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/*
* 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.
*/
// Types and classes used in media session descriptions.
#ifndef WEBRTC_PC_MEDIASESSION_H_
#define WEBRTC_PC_MEDIASESSION_H_
#include <algorithm>
#include <map>
#include <string>
#include <vector>
#include "webrtc/api/mediatypes.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/mediaengine.h" // For DataChannelType
#include "webrtc/media/base/streamparams.h"
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/p2p/base/jseptransport.h"
#include "webrtc/p2p/base/transportdescriptionfactory.h"
namespace cricket {
class ChannelManager;
typedef std::vector<AudioCodec> AudioCodecs;
typedef std::vector<VideoCodec> VideoCodecs;
typedef std::vector<DataCodec> DataCodecs;
typedef std::vector<CryptoParams> CryptoParamsVec;
typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
enum MediaContentDirection {
MD_INACTIVE,
MD_SENDONLY,
MD_RECVONLY,
MD_SENDRECV
};
std::string MediaContentDirectionToString(MediaContentDirection direction);
enum CryptoType {
CT_NONE,
CT_SDES,
CT_DTLS
};
// RTC4585 RTP/AVPF
extern const char kMediaProtocolAvpf[];
// RFC5124 RTP/SAVPF
extern const char kMediaProtocolSavpf[];
extern const char kMediaProtocolDtlsSavpf[];
extern const char kMediaProtocolRtpPrefix[];
extern const char kMediaProtocolSctp[];
extern const char kMediaProtocolDtlsSctp[];
extern const char kMediaProtocolUdpDtlsSctp[];
extern const char kMediaProtocolTcpDtlsSctp[];
// Options to control how session descriptions are generated.
const int kAutoBandwidth = -1;
const int kBufferedModeDisabled = 0;
// Default RTCP CNAME for unit tests.
const char kDefaultRtcpCname[] = "DefaultRtcpCname";
struct RtpTransceiverDirection {
bool send;
bool recv;
RtpTransceiverDirection(bool send, bool recv) : send(send), recv(recv) {}
bool operator==(const RtpTransceiverDirection& o) const {
return send == o.send && recv == o.recv;
}
bool operator!=(const RtpTransceiverDirection& o) const {
return !(*this == o);
}
static RtpTransceiverDirection FromMediaContentDirection(
MediaContentDirection md);
MediaContentDirection ToMediaContentDirection() const;
RtpTransceiverDirection Reversed() const {
return RtpTransceiverDirection(recv, send);
}
};
RtpTransceiverDirection
NegotiateRtpTransceiverDirection(RtpTransceiverDirection offer,
RtpTransceiverDirection wants);
// Options for an RtpSender contained with an media description/"m=" section.
struct SenderOptions {
std::string track_id;
// TODO(steveanton): As part of work towards Unified Plan, this has been
// changed to be a vector. But for now this can only have exactly one.
std::vector<std::string> stream_ids;
int num_sim_layers;
};
// Options for an individual media description/"m=" section.
struct MediaDescriptionOptions {
MediaDescriptionOptions(MediaType type,
const std::string& mid,
RtpTransceiverDirection direction,
bool stopped)
: type(type), mid(mid), direction(direction), stopped(stopped) {}
// TODO(deadbeef): When we don't support Plan B, there will only be one
// sender per media description and this can be simplified.
void AddAudioSender(const std::string& track_id,
const std::vector<std::string>& stream_ids);
void AddVideoSender(const std::string& track_id,
const std::vector<std::string>& stream_ids,
int num_sim_layers);
// Internally just uses sender_options.
void AddRtpDataChannel(const std::string& track_id,
const std::string& stream_id);
MediaType type;
std::string mid;
RtpTransceiverDirection direction;
bool stopped;
TransportOptions transport_options;
// Note: There's no equivalent "RtpReceiverOptions" because only send
// stream information goes in the local descriptions.
std::vector<SenderOptions> sender_options;
private:
// Doesn't DCHECK on |type|.
void AddSenderInternal(const std::string& track_id,
const std::vector<std::string>& stream_ids,
int num_sim_layers);
};
// Provides a mechanism for describing how m= sections should be generated.
// The m= section with index X will use media_description_options[X]. There
// must be an option for each existing section if creating an answer, or a
// subsequent offer.
struct MediaSessionOptions {
MediaSessionOptions() {}
bool has_audio() const { return HasMediaDescription(MEDIA_TYPE_AUDIO); }
bool has_video() const { return HasMediaDescription(MEDIA_TYPE_VIDEO); }
bool has_data() const { return HasMediaDescription(MEDIA_TYPE_DATA); }
bool HasMediaDescription(MediaType type) const;
DataChannelType data_channel_type = DCT_NONE;
bool is_muc = false;
bool vad_enabled = true; // When disabled, removes all CN codecs from SDP.
bool rtcp_mux_enabled = true;
bool bundle_enabled = false;
std::string rtcp_cname = kDefaultRtcpCname;
rtc::CryptoOptions crypto_options;
// List of media description options in the same order that the media
// descriptions will be generated.
std::vector<MediaDescriptionOptions> media_description_options;
};
// "content" (as used in XEP-0166) descriptions for voice and video.
class MediaContentDescription : public ContentDescription {
public:
MediaContentDescription() {}
virtual MediaType type() const = 0;
virtual bool has_codecs() const = 0;
// |protocol| is the expected media transport protocol, such as RTP/AVPF,
// RTP/SAVPF or SCTP/DTLS.
std::string protocol() const { return protocol_; }
void set_protocol(const std::string& protocol) { protocol_ = protocol; }
MediaContentDirection direction() const { return direction_; }
void set_direction(MediaContentDirection direction) {
direction_ = direction;
}
bool rtcp_mux() const { return rtcp_mux_; }
void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
void set_rtcp_reduced_size(bool reduced_size) {
rtcp_reduced_size_ = reduced_size;
}
int bandwidth() const { return bandwidth_; }
void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
void AddCrypto(const CryptoParams& params) {
cryptos_.push_back(params);
}
void set_cryptos(const std::vector<CryptoParams>& cryptos) {
cryptos_ = cryptos;
}
CryptoType crypto_required() const { return crypto_required_; }
void set_crypto_required(CryptoType type) {
crypto_required_ = type;
}
const RtpHeaderExtensions& rtp_header_extensions() const {
return rtp_header_extensions_;
}
void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
rtp_header_extensions_ = extensions;
rtp_header_extensions_set_ = true;
}
void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
rtp_header_extensions_.push_back(ext);
rtp_header_extensions_set_ = true;
}
void AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) {
webrtc::RtpExtension webrtc_extension;
webrtc_extension.uri = ext.uri;
webrtc_extension.id = ext.id;
rtp_header_extensions_.push_back(webrtc_extension);
rtp_header_extensions_set_ = true;
}
void ClearRtpHeaderExtensions() {
rtp_header_extensions_.clear();
rtp_header_extensions_set_ = true;
}
// We can't always tell if an empty list of header extensions is
// because the other side doesn't support them, or just isn't hooked up to
// signal them. For now we assume an empty list means no signaling, but
// provide the ClearRtpHeaderExtensions method to allow "no support" to be
// clearly indicated (i.e. when derived from other information).
bool rtp_header_extensions_set() const {
return rtp_header_extensions_set_;
}
// True iff the client supports multiple streams.
void set_multistream(bool multistream) { multistream_ = multistream; }
bool multistream() const { return multistream_; }
const StreamParamsVec& streams() const {
return streams_;
}
// TODO(pthatcher): Remove this by giving mediamessage.cc access
// to MediaContentDescription
StreamParamsVec& mutable_streams() {
return streams_;
}
void AddStream(const StreamParams& stream) {
streams_.push_back(stream);
}
// Legacy streams have an ssrc, but nothing else.
void AddLegacyStream(uint32_t ssrc) {
streams_.push_back(StreamParams::CreateLegacy(ssrc));
}
void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
StreamParams sp = StreamParams::CreateLegacy(ssrc);
sp.AddFidSsrc(ssrc, fid_ssrc);
streams_.push_back(sp);
}
// Sets the CNAME of all StreamParams if it have not been set.
void SetCnameIfEmpty(const std::string& cname) {
for (cricket::StreamParamsVec::iterator it = streams_.begin();
it != streams_.end(); ++it) {
if (it->cname.empty())
it->cname = cname;
}
}
uint32_t first_ssrc() const {
if (streams_.empty()) {
return 0;
}
return streams_[0].first_ssrc();
}
bool has_ssrcs() const {
if (streams_.empty()) {
return false;
}
return streams_[0].has_ssrcs();
}
void set_conference_mode(bool enable) { conference_mode_ = enable; }
bool conference_mode() const { return conference_mode_; }
void set_partial(bool partial) { partial_ = partial; }
bool partial() const { return partial_; }
void set_buffered_mode_latency(int latency) {
buffered_mode_latency_ = latency;
}
int buffered_mode_latency() const { return buffered_mode_latency_; }
// https://tools.ietf.org/html/rfc4566#section-5.7
// May be present at the media or session level of SDP. If present at both
// levels, the media-level attribute overwrites the session-level one.
void set_connection_address(const rtc::SocketAddress& address) {
connection_address_ = address;
}
const rtc::SocketAddress& connection_address() const {
return connection_address_;
}
protected:
bool rtcp_mux_ = false;
bool rtcp_reduced_size_ = false;
int bandwidth_ = kAutoBandwidth;
std::string protocol_;
std::vector<CryptoParams> cryptos_;
CryptoType crypto_required_ = CT_NONE;
std::vector<webrtc::RtpExtension> rtp_header_extensions_;
bool rtp_header_extensions_set_ = false;
bool multistream_ = false;
StreamParamsVec streams_;
bool conference_mode_ = false;
bool partial_ = false;
int buffered_mode_latency_ = kBufferedModeDisabled;
MediaContentDirection direction_ = MD_SENDRECV;
rtc::SocketAddress connection_address_;
};
template <class C>
class MediaContentDescriptionImpl : public MediaContentDescription {
public:
typedef C CodecType;
// Codecs should be in preference order (most preferred codec first).
const std::vector<C>& codecs() const { return codecs_; }
void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
virtual bool has_codecs() const { return !codecs_.empty(); }
bool HasCodec(int id) {
bool found = false;
for (typename std::vector<C>::iterator iter = codecs_.begin();
iter != codecs_.end(); ++iter) {
if (iter->id == id) {
found = true;
break;
}
}
return found;
}
void AddCodec(const C& codec) {
codecs_.push_back(codec);
}
void AddOrReplaceCodec(const C& codec) {
for (typename std::vector<C>::iterator iter = codecs_.begin();
iter != codecs_.end(); ++iter) {
if (iter->id == codec.id) {
*iter = codec;
return;
}
}
AddCodec(codec);
}
void AddCodecs(const std::vector<C>& codecs) {
typename std::vector<C>::const_iterator codec;
for (codec = codecs.begin(); codec != codecs.end(); ++codec) {
AddCodec(*codec);
}
}
private:
std::vector<C> codecs_;
};
class AudioContentDescription : public MediaContentDescriptionImpl<AudioCodec> {
public:
AudioContentDescription() :
agc_minus_10db_(false) {}
virtual ContentDescription* Copy() const {
return new AudioContentDescription(*this);
}
virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
const std::string &lang() const { return lang_; }
void set_lang(const std::string &lang) { lang_ = lang; }
bool agc_minus_10db() const { return agc_minus_10db_; }
void set_agc_minus_10db(bool enable) {
agc_minus_10db_ = enable;
}
private:
bool agc_minus_10db_;
private:
std::string lang_;
};
class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
public:
virtual ContentDescription* Copy() const {
return new VideoContentDescription(*this);
}
virtual MediaType type() const { return MEDIA_TYPE_VIDEO; }
};
class DataContentDescription : public MediaContentDescriptionImpl<DataCodec> {
public:
DataContentDescription() {}
virtual ContentDescription* Copy() const {
return new DataContentDescription(*this);
}
virtual MediaType type() const { return MEDIA_TYPE_DATA; }
bool use_sctpmap() const { return use_sctpmap_; }
void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
private:
bool use_sctpmap_ = true;
};
// Creates media session descriptions according to the supplied codecs and
// other fields, as well as the supplied per-call options.
// When creating answers, performs the appropriate negotiation
// of the various fields to determine the proper result.
class MediaSessionDescriptionFactory {
public:
// Default ctor; use methods below to set configuration.
// The TransportDescriptionFactory is not owned by MediaSessionDescFactory,
// so it must be kept alive by the user of this class.
explicit MediaSessionDescriptionFactory(
const TransportDescriptionFactory* factory);
// This helper automatically sets up the factory to get its configuration
// from the specified ChannelManager.
MediaSessionDescriptionFactory(ChannelManager* cmanager,
const TransportDescriptionFactory* factory);
const AudioCodecs& audio_sendrecv_codecs() const;
const AudioCodecs& audio_send_codecs() const;
const AudioCodecs& audio_recv_codecs() const;
void set_audio_codecs(const AudioCodecs& send_codecs,
const AudioCodecs& recv_codecs);
void set_audio_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
audio_rtp_extensions_ = extensions;
}
const RtpHeaderExtensions& audio_rtp_header_extensions() const {
return audio_rtp_extensions_;
}
const VideoCodecs& video_codecs() const { return video_codecs_; }
void set_video_codecs(const VideoCodecs& codecs) { video_codecs_ = codecs; }
void set_video_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
video_rtp_extensions_ = extensions;
}
const RtpHeaderExtensions& video_rtp_header_extensions() const {
return video_rtp_extensions_;
}
const DataCodecs& data_codecs() const { return data_codecs_; }
void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; }
SecurePolicy secure() const { return secure_; }
void set_secure(SecurePolicy s) { secure_ = s; }
void set_enable_encrypted_rtp_header_extensions(bool enable) {
enable_encrypted_rtp_header_extensions_ = enable;
}
SessionDescription* CreateOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
SessionDescription* CreateAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
private:
const AudioCodecs& GetAudioCodecsForOffer(
const RtpTransceiverDirection& direction) const;
const AudioCodecs& GetAudioCodecsForAnswer(
const RtpTransceiverDirection& offer,
const RtpTransceiverDirection& answer) const;
void GetCodecsForOffer(const SessionDescription* current_description,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs,
DataCodecs* data_codecs) const;
void GetCodecsForAnswer(const SessionDescription* current_description,
const SessionDescription* remote_offer,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs,
DataCodecs* data_codecs) const;
void GetRtpHdrExtsToOffer(const SessionDescription* current_description,
RtpHeaderExtensions* audio_extensions,
RtpHeaderExtensions* video_extensions) const;
bool AddTransportOffer(
const std::string& content_name,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
SessionDescription* offer) const;
TransportDescription* CreateTransportAnswer(
const std::string& content_name,
const SessionDescription* offer_desc,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
bool require_transport_attributes) const;
bool AddTransportAnswer(
const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const;
// Helpers for adding media contents to the SessionDescription. Returns true
// it succeeds or the media content is not needed, or false if there is any
// error.
bool 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;
bool 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;
bool 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 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;
bool 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;
bool 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;
void ComputeAudioCodecsIntersectionAndUnion();
AudioCodecs audio_send_codecs_;
AudioCodecs audio_recv_codecs_;
// Intersection of send and recv.
AudioCodecs audio_sendrecv_codecs_;
// Union of send and recv.
AudioCodecs all_audio_codecs_;
RtpHeaderExtensions audio_rtp_extensions_;
VideoCodecs video_codecs_;
RtpHeaderExtensions video_rtp_extensions_;
DataCodecs data_codecs_;
bool enable_encrypted_rtp_header_extensions_ = false;
// TODO(zhihuang): Rename secure_ to sdec_policy_; rename the related getter
// and setter.
SecurePolicy secure_ = SEC_DISABLED;
std::string lang_;
const TransportDescriptionFactory* transport_desc_factory_;
};
// Convenience functions.
bool IsMediaContent(const ContentInfo* content);
bool IsAudioContent(const ContentInfo* content);
bool IsVideoContent(const ContentInfo* content);
bool IsDataContent(const ContentInfo* content);
const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
MediaType media_type);
const ContentInfo* GetFirstAudioContent(const ContentInfos& contents);
const ContentInfo* GetFirstVideoContent(const ContentInfos& contents);
const ContentInfo* GetFirstDataContent(const ContentInfos& contents);
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc);
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc);
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc);
const AudioContentDescription* GetFirstAudioContentDescription(
const SessionDescription* sdesc);
const VideoContentDescription* GetFirstVideoContentDescription(
const SessionDescription* sdesc);
const DataContentDescription* GetFirstDataContentDescription(
const SessionDescription* sdesc);
// Non-const versions of the above functions.
// Useful when modifying an existing description.
ContentInfo* GetFirstMediaContent(ContentInfos& contents, MediaType media_type);
ContentInfo* GetFirstAudioContent(ContentInfos& contents);
ContentInfo* GetFirstVideoContent(ContentInfos& contents);
ContentInfo* GetFirstDataContent(ContentInfos& contents);
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
AudioContentDescription* GetFirstAudioContentDescription(
SessionDescription* sdesc);
VideoContentDescription* GetFirstVideoContentDescription(
SessionDescription* sdesc);
DataContentDescription* GetFirstDataContentDescription(
SessionDescription* sdesc);
// Helper functions to return crypto suites used for SDES.
void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites);
void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites);
void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites);
void GetSupportedAudioSdesCryptoSuiteNames(
const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
void GetSupportedVideoSdesCryptoSuiteNames(
const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
void GetSupportedDataSdesCryptoSuiteNames(
const rtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names);
} // namespace cricket
#endif // WEBRTC_PC_MEDIASESSION_H_