webrtc/talk/session/media/mediasession.h
jiayl@webrtc.org 8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 22:07:59 +00:00

525 lines
18 KiB
C++

/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
// Types and classes used in media session descriptions.
#ifndef TALK_SESSION_MEDIA_MEDIASESSION_H_
#define TALK_SESSION_MEDIA_MEDIASESSION_H_
#include <string>
#include <vector>
#include <algorithm>
#include "talk/base/scoped_ptr.h"
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/cryptoparams.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/base/mediaengine.h" // For DataChannelType
#include "talk/media/base/streamparams.h"
#include "talk/p2p/base/sessiondescription.h"
#include "talk/p2p/base/transport.h"
#include "talk/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<RtpHeaderExtension> RtpHeaderExtensions;
enum MediaType {
MEDIA_TYPE_AUDIO,
MEDIA_TYPE_VIDEO,
MEDIA_TYPE_DATA
};
std::string MediaTypeToString(MediaType type);
enum MediaContentDirection {
MD_INACTIVE,
MD_SENDONLY,
MD_RECVONLY,
MD_SENDRECV
};
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[];
// Options to control how session descriptions are generated.
const int kAutoBandwidth = -1;
const int kBufferedModeDisabled = 0;
struct MediaSessionOptions {
MediaSessionOptions() :
has_audio(true), // Audio enabled by default.
has_video(false),
data_channel_type(DCT_NONE),
is_muc(false),
vad_enabled(true), // When disabled, removes all CN codecs from SDP.
rtcp_mux_enabled(true),
bundle_enabled(false),
video_bandwidth(kAutoBandwidth),
data_bandwidth(kDataMaxBandwidth) {
}
bool has_data() const { return data_channel_type != DCT_NONE; }
// Add a stream with MediaType type and id.
// All streams with the same sync_label will get the same CNAME.
// All ids must be unique.
void AddStream(MediaType type,
const std::string& id,
const std::string& sync_label);
void AddVideoStream(const std::string& id,
const std::string& sync_label,
int num_sim_layers);
void RemoveStream(MediaType type, const std::string& id);
// Helper function.
void AddStreamInternal(MediaType type,
const std::string& id,
const std::string& sync_label,
int num_sim_layers);
bool has_audio;
bool has_video;
DataChannelType data_channel_type;
bool is_muc;
bool vad_enabled;
bool rtcp_mux_enabled;
bool bundle_enabled;
// bps. -1 == auto.
int video_bandwidth;
int data_bandwidth;
TransportOptions transport_options;
struct Stream {
Stream(MediaType type,
const std::string& id,
const std::string& sync_label,
int num_sim_layers)
: type(type), id(id), sync_label(sync_label),
num_sim_layers(num_sim_layers) {
}
MediaType type;
std::string id;
std::string sync_label;
int num_sim_layers;
};
typedef std::vector<Stream> Streams;
Streams streams;
};
// "content" (as used in XEP-0166) descriptions for voice and video.
class MediaContentDescription : public ContentDescription {
public:
MediaContentDescription()
: rtcp_mux_(false),
bandwidth_(kAutoBandwidth),
crypto_required_(CT_NONE),
rtp_header_extensions_set_(false),
multistream_(false),
conference_mode_(false),
partial_(false),
buffered_mode_latency_(kBufferedModeDisabled),
direction_(MD_SENDRECV) {
}
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; }
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 RtpHeaderExtension& ext) {
rtp_header_extensions_.push_back(ext);
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 ssrc) {
streams_.push_back(StreamParams::CreateLegacy(ssrc));
}
void AddLegacyStream(uint32 ssrc, uint32 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.
// This can be used to set the CNAME of legacy streams.
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 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_; }
protected:
bool rtcp_mux_;
int bandwidth_;
std::string protocol_;
std::vector<CryptoParams> cryptos_;
CryptoType crypto_required_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool rtp_header_extensions_set_;
bool multistream_;
StreamParamsVec streams_;
bool conference_mode_;
bool partial_;
int buffered_mode_latency_;
MediaContentDirection direction_;
};
template <class C>
class MediaContentDescriptionImpl : public MediaContentDescription {
public:
struct PreferenceSort {
bool operator()(C a, C b) { return a.preference > b.preference; }
};
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);
}
}
void SortCodecs() {
std::sort(codecs_.begin(), codecs_.end(), PreferenceSort());
}
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:
virtual ContentDescription* Copy() const {
return new DataContentDescription(*this);
}
virtual MediaType type() const { return MEDIA_TYPE_DATA; }
};
// 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_codecs() const { return audio_codecs_; }
void set_audio_codecs(const AudioCodecs& codecs) { audio_codecs_ = 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; }
// Decides if a StreamParams shall be added to the audio and video media
// content in SessionDescription when CreateOffer and CreateAnswer is called
// even if |options| don't include a Stream. This is needed to support legacy
// applications. |add_legacy_| is true per default.
void set_add_legacy_streams(bool add_legacy) { add_legacy_ = add_legacy; }
SessionDescription* CreateOffer(
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
SessionDescription* CreateAnswer(
const SessionDescription* offer,
const MediaSessionOptions& options,
const SessionDescription* current_description) const;
private:
void GetCodecsToOffer(const SessionDescription* current_description,
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) const;
bool AddTransportAnswer(
const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const;
AudioCodecs audio_codecs_;
RtpHeaderExtensions audio_rtp_extensions_;
VideoCodecs video_codecs_;
RtpHeaderExtensions video_rtp_extensions_;
DataCodecs data_codecs_;
SecurePolicy secure_;
bool add_legacy_;
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* 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);
bool GetStreamBySsrc(
const SessionDescription* sdesc, MediaType media_type,
uint32 ssrc, StreamParams* stream_out);
bool GetStreamByIds(
const SessionDescription* sdesc, MediaType media_type,
const std::string& groupid, const std::string& id,
StreamParams* stream_out);
// Functions for translating media candidate names.
// For converting between media ICE component and G-ICE channel
// names. For example:
// "rtp" <=> 1
// "rtcp" <=> 2
// "video_rtp" <=> 1
// "video_rtcp" <=> 2
// Will not convert in the general case of arbitrary channel names,
// but is useful for cases where we have candidates for media
// channels.
// returns false if there is no mapping.
bool GetMediaChannelNameFromComponent(
int component, cricket::MediaType media_type, std::string* channel_name);
bool GetMediaComponentFromChannelName(
const std::string& channel_name, int* component);
bool GetMediaTypeFromChannelName(
const std::string& channel_name, cricket::MediaType* media_type);
void GetSupportedAudioCryptoSuites(std::vector<std::string>* crypto_suites);
void GetSupportedVideoCryptoSuites(std::vector<std::string>* crypto_suites);
void GetSupportedDataCryptoSuites(std::vector<std::string>* crypto_suites);
void GetSupportedDefaultCryptoSuites(std::vector<std::string>* crypto_suites);
} // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIASESSION_H_