_bundle.AddContentName(content.name); |
| } |
| offer->AddGroup(offer_bundle); |
| if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateOffer failed to UpdateTransportInfoForBundle."; |
| return nullptr; |
| } |
| if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) { |
| RTC_LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle."; |
| return nullptr; |
| } |
| } |
| |
| // The following determines how to signal MSIDs to ensure compatibility with |
| // older endpoints (in particular, older Plan B endpoints). |
| if (session_options.is_unified_plan) { |
| // Be conservative and signal using both a=msid and a=ssrc lines. Unified |
| // Plan answerers will look at a=msid and Plan B answerers will look at the |
| // a=ssrc MSID line. |
| offer->set_msid_signaling(cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute); |
| } else { |
| // Plan B always signals MSID using a=ssrc lines. |
| offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute); |
| } |
| |
| return offer.release(); |
| } |
| |
| SessionDescription* MediaSessionDescriptionFactory::CreateAnswer( |
| const SessionDescription* offer, |
| const MediaSessionOptions& session_options, |
| const SessionDescription* current_description) const { |
| if (!offer) { |
| return nullptr; |
| } |
| // The answer contains the intersection of the codecs in the offer with the |
| // codecs we support. As indicated by XEP-0167, we retain the same payload ids |
| // from the offer in the answer. |
| std::unique_ptr<SessionDescription> answer(new SessionDescription()); |
| |
| StreamParamsVec current_streams; |
| GetCurrentStreamParams(current_description, ¤t_streams); |
| |
| // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE |
| // group in the answer with the appropriate content names. |
| const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE); |
| ContentGroup answer_bundle(GROUP_TYPE_BUNDLE); |
| // Transport info shared by the bundle group. |
| std::unique_ptr<TransportInfo> bundle_transport; |
| |
| // Get list of all possible codecs that respects existing payload type |
| // mappings and uses a single payload type space. |
| // |
| // Note that these lists may be further filtered for each m= section; this |
| // step is done just to establish the payload type mappings shared by all |
| // sections. |
| AudioCodecs answer_audio_codecs; |
| VideoCodecs answer_video_codecs; |
| DataCodecs answer_data_codecs; |
| GetCodecsForAnswer(current_description, offer, &answer_audio_codecs, |
| &answer_video_codecs, &answer_data_codecs); |
| |
| if (!session_options.vad_enabled) { |
| // If application doesn't want CN codecs in answer. |
| StripCNCodecs(&answer_audio_codecs); |
| } |
| FilterDataCodecs(&answer_data_codecs, |
| session_options.data_channel_type == DCT_SCTP); |
| |
| // Must have options for exactly as many sections as in the offer. |
| RTC_DCHECK(offer->contents().size() == |
| session_options.media_description_options.size()); |
| // Iterate through the media description options, matching with existing |
| // media descriptions in |current_description|. |
| size_t msection_index = 0; |
| for (const MediaDescriptionOptions& media_description_options : |
| session_options.media_description_options) { |
| const ContentInfo* offer_content = &offer->contents()[msection_index]; |
| // Media types and MIDs must match between the remote offer and the |
| // MediaDescriptionOptions. |
| RTC_DCHECK( |
| IsMediaContentOfType(offer_content, media_description_options.type)); |
| RTC_DCHECK(media_description_options.mid == offer_content->name); |
| const ContentInfo* current_content = nullptr; |
| if (current_description && |
| msection_index < current_description->contents().size()) { |
| current_content = ¤t_description->contents()[msection_index]; |
| } |
| switch (media_description_options.type) { |
| case MEDIA_TYPE_AUDIO: |
| if (!AddAudioContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, |
| bundle_transport.get(), answer_audio_codecs, ¤t_streams, |
| answer.get())) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_VIDEO: |
| if (!AddVideoContentForAnswer( |
| media_description_options, session_options, offer_content, |
| offer, current_content, current_description, |
| bundle_transport.get(), answer_video_codecs, ¤t_streams, |
| answer.get())) { |
| return nullptr; |
| } |
| break; |
| case MEDIA_TYPE_DATA: |
| if (!AddDataContentForAnswer(media_description_options, session_options, |
| offer_content, offer, current_content, |
| current_description, |
| bundle_transport.get(), answer_data_codecs, |
| ¤t_streams, answer.get())) { |
| return nullptr; |
| } |
| break; |
| default: |
| RTC_NOTREACHED(); |
| } |
| ++msection_index; |
| // See if we can add the newly generated m= section to the BUNDLE group in |
| // the answer. |
| ContentInfo& added = answer->contents().back(); |
| if (!added.rejected && session_options.bundle_enabled && offer_bundle && |
| offer_bundle->HasContentName(added.name)) { |
| answer_bundle.AddContentName(added.name); |
| bundle_transport.reset( |
| new TransportInfo(*answer->GetTransportInfoByName(added.name))); |
| } |
| } |
| |
| // Only put BUNDLE group in answer if nonempty. |
| if (answer_bundle.FirstContentName()) { |
| answer->AddGroup(answer_bundle); |
| |
| // Share the same ICE credentials and crypto params across all contents, |
| // as BUNDLE requires. |
| if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateAnswer failed to UpdateTransportInfoForBundle."; |
| return NULL; |
| } |
| |
| if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) { |
| RTC_LOG(LS_ERROR) |
| << "CreateAnswer failed to UpdateCryptoParamsForBundle."; |
| return NULL; |
| } |
| } |
| |
| // The following determines how to signal MSIDs to ensure compatibility with |
| // older endpoints (in particular, older Plan B endpoints). |
| if (session_options.is_unified_plan) { |
| // Unified Plan needs to look at what the offer included to find the most |
| // compatible answer. |
| if (offer->msid_signaling() == 0) { |
| // We end up here in one of three cases: |
| // 1. An empty offer. We'll reply with an empty answer so it doesn't |
| // matter what we pick here. |
| // 2. A data channel only offer. We won't add any MSIDs to the answer so |
| // it also doesn't matter what we pick here. |
| // 3. Media that's either sendonly or inactive from the remote endpoint. |
| // We don't have any information to say whether the endpoint is Plan B |
| // or Unified Plan, so be conservative and send both. |
| answer->set_msid_signaling(cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute); |
| } else if (offer->msid_signaling() == |
| (cricket::kMsidSignalingMediaSection | |
| cricket::kMsidSignalingSsrcAttribute)) { |
| // If both a=msid and a=ssrc MSID signaling methods were used, we're |
| // probably talking to a Unified Plan endpoint so respond with just |
| // a=msid. |
| answer->set_msid_signaling(cricket::kMsidSignalingMediaSection); |
| } else { |
| // Otherwise, it's clear which method the offerer is using so repeat that |
| // back to them. |
| answer->set_msid_signaling(offer->msid_signaling()); |
| } |
| } else { |
| // Plan B always signals MSID using a=ssrc lines. |
| answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute); |
| } |
| |
| return answer.release(); |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer( |
| const RtpTransceiverDirection& direction) const { |
| switch (direction) { |
| // If stream is inactive - generate list as if sendrecv. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kInactive: |
| return audio_sendrecv_codecs_; |
| case RtpTransceiverDirection::kSendOnly: |
| return audio_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return audio_recv_codecs_; |
| } |
| RTC_NOTREACHED(); |
| return audio_sendrecv_codecs_; |
| } |
| |
| const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer( |
| const RtpTransceiverDirection& offer, |
| const RtpTransceiverDirection& answer) const { |
| switch (answer) { |
| // For inactive and sendrecv answers, generate lists as if we were to accept |
| // the offer's direction. See RFC 3264 Section 6.1. |
| case RtpTransceiverDirection::kSendRecv: |
| case RtpTransceiverDirection::kInactive: |
| return GetAudioCodecsForOffer( |
| webrtc::RtpTransceiverDirectionReversed(offer)); |
| case RtpTransceiverDirection::kSendOnly: |
| return audio_send_codecs_; |
| case RtpTransceiverDirection::kRecvOnly: |
| return audio_recv_codecs_; |
| } |
| RTC_NOTREACHED(); |
| return audio_sendrecv_codecs_; |
| } |
| |
| void MergeCodecsFromDescription(const SessionDescription* description, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| DataCodecs* data_codecs, |
| UsedPayloadTypes* used_pltypes) { |
| RTC_DCHECK(description); |
| for (const ContentInfo& content : description->contents()) { |
| if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content.media_description()->as_audio(); |
| MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes); |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content.media_description()->as_video(); |
| MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes); |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) { |
| const DataContentDescription* data = |
| content.media_description()->as_data(); |
| MergeCodecs<DataCodec>(data->codecs(), data_codecs, used_pltypes); |
| } |
| } |
| } |
| |
| // Getting codecs for an offer involves these steps: |
| // |
| // 1. Construct payload type -> codec mappings for current description. |
| // 2. Add any reference codecs that weren't already present |
| // 3. For each individual media description (m= section), filter codecs based |
| // on the directional attribute (happens in another method). |
| void MediaSessionDescriptionFactory::GetCodecsForOffer( |
| const SessionDescription* current_description, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| DataCodecs* data_codecs) const { |
| UsedPayloadTypes used_pltypes; |
| audio_codecs->clear(); |
| video_codecs->clear(); |
| data_codecs->clear(); |
| |
| // First - get all codecs from the current description if the media type |
| // is used. Add them to |used_pltypes| so the payload type is not reused if a |
| // new media type is added. |
| if (current_description) { |
| MergeCodecsFromDescription(current_description, audio_codecs, video_codecs, |
| data_codecs, &used_pltypes); |
| } |
| |
| // Add our codecs that are not in |current_description|. |
| MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes); |
| MergeCodecs<VideoCodec>(video_codecs_, video_codecs, &used_pltypes); |
| MergeCodecs<DataCodec>(data_codecs_, data_codecs, &used_pltypes); |
| } |
| |
| // Getting codecs for an answer involves these steps: |
| // |
| // 1. Construct payload type -> codec mappings for current description. |
| // 2. Add any codecs from the offer that weren't already present. |
| // 3. Add any remaining codecs that weren't already present. |
| // 4. For each individual media description (m= section), filter codecs based |
| // on the directional attribute (happens in another method). |
| void MediaSessionDescriptionFactory::GetCodecsForAnswer( |
| const SessionDescription* current_description, |
| const SessionDescription* remote_offer, |
| AudioCodecs* audio_codecs, |
| VideoCodecs* video_codecs, |
| DataCodecs* data_codecs) const { |
| UsedPayloadTypes used_pltypes; |
| audio_codecs->clear(); |
| video_codecs->clear(); |
| data_codecs->clear(); |
| |
| // First - get all codecs from the current description if the media type |
| // is used. Add them to |used_pltypes| so the payload type is not reused if a |
| // new media type is added. |
| if (current_description) { |
| MergeCodecsFromDescription(current_description, audio_codecs, video_codecs, |
| data_codecs, &used_pltypes); |
| } |
| |
| // Second - filter out codecs that we don't support at all and should ignore. |
| AudioCodecs filtered_offered_audio_codecs; |
| VideoCodecs filtered_offered_video_codecs; |
| DataCodecs filtered_offered_data_codecs; |
| for (const ContentInfo& content : remote_offer->contents()) { |
| if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content.media_description()->as_audio(); |
| for (const AudioCodec& offered_audio_codec : audio->codecs()) { |
| if (!FindMatchingCodec<AudioCodec>(audio->codecs(), |
| filtered_offered_audio_codecs, |
| offered_audio_codec, nullptr) && |
| FindMatchingCodec<AudioCodec>(audio->codecs(), all_audio_codecs_, |
| offered_audio_codec, nullptr)) { |
| filtered_offered_audio_codecs.push_back(offered_audio_codec); |
| } |
| } |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content.media_description()->as_video(); |
| for (const VideoCodec& offered_video_codec : video->codecs()) { |
| if (!FindMatchingCodec<VideoCodec>(video->codecs(), |
| filtered_offered_video_codecs, |
| offered_video_codec, nullptr) && |
| FindMatchingCodec<VideoCodec>(video->codecs(), video_codecs_, |
| offered_video_codec, nullptr)) { |
| filtered_offered_video_codecs.push_back(offered_video_codec); |
| } |
| } |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) { |
| const DataContentDescription* data = |
| content.media_description()->as_data(); |
| for (const DataCodec& offered_data_codec : data->codecs()) { |
| if (!FindMatchingCodec<DataCodec>(data->codecs(), |
| filtered_offered_data_codecs, |
| offered_data_codec, nullptr) && |
| FindMatchingCodec<DataCodec>(data->codecs(), data_codecs_, |
| offered_data_codec, nullptr)) { |
| filtered_offered_data_codecs.push_back(offered_data_codec); |
| } |
| } |
| } |
| } |
| |
| // Add codecs that are not in |current_description| but were in |
| // |remote_offer|. |
| MergeCodecs<AudioCodec>(filtered_offered_audio_codecs, audio_codecs, |
| &used_pltypes); |
| MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs, |
| &used_pltypes); |
| MergeCodecs<DataCodec>(filtered_offered_data_codecs, data_codecs, |
| &used_pltypes); |
| } |
| |
| void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer( |
| const SessionDescription* current_description, |
| RtpHeaderExtensions* offer_audio_extensions, |
| RtpHeaderExtensions* offer_video_extensions) const { |
| // All header extensions allocated from the same range to avoid potential |
| // issues when using BUNDLE. |
| UsedRtpHeaderExtensionIds used_ids; |
| RtpHeaderExtensions all_regular_extensions; |
| RtpHeaderExtensions all_encrypted_extensions; |
| offer_audio_extensions->clear(); |
| offer_video_extensions->clear(); |
| |
| // First - get all extensions from the current description if the media type |
| // is used. |
| // Add them to |used_ids| so the local ids are not reused if a new media |
| // type is added. |
| if (current_description) { |
| for (const ContentInfo& content : current_description->contents()) { |
| if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) { |
| const AudioContentDescription* audio = |
| content.media_description()->as_audio(); |
| MergeRtpHdrExts(audio->rtp_header_extensions(), offer_audio_extensions, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) { |
| const VideoContentDescription* video = |
| content.media_description()->as_video(); |
| MergeRtpHdrExts(video->rtp_header_extensions(), offer_video_extensions, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| } |
| } |
| } |
| |
| // Add our default RTP header extensions that are not in |
| // |current_description|. |
| MergeRtpHdrExts(audio_rtp_header_extensions(), offer_audio_extensions, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| MergeRtpHdrExts(video_rtp_header_extensions(), offer_video_extensions, |
| &all_regular_extensions, &all_encrypted_extensions, |
| &used_ids); |
| |
| // TODO(jbauch): Support adding encrypted header extensions to existing |
| // sessions. |
| if (enable_encrypted_rtp_header_extensions_ && !current_description) { |
| AddEncryptedVersionsOfHdrExts(offer_audio_extensions, |
| &all_encrypted_extensions, &used_ids); |
| AddEncryptedVersionsOfHdrExts(offer_video_extensions, |
| &all_encrypted_extensions, &used_ids); |
| } |
| } |
| |
| bool MediaSessionDescriptionFactory::AddTransportOffer( |
| const std::string& content_name, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| SessionDescription* offer_desc) const { |
| if (!transport_desc_factory_) |
| return false; |
| const TransportDescription* current_tdesc = |
| GetTransportDescription(content_name, current_desc); |
| std::unique_ptr<TransportDescription> new_tdesc( |
| transport_desc_factory_->CreateOffer(transport_options, current_tdesc)); |
| bool ret = (new_tdesc.get() != NULL && |
| offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc))); |
| if (!ret) { |
| RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name=" |
| << content_name; |
| } |
| return ret; |
| } |
| |
| TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer( |
| const std::string& content_name, |
| const SessionDescription* offer_desc, |
| const TransportOptions& transport_options, |
| const SessionDescription* current_desc, |
| bool require_transport_attributes) const { |
| if (!transport_desc_factory_) |
| return NULL; |
| const TransportDescription* offer_tdesc = |
| GetTransportDescription(content_name, offer_desc); |
| const TransportDescription* current_tdesc = |
| GetTransportDescription(content_name, current_desc); |
| return transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options, |
| require_transport_attributes, |
| current_tdesc); |
| } |
| |
| bool MediaSessionDescriptionFactory::AddTransportAnswer( |
| const std::string& content_name, |
| const TransportDescription& transport_desc, |
| SessionDescription* answer_desc) const { |
| if (!answer_desc->AddTransportInfo(TransportInfo(content_name, |
| transport_desc))) { |
| RTC_LOG(LS_ERROR) << "Failed to AddTransportAnswer, content name=" |
| << content_name; |
| return false; |
| } |
| return true; |
| } |
| |
| // |audio_codecs| = set of all possible codecs that can be used, with correct |
| // payload type mappings |
| // |
| // |supported_audio_codecs| = set of codecs that are supported for the direction |
| // of this m= section |
| // |
| // acd->codecs() = set of previously negotiated codecs for this m= section |
| // |
| // The payload types should come from audio_codecs, but the order should come |
| // from acd->codecs() and then supported_codecs, to ensure that re-offers don't |
| // change existing codec priority, and that new codecs are added with the right |
| // priority. |
| bool MediaSessionDescriptionFactory::AddAudioContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& audio_rtp_extensions, |
| const AudioCodecs& audio_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const { |
| // Filter audio_codecs (which includes all codecs, with correctly remapped |
| // payload types) based on transceiver direction. |
| const AudioCodecs& supported_audio_codecs = |
| GetAudioCodecsForOffer(media_description_options.direction); |
| |
| AudioCodecs filtered_codecs; |
| // Add the codecs from current content if it exists and is not being recycled. |
| if (current_content && !current_content->rejected) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* acd = |
| current_content->media_description()->as_audio(); |
| for (const AudioCodec& codec : acd->codecs()) { |
| if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported audio codecs. |
| AudioCodec found_codec; |
| for (const AudioCodec& codec : supported_audio_codecs) { |
| if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs, |
| codec, &found_codec) && |
| !FindMatchingCodec<AudioCodec>(supported_audio_codecs, filtered_codecs, |
| codec, nullptr)) { |
| // Use the |found_codec| from |audio_codecs| because it has the correctly |
| // mapped payload type. |
| filtered_codecs.push_back(found_codec); |
| } |
| } |
| |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| |
| std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription()); |
| std::vector<std::string> crypto_suites; |
| GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options, |
| &crypto_suites); |
| if (!CreateMediaContentOffer( |
| media_description_options.sender_options, session_options, |
| filtered_codecs, sdes_policy, GetCryptos(current_content), |
| crypto_suites, audio_rtp_extensions, current_streams, audio.get())) { |
| return false; |
| } |
| |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| SetMediaProtocol(secure_transport, audio.get()); |
| |
| audio->set_direction(media_description_options.direction); |
| |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp, |
| media_description_options.stopped, audio.release()); |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc)) { |
| return false; |
| } |
| |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddVideoContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const RtpHeaderExtensions& video_rtp_extensions, |
| const VideoCodecs& video_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const { |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| |
| std::unique_ptr<VideoContentDescription> video(new VideoContentDescription()); |
| std::vector<std::string> crypto_suites; |
| GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options, |
| &crypto_suites); |
| |
| VideoCodecs filtered_codecs; |
| // Add the codecs from current content if it exists and is not being recycled. |
| if (current_content && !current_content->rejected) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* vcd = |
| current_content->media_description()->as_video(); |
| for (const VideoCodec& codec : vcd->codecs()) { |
| if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported video codecs. |
| VideoCodec found_codec; |
| for (const VideoCodec& codec : video_codecs_) { |
| if (FindMatchingCodec<VideoCodec>(video_codecs_, video_codecs, codec, |
| &found_codec) && |
| !FindMatchingCodec<VideoCodec>(video_codecs_, filtered_codecs, codec, |
| nullptr)) { |
| // Use the |found_codec| from |video_codecs| because it has the correctly |
| // mapped payload type. |
| filtered_codecs.push_back(found_codec); |
| } |
| } |
| |
| if (!CreateMediaContentOffer( |
| media_description_options.sender_options, session_options, |
| filtered_codecs, sdes_policy, GetCryptos(current_content), |
| crypto_suites, video_rtp_extensions, current_streams, video.get())) { |
| return false; |
| } |
| |
| video->set_bandwidth(kAutoBandwidth); |
| |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| SetMediaProtocol(secure_transport, video.get()); |
| |
| video->set_direction(media_description_options.direction); |
| |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp, |
| media_description_options.stopped, video.release()); |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc)) { |
| return false; |
| } |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddDataContentForOffer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const DataCodecs& data_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* desc) const { |
| bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED); |
| |
| std::unique_ptr<DataContentDescription> data(new DataContentDescription()); |
| bool is_sctp = (session_options.data_channel_type == DCT_SCTP); |
| // If the DataChannel type is not specified, use the DataChannel type in |
| // the current description. |
| if (session_options.data_channel_type == DCT_NONE && current_content) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_DATA)); |
| is_sctp = (current_content->media_description()->protocol() == |
| kMediaProtocolSctp); |
| } |
| |
| cricket::SecurePolicy sdes_policy = |
| IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED |
| : secure(); |
| std::vector<std::string> crypto_suites; |
| if (is_sctp) { |
| // SDES doesn't make sense for SCTP, so we disable it, and we only |
| // get SDES crypto suites for RTP-based data channels. |
| sdes_policy = cricket::SEC_DISABLED; |
| // Unlike SetMediaProtocol below, we need to set the protocol |
| // before we call CreateMediaContentOffer. Otherwise, |
| // CreateMediaContentOffer won't know this is SCTP and will |
| // generate SSRCs rather than SIDs. |
| // TODO(deadbeef): Offer kMediaProtocolUdpDtlsSctp (or TcpDtlsSctp), once |
| // it's safe to do so. Older versions of webrtc would reject these |
| // protocols; see https://bugs.chromium.org/p/webrtc/issues/detail?id=7706. |
| data->set_protocol( |
| secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp); |
| } else { |
| GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options, |
| &crypto_suites); |
| } |
| |
| // Even SCTP uses a "codec". |
| if (!CreateMediaContentOffer( |
| media_description_options.sender_options, session_options, |
| data_codecs, sdes_policy, GetCryptos(current_content), crypto_suites, |
| RtpHeaderExtensions(), current_streams, data.get())) { |
| return false; |
| } |
| |
| if (is_sctp) { |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kSctp, |
| data.release()); |
| } else { |
| data->set_bandwidth(kDataMaxBandwidth); |
| SetMediaProtocol(secure_transport, data.get()); |
| desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp, |
| media_description_options.stopped, data.release()); |
| } |
| if (!AddTransportOffer(media_description_options.mid, |
| media_description_options.transport_options, |
| current_description, desc)) { |
| return false; |
| } |
| return true; |
| } |
| |
| // |audio_codecs| = set of all possible codecs that can be used, with correct |
| // payload type mappings |
| // |
| // |supported_audio_codecs| = set of codecs that are supported for the direction |
| // of this m= section |
| // |
| // acd->codecs() = set of previously negotiated codecs for this m= section |
| // |
| // The payload types should come from audio_codecs, but the order should come |
| // from acd->codecs() and then supported_codecs, to ensure that re-offers don't |
| // change existing codec priority, and that new codecs are added with the right |
| // priority. |
| bool MediaSessionDescriptionFactory::AddAudioContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const AudioCodecs& audio_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const { |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* offer_audio_description = |
| offer_content->media_description()->as_audio(); |
| |
| std::unique_ptr<TransportDescription> audio_transport( |
| CreateTransportAnswer(media_description_options.mid, offer_description, |
| media_description_options.transport_options, |
| current_description, bundle_transport != nullptr)); |
| if (!audio_transport) { |
| return false; |
| } |
| |
| // Pick codecs based on the requested communications direction in the offer |
| // and the selected direction in the answer. |
| // Note these will be filtered one final time in CreateMediaContentAnswer. |
| auto wants_rtd = media_description_options.direction; |
| auto offer_rtd = offer_audio_description->direction(); |
| auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd); |
| AudioCodecs supported_audio_codecs = |
| GetAudioCodecsForAnswer(offer_rtd, answer_rtd); |
| |
| AudioCodecs filtered_codecs; |
| // Add the codecs from current content if it exists and is not being recycled. |
| if (current_content && !current_content->rejected) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)); |
| const AudioContentDescription* acd = |
| current_content->media_description()->as_audio(); |
| for (const AudioCodec& codec : acd->codecs()) { |
| if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported audio codecs. |
| for (const AudioCodec& codec : supported_audio_codecs) { |
| if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs, |
| codec, nullptr) && |
| !FindMatchingCodec<AudioCodec>(supported_audio_codecs, filtered_codecs, |
| codec, nullptr)) { |
| // We should use the local codec with local parameters and the codec id |
| // would be correctly mapped in |NegotiateCodecs|. |
| filtered_codecs.push_back(codec); |
| } |
| } |
| |
| bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| std::unique_ptr<AudioContentDescription> audio_answer( |
| new AudioContentDescription()); |
| // Do not require or create SDES cryptos if DTLS is used. |
| cricket::SecurePolicy sdes_policy = |
| audio_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| if (!CreateMediaContentAnswer( |
| offer_audio_description, media_description_options, session_options, |
| filtered_codecs, sdes_policy, GetCryptos(current_content), |
| audio_rtp_extensions_, enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, audio_answer.get())) { |
| return false; // Fails the session setup. |
| } |
| |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : audio_transport->secure(); |
| bool rejected = media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO, |
| audio_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(audio_transport.get()), answer)) { |
| return false; |
| } |
| |
| if (rejected) { |
| RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid |
| << "' being rejected in answer."; |
| } |
| |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, audio_answer.release()); |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddVideoContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const VideoCodecs& video_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const { |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* offer_video_description = |
| offer_content->media_description()->as_video(); |
| |
| std::unique_ptr<TransportDescription> video_transport( |
| CreateTransportAnswer(media_description_options.mid, offer_description, |
| media_description_options.transport_options, |
| current_description, bundle_transport != nullptr)); |
| if (!video_transport) { |
| return false; |
| } |
| |
| VideoCodecs filtered_codecs; |
| // Add the codecs from current content if it exists and is not being recycled. |
| if (current_content && !current_content->rejected) { |
| RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)); |
| const VideoContentDescription* vcd = |
| current_content->media_description()->as_video(); |
| for (const VideoCodec& codec : vcd->codecs()) { |
| if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec, |
| nullptr)) { |
| filtered_codecs.push_back(codec); |
| } |
| } |
| } |
| // Add other supported video codecs. |
| for (const VideoCodec& codec : video_codecs_) { |
| if (FindMatchingCodec<VideoCodec>(video_codecs_, video_codecs, codec, |
| nullptr) && |
| !FindMatchingCodec<VideoCodec>(video_codecs_, filtered_codecs, codec, |
| nullptr)) { |
| // We should use the local codec with local parameters and the codec id |
| // would be correctly mapped in |NegotiateCodecs|. |
| filtered_codecs.push_back(codec); |
| } |
| } |
| |
| bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| |
| std::unique_ptr<VideoContentDescription> video_answer( |
| new VideoContentDescription()); |
| // Do not require or create SDES cryptos if DTLS is used. |
| cricket::SecurePolicy sdes_policy = |
| video_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| if (!CreateMediaContentAnswer( |
| offer_video_description, media_description_options, session_options, |
| filtered_codecs, sdes_policy, GetCryptos(current_content), |
| video_rtp_extensions_, enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, video_answer.get())) { |
| return false; // Failed the sessin setup. |
| } |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : video_transport->secure(); |
| bool rejected = media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO, |
| video_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(video_transport.get()), answer)) { |
| return false; |
| } |
| |
| if (!rejected) { |
| video_answer->set_bandwidth(kAutoBandwidth); |
| } else { |
| RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid |
| << "' being rejected in answer."; |
| } |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, video_answer.release()); |
| return true; |
| } |
| |
| bool MediaSessionDescriptionFactory::AddDataContentForAnswer( |
| const MediaDescriptionOptions& media_description_options, |
| const MediaSessionOptions& session_options, |
| const ContentInfo* offer_content, |
| const SessionDescription* offer_description, |
| const ContentInfo* current_content, |
| const SessionDescription* current_description, |
| const TransportInfo* bundle_transport, |
| const DataCodecs& data_codecs, |
| StreamParamsVec* current_streams, |
| SessionDescription* answer) const { |
| std::unique_ptr<TransportDescription> data_transport( |
| CreateTransportAnswer(media_description_options.mid, offer_description, |
| media_description_options.transport_options, |
| current_description, bundle_transport != nullptr)); |
| if (!data_transport) { |
| return false; |
| } |
| |
| std::unique_ptr<DataContentDescription> data_answer( |
| new DataContentDescription()); |
| // Do not require or create SDES cryptos if DTLS is used. |
| cricket::SecurePolicy sdes_policy = |
| data_transport->secure() ? cricket::SEC_DISABLED : secure(); |
| bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && |
| session_options.bundle_enabled; |
| RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_DATA)); |
| const DataContentDescription* offer_data_description = |
| offer_content->media_description()->as_data(); |
| if (!CreateMediaContentAnswer( |
| offer_data_description, media_description_options, session_options, |
| data_codecs, sdes_policy, GetCryptos(current_content), |
| RtpHeaderExtensions(), enable_encrypted_rtp_header_extensions_, |
| current_streams, bundle_enabled, data_answer.get())) { |
| return false; // Fails the session setup. |
| } |
| |
| // Respond with sctpmap if the offer uses sctpmap. |
| bool offer_uses_sctpmap = offer_data_description->use_sctpmap(); |
| data_answer->set_use_sctpmap(offer_uses_sctpmap); |
| |
| bool secure = bundle_transport ? bundle_transport->description.secure() |
| : data_transport->secure(); |
| |
| bool rejected = session_options.data_channel_type == DCT_NONE || |
| media_description_options.stopped || |
| offer_content->rejected || |
| !IsMediaProtocolSupported(MEDIA_TYPE_DATA, |
| data_answer->protocol(), secure); |
| if (!AddTransportAnswer(media_description_options.mid, |
| *(data_transport.get()), answer)) { |
| return false; |
| } |
| |
| if (!rejected) { |
| data_answer->set_bandwidth(kDataMaxBandwidth); |
| } else { |
| // RFC 3264 |
| // The answer MUST contain the same number of m-lines as the offer. |
| RTC_LOG(LS_INFO) << "Data is not supported in the answer."; |
| } |
| answer->AddContent(media_description_options.mid, offer_content->type, |
| rejected, data_answer.release()); |
| return true; |
| } |
| |
| void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() { |
| audio_sendrecv_codecs_.clear(); |
| all_audio_codecs_.clear(); |
| // Compute the audio codecs union. |
| for (const AudioCodec& send : audio_send_codecs_) { |
| all_audio_codecs_.push_back(send); |
| if (!FindMatchingCodec<AudioCodec>(audio_send_codecs_, audio_recv_codecs_, |
| send, nullptr)) { |
| // It doesn't make sense to have an RTX codec we support sending but not |
| // receiving. |
| RTC_DCHECK(!IsRtxCodec(send)); |
| } |
| } |
| for (const AudioCodec& recv : audio_recv_codecs_) { |
| if (!FindMatchingCodec<AudioCodec>(audio_recv_codecs_, audio_send_codecs_, |
| recv, nullptr)) { |
| all_audio_codecs_.push_back(recv); |
| } |
| } |
| // Use NegotiateCodecs to merge our codec lists, since the operation is |
| // essentially the same. Put send_codecs as the offered_codecs, which is the |
| // order we'd like to follow. The reasoning is that encoding is usually more |
| // expensive than decoding, and prioritizing a codec in the send list probably |
| // means it's a codec we can handle efficiently. |
| NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_, |
| &audio_sendrecv_codecs_); |
| } |
| |
| bool IsMediaContent(const ContentInfo* content) { |
| return (content && (content->type == MediaProtocolType::kRtp || |
| content->type == MediaProtocolType::kSctp)); |
| } |
| |
| bool IsAudioContent(const ContentInfo* content) { |
| return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO); |
| } |
| |
| bool IsVideoContent(const ContentInfo* content) { |
| return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO); |
| } |
| |
| bool IsDataContent(const ContentInfo* content) { |
| return IsMediaContentOfType(content, MEDIA_TYPE_DATA); |
| } |
| |
| const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, |
| MediaType media_type) { |
| for (const ContentInfo& content : contents) { |
| if (IsMediaContentOfType(&content, media_type)) { |
| return &content; |
| } |
| } |
| return nullptr; |
| } |
| |
| const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO); |
| } |
| |
| const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO); |
| } |
| |
| const ContentInfo* GetFirstDataContent(const ContentInfos& contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_DATA); |
| } |
| |
| const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc, |
| MediaType media_type) { |
| if (sdesc == nullptr) { |
| return nullptr; |
| } |
| |
| return GetFirstMediaContent(sdesc->contents(), media_type); |
| } |
| |
| const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO); |
| } |
| |
| const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO); |
| } |
| |
| const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA); |
| } |
| |
| const MediaContentDescription* GetFirstMediaContentDescription( |
| const SessionDescription* sdesc, MediaType media_type) { |
| const ContentInfo* content = GetFirstMediaContent(sdesc, media_type); |
| return (content ? content->media_description() : nullptr); |
| } |
| |
| const AudioContentDescription* GetFirstAudioContentDescription( |
| const SessionDescription* sdesc) { |
| return static_cast<const AudioContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO)); |
| } |
| |
| const VideoContentDescription* GetFirstVideoContentDescription( |
| const SessionDescription* sdesc) { |
| return static_cast<const VideoContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO)); |
| } |
| |
| const DataContentDescription* GetFirstDataContentDescription( |
| const SessionDescription* sdesc) { |
| return static_cast<const DataContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA)); |
| } |
| |
| // |
| // Non-const versions of the above functions. |
| // |
| |
| ContentInfo* GetFirstMediaContent(ContentInfos* contents, |
| MediaType media_type) { |
| for (ContentInfo& content : *contents) { |
| if (IsMediaContentOfType(&content, media_type)) { |
| return &content; |
| } |
| } |
| return nullptr; |
| } |
| |
| ContentInfo* GetFirstAudioContent(ContentInfos* contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO); |
| } |
| |
| ContentInfo* GetFirstVideoContent(ContentInfos* contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO); |
| } |
| |
| ContentInfo* GetFirstDataContent(ContentInfos* contents) { |
| return GetFirstMediaContent(contents, MEDIA_TYPE_DATA); |
| } |
| |
| ContentInfo* GetFirstMediaContent(SessionDescription* sdesc, |
| MediaType media_type) { |
| if (sdesc == nullptr) { |
| return nullptr; |
| } |
| |
| return GetFirstMediaContent(&sdesc->contents(), media_type); |
| } |
| |
| ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO); |
| } |
| |
| ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO); |
| } |
| |
| ContentInfo* GetFirstDataContent(SessionDescription* sdesc) { |
| return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA); |
| } |
| |
| MediaContentDescription* GetFirstMediaContentDescription( |
| SessionDescription* sdesc, |
| MediaType media_type) { |
| ContentInfo* content = GetFirstMediaContent(sdesc, media_type); |
| return (content ? content->media_description() : nullptr); |
| } |
| |
| AudioContentDescription* GetFirstAudioContentDescription( |
| SessionDescription* sdesc) { |
| return static_cast<AudioContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO)); |
| } |
| |
| VideoContentDescription* GetFirstVideoContentDescription( |
| SessionDescription* sdesc) { |
| return static_cast<VideoContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO)); |
| } |
| |
| DataContentDescription* GetFirstDataContentDescription( |
| SessionDescription* sdesc) { |
| return static_cast<DataContentDescription*>( |
| GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA)); |
| } |
| |
| } // namespace cricket |