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webrtc / src / c20978e581a3f71d96ef2f6078fbad350922fff8 / . / webrtc / pc / mediasession.cc
blob: de8f72c92612714268570a7a2403e2c6d16827c1 [file] [log] [blame]
/*
* 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 "webrtc/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 "webrtc/common_types.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/h264_profile_level_id.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/srtpfilter.h"
#include "webrtc/rtc_base/base64.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/helpers.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/stringutils.h"
namespace {
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);
}
RtpTransceiverDirection RtpTransceiverDirection::FromMediaContentDirection(
MediaContentDirection md) {
const bool send = (md == MD_SENDRECV || md == MD_SENDONLY);
const bool recv = (md == MD_SENDRECV || md == MD_RECVONLY);
return RtpTransceiverDirection(send, recv);
}
MediaContentDirection RtpTransceiverDirection::ToMediaContentDirection() const {
if (send && recv) {
return MD_SENDRECV;
} else if (send) {
return MD_SENDONLY;
} else if (recv) {
return MD_RECVONLY;
}
return MD_INACTIVE;
}
RtpTransceiverDirection
NegotiateRtpTransceiverDirection(RtpTransceiverDirection offer,
RtpTransceiverDirection wants) {
return RtpTransceiverDirection(offer.recv && wants.send,
offer.send && wants.recv);
}
static bool IsMediaContentOfType(const ContentInfo* content,
MediaType media_type) {
if (!IsMediaContent(content)) {
return false;
}
const MediaContentDescription* mdesc =
static_cast<const MediaContentDescription*>(content->description);
return mdesc && mdesc->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 MediaContentDescription* media) {
if (!media) {
return NULL;
}
return &media->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 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);
}
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 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)) {
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) {
if (!sdesc)
return;
const ContentInfos& contents = sdesc->contents();
for (ContentInfos::const_iterator content = contents.begin();
content != contents.end(); ++content) {
if (!IsMediaContent(&*content)) {
continue;
}
const MediaContentDescription* media =
static_cast<const MediaContentDescription*>(
content->description);
const StreamParamsVec& streams = media->streams();
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
stream_params->push_back(*it);
}
}
}
// 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();
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 Stream in Streams with media type
// media_type to content_description.
// |current_params| - All currently known StreamParams of any media type.
template <class C>
static bool AddStreamParams(MediaType media_type,
const MediaSessionOptions& options,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* content_description,
const bool add_legacy_stream) {
// 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 MediaSessionOptions::Streams& streams = options.streams;
if (streams.empty() && add_legacy_stream) {
// TODO(perkj): Remove this legacy stream when all apps use StreamParams.
std::vector<uint32_t> ssrcs;
int num_ssrcs = include_rtx_streams ? 2 : 1;
GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
if (include_rtx_streams) {
content_description->AddLegacyStream(ssrcs[0], ssrcs[1]);
content_description->set_multistream(true);
} else {
content_description->AddLegacyStream(ssrcs[0]);
}
return true;
}
const bool include_flexfec_stream =
ContainsFlexfecCodec(content_description->codecs());
MediaSessionOptions::Streams::const_iterator stream_it;
for (stream_it = streams.begin();
stream_it != streams.end(); ++stream_it) {
if (stream_it->type != media_type)
continue; // Wrong media type.
StreamParams* param = GetStreamByIds(*current_streams, "", stream_it->id);
// groupid is empty for StreamParams generated using
// MediaSessionDescriptionFactory.
if (!param) {
// This is a new stream.
std::vector<uint32_t> ssrcs;
GenerateSsrcs(*current_streams, stream_it->num_sim_layers, &ssrcs);
StreamParams stream_param;
stream_param.id = stream_it->id;
// Add the generated ssrc.
for (size_t i = 0; i < ssrcs.size(); ++i) {
stream_param.ssrcs.push_back(ssrcs[i]);
}
if (stream_it->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]);
}
content_description->set_multistream(true);
}
// 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]);
content_description->set_multistream(true);
} else if (!ssrcs.empty()) {
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 = options.rtcp_cname;
stream_param.sync_label = stream_it->sync_label;
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->sync_label = stream_it->sync_label;
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 (!IsMediaContent(content) || !content->description) {
return false;
}
const MediaContentDescription* media_desc =
static_cast<const MediaContentDescription*>(content->description);
*cryptos = media_desc->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());
}
static 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 (IsMediaContent(content)) {
MediaContentDescription* media_desc =
static_cast<MediaContentDescription*>(content->description);
if (!media_desc) {
return false;
}
is_rtp = IsRtpProtocol(media_desc->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 =
static_cast<MediaContentDescription*>(content->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;
}
static TransportOptions GetTransportOptions(const MediaSessionOptions& options,
const std::string& content_name) {
TransportOptions transport_options;
auto it = options.transport_options.find(content_name);
if (it != options.transport_options.end()) {
transport_options = it->second;
}
transport_options.enable_ice_renomination = options.enable_ice_renomination;
return transport_options;
}
// Create a media content to be offered in a session-initiate,
// according to the given options.rtcp_mux, options.is_muc,
// options.streams, codecs, secure_transport, crypto, and 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). If
// add_legacy_stream is true, and current_streams is empty, a legacy
// stream is created. The created content is added to the offer.
template <class C>
static bool CreateMediaContentOffer(
const MediaSessionOptions& 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,
bool add_legacy_stream,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* offer) {
offer->AddCodecs(codecs);
offer->set_rtcp_mux(options.rtcp_mux_enabled);
if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
offer->set_rtcp_reduced_size(true);
}
offer->set_multistream(options.is_muc);
offer->set_rtp_header_extensions(rtp_extensions);
if (!AddStreamParams(offer->type(), options, current_streams, offer,
add_legacy_stream)) {
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) {
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)) {
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;
}
// Adds all codecs from |reference_codecs| to |offered_codecs| that dont'
// already exist in |offered_codecs| and ensure the payload types don't
// collide.
template <class C>
static void FindCodecsToOffer(
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;
std::string associated_pt_str;
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_pt_str)) {
LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
<< " is missing an associated payload type.";
continue;
}
int associated_pt;
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
<< " of RTX codec " << rtx_codec.name
<< " to an integer.";
continue;
}
// Find the associated reference codec for the reference RTX codec.
const C* associated_codec =
FindCodecById(reference_codecs, associated_pt);
if (!associated_codec) {
LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
<< associated_pt << " for RTX codec " << rtx_codec.name
<< ".";
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)) {
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 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;
}
// Iterates through |offered_extensions|, adding each one to
// |regular_extensions| (or |encrypted_extensions| if encrypted) and |used_ids|,
// and resolving ID conflicts.
// If an offered extension has the same URI as one in |regular_extensions| or
// |encrypted_extensions|, it will re-use the same ID and won't be treated as
// a conflict.
static void FindAndSetRtpHdrExtUsed(RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* regular_extensions,
RtpHeaderExtensions* encrypted_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
for (auto& extension : *offered_extensions) {
webrtc::RtpExtension existing;
if ((extension.encrypt &&
FindByUri(*encrypted_extensions, extension, &existing)) ||
(!extension.encrypt &&
FindByUri(*regular_extensions, extension, &existing))) {
extension.id = existing.id;
} else {
used_ids->FindAndSetIdUsed(&extension);
if (extension.encrypt) {
encrypted_extensions->push_back(extension);
} else {
regular_extensions->push_back(extension);
}
}
}
}
// Adds |reference_extensions| to |offered_extensions|, while updating
// |all_extensions| and |used_ids|.
static void FindRtpHdrExtsToOffer(
const RtpHeaderExtensions& reference_extensions,
RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* all_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
for (auto reference_extension : reference_extensions) {
if (!FindByUri(*offered_extensions, reference_extension, NULL)) {
webrtc::RtpExtension existing;
if (FindByUri(*all_extensions, reference_extension, &existing)) {
offered_extensions->push_back(existing);
} else {
used_ids->FindAndSetIdUsed(&reference_extension);
all_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 in a session-accept,
// according to the given options.rtcp_mux, options.streams, codecs,
// crypto, and 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). If add_legacy_stream is
// true, and current_streams is empty, a legacy stream is created.
// The codecs, rtcp_mux, and crypto are all negotiated with the offer
// from the incoming session-initiate. 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 MediaSessionOptions& 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 add_legacy_stream,
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(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, 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(answer->type(), options, current_streams, answer,
add_legacy_stream)) {
return false; // Something went seriously wrong.
}
// Make sure the answer media content direction is per default set as
// described in RFC3264 section 6.1.
const bool is_data = !IsRtpProtocol(answer->protocol());
const bool has_send_streams = !answer->streams().empty();
const bool wants_send = has_send_streams || is_data;
const bool recv_audio =
answer->type() == cricket::MEDIA_TYPE_AUDIO && options.recv_audio;
const bool recv_video =
answer->type() == cricket::MEDIA_TYPE_VIDEO && options.recv_video;
const bool recv_data =
answer->type() == cricket::MEDIA_TYPE_DATA;
const bool wants_receive = recv_audio || recv_video || recv_data;
auto offer_rtd =
RtpTransceiverDirection::FromMediaContentDirection(offer->direction());
auto wants_rtd = RtpTransceiverDirection(wants_send, wants_receive);
answer->set_direction(NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd)
.ToMediaContentDirection());
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 std::string& content_name,
const SessionDescription* current_description) {
if (!current_description)
return false;
const ContentInfo* content =
current_description->GetContentByName(content_name);
if (!content)
return false;
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_description);
if (!current_tdesc)
return false;
return current_tdesc->secure();
}
std::string MediaContentDirectionToString(MediaContentDirection direction) {
std::string dir_str;
switch (direction) {
case MD_INACTIVE:
dir_str = "inactive";
break;
case MD_SENDONLY:
dir_str = "sendonly";
break;
case MD_RECVONLY:
dir_str = "recvonly";
break;
case MD_SENDRECV:
dir_str = "sendrecv";
break;
default:
RTC_NOTREACHED();
break;
}
return dir_str;
}
void MediaSessionOptions::AddSendStream(MediaType type,
const std::string& id,
const std::string& sync_label) {
AddSendStreamInternal(type, id, sync_label, 1);
}
void MediaSessionOptions::AddSendVideoStream(
const std::string& id,
const std::string& sync_label,
int num_sim_layers) {
AddSendStreamInternal(MEDIA_TYPE_VIDEO, id, sync_label, num_sim_layers);
}
void MediaSessionOptions::AddSendStreamInternal(
MediaType type,
const std::string& id,
const std::string& sync_label,
int num_sim_layers) {
streams.push_back(Stream(type, id, sync_label, num_sim_layers));
// If we haven't already set the data_channel_type, and we add a
// stream, we assume it's an RTP data stream.
if (type == MEDIA_TYPE_DATA && data_channel_type == DCT_NONE)
data_channel_type = DCT_RTP;
}
void MediaSessionOptions::RemoveSendStream(MediaType type,
const std::string& id) {
Streams::iterator stream_it = streams.begin();
for (; stream_it != streams.end(); ++stream_it) {
if (stream_it->type == type && stream_it->id == id) {
streams.erase(stream_it);
return;
}
}
RTC_NOTREACHED();
}
bool MediaSessionOptions::HasSendMediaStream(MediaType type) const {
Streams::const_iterator stream_it = streams.begin();
for (; stream_it != streams.end(); ++stream_it) {
if (stream_it->type == type) {
return true;
}
}
return false;
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
const TransportDescriptionFactory* transport_desc_factory)
: secure_(SEC_DISABLED),
add_legacy_(true),
transport_desc_factory_(transport_desc_factory) {
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
ChannelManager* channel_manager,
const TransportDescriptionFactory* transport_desc_factory)
: secure_(SEC_DISABLED),
add_legacy_(true),
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_);
NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
&audio_sendrecv_codecs_);
}
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;
audio_sendrecv_codecs_.clear();
// 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(recv_codecs, send_codecs, &audio_sendrecv_codecs_);
}
SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description) const {
std::unique_ptr<SessionDescription> offer(new SessionDescription());
StreamParamsVec current_streams;
GetCurrentStreamParams(current_description, &current_streams);
const bool wants_send =
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
const AudioCodecs& supported_audio_codecs =
GetAudioCodecsForOffer({wants_send, options.recv_audio});
AudioCodecs audio_codecs;
VideoCodecs video_codecs;
DataCodecs data_codecs;
GetCodecsToOffer(current_description, supported_audio_codecs,
video_codecs_, data_codecs_,
&audio_codecs, &video_codecs, &data_codecs);
if (!options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&audio_codecs);
}
RtpHeaderExtensions audio_rtp_extensions;
RtpHeaderExtensions video_rtp_extensions;
GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions,
&video_rtp_extensions);
bool audio_added = false;
bool video_added = false;
bool data_added = false;
// Iterate through the contents of |current_description| to maintain the order
// of the m-lines in the new offer.
if (current_description) {
ContentInfos::const_iterator it = current_description->contents().begin();
for (; it != current_description->contents().end(); ++it) {
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
if (!AddAudioContentForOffer(options, current_description,
audio_rtp_extensions, audio_codecs,
&current_streams, offer.get())) {
return NULL;
}
audio_added = true;
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
if (!AddVideoContentForOffer(options, current_description,
video_rtp_extensions, video_codecs,
&current_streams, offer.get())) {
return NULL;
}
video_added = true;
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) {
MediaSessionOptions options_copy(options);
if (IsSctp(static_cast<const MediaContentDescription*>(it->description)
->protocol())) {
options_copy.data_channel_type = DCT_SCTP;
}
if (!AddDataContentForOffer(options_copy, current_description,
&data_codecs, &current_streams,
offer.get())) {
return NULL;
}
data_added = true;
} else {
RTC_NOTREACHED();
}
}
}
// Append contents that are not in |current_description|.
if (!audio_added && options.has_audio() &&
!AddAudioContentForOffer(options, current_description,
audio_rtp_extensions, audio_codecs,
&current_streams, offer.get())) {
return NULL;
}
if (!video_added && options.has_video() &&
!AddVideoContentForOffer(options, current_description,
video_rtp_extensions, video_codecs,
&current_streams, offer.get())) {
return NULL;
}
if (!data_added && options.has_data() &&
!AddDataContentForOffer(options, current_description, &data_codecs,
&current_streams, offer.get())) {
return NULL;
}
// Bundle the contents together, if we've been asked to do so, and update any
// parameters that need to be tweaked for BUNDLE.
if (options.bundle_enabled) {
ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
for (ContentInfos::const_iterator content = offer->contents().begin();
content != offer->contents().end(); ++content) {
offer_bundle.AddContentName(content->name);
}
offer->AddGroup(offer_bundle);
if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
LOG(LS_ERROR) << "CreateOffer failed to UpdateTransportInfoForBundle.";
return NULL;
}
if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
LOG(LS_ERROR) << "CreateOffer failed to UpdateCryptoParamsForBundle.";
return NULL;
}
}
return offer.release();
}
SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
const SessionDescription* offer, const MediaSessionOptions& 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, &current_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;
ContentInfos::const_iterator it = offer->contents().begin();
for (; it != offer->contents().end(); ++it) {
if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
if (!AddAudioContentForAnswer(offer, options, current_description,
bundle_transport.get(), &current_streams,
answer.get())) {
return NULL;
}
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
if (!AddVideoContentForAnswer(offer, options, current_description,
bundle_transport.get(), &current_streams,
answer.get())) {
return NULL;
}
} else {
RTC_DCHECK(IsMediaContentOfType(&*it, MEDIA_TYPE_DATA));
if (!AddDataContentForAnswer(offer, options, current_description,
bundle_transport.get(), &current_streams,
answer.get())) {
return NULL;
}
}
// 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 && 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())) {
LOG(LS_ERROR) << "CreateAnswer failed to UpdateTransportInfoForBundle.";
return NULL;
}
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
LOG(LS_ERROR) << "CreateAnswer failed to UpdateCryptoParamsForBundle.";
return NULL;
}
}
return answer.release();
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
const RtpTransceiverDirection& direction) const {
// If stream is inactive - generate list as if sendrecv.
if (direction.send == direction.recv) {
return audio_sendrecv_codecs_;
} else if (direction.send) {
return audio_send_codecs_;
} else {
return audio_recv_codecs_;
}
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
const RtpTransceiverDirection& offer,
const RtpTransceiverDirection& answer) const {
// For inactive and sendrecv answers, generate lists as if we were to accept
// the offer's direction. See RFC 3264 Section 6.1.
if (answer.send == answer.recv) {
if (offer.send == offer.recv) {
return audio_sendrecv_codecs_;
} else if (offer.send) {
return audio_recv_codecs_;
} else {
return audio_send_codecs_;
}
} else if (answer.send) {
return audio_send_codecs_;
} else {
return audio_recv_codecs_;
}
}
void MediaSessionDescriptionFactory::GetCodecsToOffer(
const SessionDescription* current_description,
const AudioCodecs& supported_audio_codecs,
const VideoCodecs& supported_video_codecs,
const DataCodecs& supported_data_codecs,
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 payloadtype is not reused if a new media
// type is added.
if (current_description) {
const AudioContentDescription* audio =
GetFirstAudioContentDescription(current_description);
if (audio) {
*audio_codecs = audio->codecs();
used_pltypes.FindAndSetIdUsed<AudioCodec>(audio_codecs);
}
const VideoContentDescription* video =
GetFirstVideoContentDescription(current_description);
if (video) {
*video_codecs = video->codecs();
used_pltypes.FindAndSetIdUsed<VideoCodec>(video_codecs);
}
const DataContentDescription* data =
GetFirstDataContentDescription(current_description);
if (data) {
*data_codecs = data->codecs();
used_pltypes.FindAndSetIdUsed<DataCodec>(data_codecs);
}
}
// Add our codecs that are not in |current_description|.
FindCodecsToOffer<AudioCodec>(supported_audio_codecs, audio_codecs,
&used_pltypes);
FindCodecsToOffer<VideoCodec>(supported_video_codecs, video_codecs,
&used_pltypes);
FindCodecsToOffer<DataCodec>(supported_data_codecs, data_codecs,
&used_pltypes);
}
void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
const SessionDescription* current_description,
RtpHeaderExtensions* audio_extensions,
RtpHeaderExtensions* 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;
audio_extensions->clear();
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) {
const AudioContentDescription* audio =
GetFirstAudioContentDescription(current_description);
if (audio) {
*audio_extensions = audio->rtp_header_extensions();
FindAndSetRtpHdrExtUsed(audio_extensions, &all_regular_extensions,
&all_encrypted_extensions, &used_ids);
}
const VideoContentDescription* video =
GetFirstVideoContentDescription(current_description);
if (video) {
*video_extensions = video->rtp_header_extensions();
FindAndSetRtpHdrExtUsed(video_extensions, &all_regular_extensions,
&all_encrypted_extensions, &used_ids);
}
}
// Add our default RTP header extensions that are not in
// |current_description|.
FindRtpHdrExtsToOffer(audio_rtp_header_extensions(), audio_extensions,
&all_regular_extensions, &used_ids);
FindRtpHdrExtsToOffer(video_rtp_header_extensions(), video_extensions,
&all_regular_extensions, &used_ids);
// TODO(jbauch): Support adding encrypted header extensions to existing
// sessions.
if (enable_encrypted_rtp_header_extensions_ && !current_description) {
AddEncryptedVersionsOfHdrExts(audio_extensions, &all_encrypted_extensions,
&used_ids);
AddEncryptedVersionsOfHdrExts(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) {
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))) {
LOG(LS_ERROR)
<< "Failed to AddTransportAnswer, content name=" << content_name;
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
const RtpHeaderExtensions& audio_rtp_extensions,
const AudioCodecs& audio_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
const ContentInfo* current_audio_content =
GetFirstAudioContent(current_description);
std::string content_name =
current_audio_content ? current_audio_content->name : CN_AUDIO;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedAudioSdesCryptoSuiteNames(options.crypto_options, &crypto_suites);
if (!CreateMediaContentOffer(
options,
audio_codecs,
sdes_policy,
GetCryptos(GetFirstAudioContentDescription(current_description)),
crypto_suites,
audio_rtp_extensions,
add_legacy_,
current_streams,
audio.get())) {
return false;
}
audio->set_lang(lang_);
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, audio.get());
auto offer_rtd =
RtpTransceiverDirection(!audio->streams().empty(), options.recv_audio);
audio->set_direction(offer_rtd.ToMediaContentDirection());
desc->AddContent(content_name, NS_JINGLE_RTP, audio.release());
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
const RtpHeaderExtensions& video_rtp_extensions,
const VideoCodecs& video_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
const ContentInfo* current_video_content =
GetFirstVideoContent(current_description);
std::string content_name =
current_video_content ? current_video_content->name : CN_VIDEO;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<VideoContentDescription> video(new VideoContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedVideoSdesCryptoSuiteNames(options.crypto_options, &crypto_suites);
if (!CreateMediaContentOffer(
options,
video_codecs,
sdes_policy,
GetCryptos(GetFirstVideoContentDescription(current_description)),
crypto_suites,
video_rtp_extensions,
add_legacy_,
current_streams,
video.get())) {
return false;
}
video->set_bandwidth(options.video_bandwidth);
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, video.get());
if (!video->streams().empty()) {
if (options.recv_video) {
video->set_direction(MD_SENDRECV);
} else {
video->set_direction(MD_SENDONLY);
}
} else {
if (options.recv_video) {
video->set_direction(MD_RECVONLY);
} else {
video->set_direction(MD_INACTIVE);
}
}
desc->AddContent(content_name, NS_JINGLE_RTP, video.release());
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddDataContentForOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description,
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 = (options.data_channel_type == DCT_SCTP);
FilterDataCodecs(data_codecs, is_sctp);
const ContentInfo* current_data_content =
GetFirstDataContent(current_description);
std::string content_name =
current_data_content ? current_data_content->name : CN_DATA;
cricket::SecurePolicy sdes_policy =
IsDtlsActive(content_name, 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(options.crypto_options,
&crypto_suites);
}
if (!CreateMediaContentOffer(
options,
*data_codecs,
sdes_policy,
GetCryptos(GetFirstDataContentDescription(current_description)),
crypto_suites,
RtpHeaderExtensions(),
add_legacy_,
current_streams,
data.get())) {
return false;
}
if (is_sctp) {
desc->AddContent(content_name, NS_JINGLE_DRAFT_SCTP, data.release());
} else {
data->set_bandwidth(options.data_bandwidth);
SetMediaProtocol(secure_transport, data.get());
desc->AddContent(content_name, NS_JINGLE_RTP, data.release());
}
if (!AddTransportOffer(content_name,
GetTransportOptions(options, content_name),
current_description, desc)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* audio_content = GetFirstAudioContent(offer);
const AudioContentDescription* offer_audio =
static_cast<const AudioContentDescription*>(audio_content->description);
std::unique_ptr<TransportDescription> audio_transport(
CreateTransportAnswer(audio_content->name, offer,
GetTransportOptions(options, audio_content->name),
current_description, bundle_transport != nullptr));
if (!audio_transport) {
return false;
}
// Pick codecs based on the requested communications direction in the offer.
const bool wants_send =
options.HasSendMediaStream(MEDIA_TYPE_AUDIO) || add_legacy_;
auto wants_rtd = RtpTransceiverDirection(wants_send, options.recv_audio);
auto offer_rtd =
RtpTransceiverDirection::FromMediaContentDirection(
offer_audio->direction());
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
AudioCodecs audio_codecs = GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
if (!options.vad_enabled) {
StripCNCodecs(&audio_codecs);
}
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && 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,
options,
audio_codecs,
sdes_policy,
GetCryptos(GetFirstAudioContentDescription(current_description)),
audio_rtp_extensions_,
enable_encrypted_rtp_header_extensions_,
current_streams,
add_legacy_,
bundle_enabled,
audio_answer.get())) {
return false; // Fails the session setup.
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: audio_transport->secure();
bool rejected = !options.has_audio() || audio_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
audio_answer->protocol(), secure);
if (!rejected) {
AddTransportAnswer(audio_content->name, *(audio_transport.get()), answer);
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Audio is not supported in the answer.";
}
answer->AddContent(audio_content->name, audio_content->type, rejected,
audio_answer.release());
return true;
}
bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* video_content = GetFirstVideoContent(offer);
std::unique_ptr<TransportDescription> video_transport(
CreateTransportAnswer(video_content->name, offer,
GetTransportOptions(options, video_content->name),
current_description, bundle_transport != nullptr));
if (!video_transport) {
return false;
}
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();
bool bundle_enabled =
offer->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
if (!CreateMediaContentAnswer(
static_cast<const VideoContentDescription*>(
video_content->description),
options,
video_codecs_,
sdes_policy,
GetCryptos(GetFirstVideoContentDescription(current_description)),
video_rtp_extensions_,
enable_encrypted_rtp_header_extensions_,
current_streams,
add_legacy_,
bundle_enabled,
video_answer.get())) {
return false;
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: video_transport->secure();
bool rejected = !options.has_video() || video_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
video_answer->protocol(), secure);
if (!rejected) {
if (!AddTransportAnswer(video_content->name, *(video_transport.get()),
answer)) {
return false;
}
video_answer->set_bandwidth(options.video_bandwidth);
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Video is not supported in the answer.";
}
answer->AddContent(video_content->name, video_content->type, rejected,
video_answer.release());
return true;
}
bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* data_content = GetFirstDataContent(offer);
std::unique_ptr<TransportDescription> data_transport(
CreateTransportAnswer(data_content->name, offer,
GetTransportOptions(options, data_content->name),
current_description, bundle_transport != nullptr));
if (!data_transport) {
return false;
}
bool is_sctp = (options.data_channel_type == DCT_SCTP);
std::vector<DataCodec> data_codecs(data_codecs_);
FilterDataCodecs(&data_codecs, is_sctp);
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->HasGroup(GROUP_TYPE_BUNDLE) && options.bundle_enabled;
if (!CreateMediaContentAnswer(
static_cast<const DataContentDescription*>(
data_content->description),
options,
data_codecs_,
sdes_policy,
GetCryptos(GetFirstDataContentDescription(current_description)),
RtpHeaderExtensions(),
enable_encrypted_rtp_header_extensions_,
current_streams,
add_legacy_,
bundle_enabled,
data_answer.get())) {
return false; // Fails the session setup.
}
// Respond with sctpmap if the offer uses sctpmap.
const DataContentDescription* offer_data_description =
static_cast<const DataContentDescription*>(data_content->description);
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 = !options.has_data() || data_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_DATA,
data_answer->protocol(), secure);
if (!rejected) {
data_answer->set_bandwidth(options.data_bandwidth);
if (!AddTransportAnswer(data_content->name, *(data_transport.get()),
answer)) {
return false;
}
} else {
// RFC 3264
// The answer MUST contain the same number of m-lines as the offer.
LOG(LS_INFO) << "Data is not supported in the answer.";
}
answer->AddContent(data_content->name, data_content->type, rejected,
data_answer.release());
return true;
}
bool IsMediaContent(const ContentInfo* content) {
return (content &&
(content->type == NS_JINGLE_RTP ||
content->type == NS_JINGLE_DRAFT_SCTP));
}
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);
}
static 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);
const ContentDescription* description = content ? content->description : NULL;
return static_cast<const MediaContentDescription*>(description);
}
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);
}
static 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);
ContentDescription* description = content ? content->description : NULL;
return static_cast<MediaContentDescription*>(description);
}
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