Move MediaContentDescription into sessiondescription.h
Bug: webrtc:8620 Change-Id: I9b0b6d8dc9bda366e925dda9a5b92fc4e3fd9f43 Reviewed-on: https://webrtc-review.googlesource.com/35003 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21378}
This commit is contained in:
Steve Anton
Commit Bot
parent
a6b3667d82
commit
afd8e8c304
@ -18,14 +18,9 @@
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#include <string>
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#include <vector>
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#include "api/cryptoparams.h"
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#include "api/mediatypes.h"
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#include "api/rtptransceiverinterface.h"
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#include "media/base/codec.h"
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#include "media/base/mediachannel.h"
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#include "media/base/mediaconstants.h"
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#include "media/base/mediaengine.h" // For DataChannelType
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#include "media/base/streamparams.h"
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#include "p2p/base/transportdescriptionfactory.h"
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#include "pc/jseptransport.h"
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#include "pc/sessiondescription.h"
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@ -33,34 +28,6 @@
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namespace cricket {
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class ChannelManager;
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typedef std::vector<AudioCodec> AudioCodecs;
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typedef std::vector<VideoCodec> VideoCodecs;
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typedef std::vector<DataCodec> DataCodecs;
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typedef std::vector<CryptoParams> CryptoParamsVec;
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typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
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enum CryptoType {
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CT_NONE,
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CT_SDES,
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CT_DTLS
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};
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// RTC4585 RTP/AVPF
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extern const char kMediaProtocolAvpf[];
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// RFC5124 RTP/SAVPF
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extern const char kMediaProtocolSavpf[];
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extern const char kMediaProtocolDtlsSavpf[];
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extern const char kMediaProtocolRtpPrefix[];
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extern const char kMediaProtocolSctp[];
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extern const char kMediaProtocolDtlsSctp[];
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extern const char kMediaProtocolUdpDtlsSctp[];
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extern const char kMediaProtocolTcpDtlsSctp[];
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// Options to control how session descriptions are generated.
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const int kAutoBandwidth = -1;
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// Default RTCP CNAME for unit tests.
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const char kDefaultRtcpCname[] = "DefaultRtcpCname";
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@ -134,222 +101,6 @@ struct MediaSessionOptions {
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std::vector<MediaDescriptionOptions> media_description_options;
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};
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// "content" (as used in XEP-0166) descriptions for voice and video.
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class MediaContentDescription : public ContentDescription {
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public:
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MediaContentDescription() {}
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virtual MediaType type() const = 0;
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virtual bool has_codecs() const = 0;
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// |protocol| is the expected media transport protocol, such as RTP/AVPF,
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// RTP/SAVPF or SCTP/DTLS.
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std::string protocol() const { return protocol_; }
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void set_protocol(const std::string& protocol) { protocol_ = protocol; }
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webrtc::RtpTransceiverDirection direction() const {
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return direction_;
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}
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void set_direction(webrtc::RtpTransceiverDirection direction) {
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direction_ = direction;
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}
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bool rtcp_mux() const { return rtcp_mux_; }
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void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
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bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
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void set_rtcp_reduced_size(bool reduced_size) {
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rtcp_reduced_size_ = reduced_size;
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}
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int bandwidth() const { return bandwidth_; }
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void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
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const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
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void AddCrypto(const CryptoParams& params) {
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cryptos_.push_back(params);
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}
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void set_cryptos(const std::vector<CryptoParams>& cryptos) {
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cryptos_ = cryptos;
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}
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const RtpHeaderExtensions& rtp_header_extensions() const {
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return rtp_header_extensions_;
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}
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void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
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rtp_header_extensions_ = extensions;
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
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rtp_header_extensions_.push_back(ext);
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) {
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webrtc::RtpExtension webrtc_extension;
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webrtc_extension.uri = ext.uri;
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webrtc_extension.id = ext.id;
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rtp_header_extensions_.push_back(webrtc_extension);
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rtp_header_extensions_set_ = true;
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}
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void ClearRtpHeaderExtensions() {
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rtp_header_extensions_.clear();
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rtp_header_extensions_set_ = true;
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}
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// We can't always tell if an empty list of header extensions is
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// because the other side doesn't support them, or just isn't hooked up to
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// signal them. For now we assume an empty list means no signaling, but
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// provide the ClearRtpHeaderExtensions method to allow "no support" to be
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// clearly indicated (i.e. when derived from other information).
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bool rtp_header_extensions_set() const {
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return rtp_header_extensions_set_;
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}
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const StreamParamsVec& streams() const {
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return streams_;
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}
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// TODO(pthatcher): Remove this by giving mediamessage.cc access
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// to MediaContentDescription
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StreamParamsVec& mutable_streams() {
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return streams_;
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}
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void AddStream(const StreamParams& stream) {
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streams_.push_back(stream);
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}
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// Legacy streams have an ssrc, but nothing else.
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void AddLegacyStream(uint32_t ssrc) {
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streams_.push_back(StreamParams::CreateLegacy(ssrc));
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}
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void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
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StreamParams sp = StreamParams::CreateLegacy(ssrc);
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sp.AddFidSsrc(ssrc, fid_ssrc);
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streams_.push_back(sp);
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}
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// Sets the CNAME of all StreamParams if it have not been set.
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void SetCnameIfEmpty(const std::string& cname) {
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for (cricket::StreamParamsVec::iterator it = streams_.begin();
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it != streams_.end(); ++it) {
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if (it->cname.empty())
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it->cname = cname;
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}
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}
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uint32_t first_ssrc() const {
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if (streams_.empty()) {
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return 0;
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}
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return streams_[0].first_ssrc();
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}
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bool has_ssrcs() const {
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if (streams_.empty()) {
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return false;
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}
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return streams_[0].has_ssrcs();
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}
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void set_conference_mode(bool enable) { conference_mode_ = enable; }
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bool conference_mode() const { return conference_mode_; }
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// https://tools.ietf.org/html/rfc4566#section-5.7
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// May be present at the media or session level of SDP. If present at both
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// levels, the media-level attribute overwrites the session-level one.
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void set_connection_address(const rtc::SocketAddress& address) {
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connection_address_ = address;
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}
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const rtc::SocketAddress& connection_address() const {
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return connection_address_;
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}
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protected:
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bool rtcp_mux_ = false;
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bool rtcp_reduced_size_ = false;
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int bandwidth_ = kAutoBandwidth;
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std::string protocol_;
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std::vector<CryptoParams> cryptos_;
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std::vector<webrtc::RtpExtension> rtp_header_extensions_;
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bool rtp_header_extensions_set_ = false;
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StreamParamsVec streams_;
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bool conference_mode_ = false;
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webrtc::RtpTransceiverDirection direction_ =
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webrtc::RtpTransceiverDirection::kSendRecv;
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rtc::SocketAddress connection_address_;
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};
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template <class C>
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class MediaContentDescriptionImpl : public MediaContentDescription {
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public:
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typedef C CodecType;
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// Codecs should be in preference order (most preferred codec first).
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const std::vector<C>& codecs() const { return codecs_; }
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void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
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virtual bool has_codecs() const { return !codecs_.empty(); }
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bool HasCodec(int id) {
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bool found = false;
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for (typename std::vector<C>::iterator iter = codecs_.begin();
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iter != codecs_.end(); ++iter) {
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if (iter->id == id) {
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found = true;
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break;
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}
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}
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return found;
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}
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void AddCodec(const C& codec) {
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codecs_.push_back(codec);
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}
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void AddOrReplaceCodec(const C& codec) {
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for (typename std::vector<C>::iterator iter = codecs_.begin();
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iter != codecs_.end(); ++iter) {
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if (iter->id == codec.id) {
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*iter = codec;
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return;
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}
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}
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AddCodec(codec);
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}
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void AddCodecs(const std::vector<C>& codecs) {
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typename std::vector<C>::const_iterator codec;
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for (codec = codecs.begin(); codec != codecs.end(); ++codec) {
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AddCodec(*codec);
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}
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}
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private:
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std::vector<C> codecs_;
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};
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class AudioContentDescription : public MediaContentDescriptionImpl<AudioCodec> {
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public:
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AudioContentDescription() {}
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virtual ContentDescription* Copy() const {
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return new AudioContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
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};
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class VideoContentDescription : public MediaContentDescriptionImpl<VideoCodec> {
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public:
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virtual ContentDescription* Copy() const {
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return new VideoContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_VIDEO; }
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};
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class DataContentDescription : public MediaContentDescriptionImpl<DataCodec> {
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public:
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DataContentDescription() {}
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virtual ContentDescription* Copy() const {
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return new DataContentDescription(*this);
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}
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virtual MediaType type() const { return MEDIA_TYPE_DATA; }
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bool use_sctpmap() const { return use_sctpmap_; }
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void set_use_sctpmap(bool enable) { use_sctpmap_ = enable; }
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private:
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bool use_sctpmap_ = true;
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};
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// Creates media session descriptions according to the supplied codecs and
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// other fields, as well as the supplied per-call options.
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// When creating answers, performs the appropriate negotiation
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@ -14,11 +14,40 @@
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#include <string>
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#include <vector>
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#include "api/cryptoparams.h"
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#include "api/rtpparameters.h"
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#include "api/rtptransceiverinterface.h"
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#include "media/base/codec.h"
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#include "media/base/mediachannel.h"
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#include "media/base/streamparams.h"
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#include "p2p/base/transportinfo.h"
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#include "rtc_base/constructormagic.h"
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namespace cricket {
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typedef std::vector<AudioCodec> AudioCodecs;
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typedef std::vector<VideoCodec> VideoCodecs;
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typedef std::vector<DataCodec> DataCodecs;
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typedef std::vector<CryptoParams> CryptoParamsVec;
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typedef std::vector<webrtc::RtpExtension> RtpHeaderExtensions;
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// RTC4585 RTP/AVPF
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extern const char kMediaProtocolAvpf[];
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// RFC5124 RTP/SAVPF
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extern const char kMediaProtocolSavpf[];
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extern const char kMediaProtocolDtlsSavpf[];
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extern const char kMediaProtocolRtpPrefix[];
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extern const char kMediaProtocolSctp[];
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extern const char kMediaProtocolDtlsSctp[];
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extern const char kMediaProtocolUdpDtlsSctp[];
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extern const char kMediaProtocolTcpDtlsSctp[];
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// Options to control how session descriptions are generated.
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const int kAutoBandwidth = -1;
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// Describes a session content. Individual content types inherit from
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// this class. Analagous to a <jingle><content><description> or
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// <session><description>.
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@ -28,6 +57,208 @@ class ContentDescription {
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virtual ContentDescription* Copy() const = 0;
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};
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// "content" (as used in XEP-0166) descriptions for voice and video.
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class MediaContentDescription : public ContentDescription {
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public:
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MediaContentDescription() {}
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virtual MediaType type() const = 0;
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virtual bool has_codecs() const = 0;
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// |protocol| is the expected media transport protocol, such as RTP/AVPF,
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// RTP/SAVPF or SCTP/DTLS.
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std::string protocol() const { return protocol_; }
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void set_protocol(const std::string& protocol) { protocol_ = protocol; }
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webrtc::RtpTransceiverDirection direction() const { return direction_; }
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void set_direction(webrtc::RtpTransceiverDirection direction) {
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direction_ = direction;
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}
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bool rtcp_mux() const { return rtcp_mux_; }
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void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
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bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
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void set_rtcp_reduced_size(bool reduced_size) {
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rtcp_reduced_size_ = reduced_size;
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}
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int bandwidth() const { return bandwidth_; }
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void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
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const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
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void AddCrypto(const CryptoParams& params) { cryptos_.push_back(params); }
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void set_cryptos(const std::vector<CryptoParams>& cryptos) {
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cryptos_ = cryptos;
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}
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const RtpHeaderExtensions& rtp_header_extensions() const {
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return rtp_header_extensions_;
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}
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void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
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rtp_header_extensions_ = extensions;
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
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rtp_header_extensions_.push_back(ext);
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rtp_header_extensions_set_ = true;
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}
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void AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) {
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webrtc::RtpExtension webrtc_extension;
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webrtc_extension.uri = ext.uri;
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webrtc_extension.id = ext.id;
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rtp_header_extensions_.push_back(webrtc_extension);
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rtp_header_extensions_set_ = true;
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}
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void ClearRtpHeaderExtensions() {
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rtp_header_extensions_.clear();
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rtp_header_extensions_set_ = true;
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}
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// We can't always tell if an empty list of header extensions is
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// because the other side doesn't support them, or just isn't hooked up to
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// signal them. For now we assume an empty list means no signaling, but
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// provide the ClearRtpHeaderExtensions method to allow "no support" to be
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// clearly indicated (i.e. when derived from other information).
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bool rtp_header_extensions_set() const { return rtp_header_extensions_set_; }
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const StreamParamsVec& streams() const { return streams_; }
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// TODO(pthatcher): Remove this by giving mediamessage.cc access
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// to MediaContentDescription
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StreamParamsVec& mutable_streams() { return streams_; }
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void AddStream(const StreamParams& stream) { streams_.push_back(stream); }
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// Legacy streams have an ssrc, but nothing else.
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void AddLegacyStream(uint32_t ssrc) {
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streams_.push_back(StreamParams::CreateLegacy(ssrc));
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}
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void AddLegacyStream(uint32_t ssrc, uint32_t fid_ssrc) {
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StreamParams sp = StreamParams::CreateLegacy(ssrc);
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sp.AddFidSsrc(ssrc, fid_ssrc);
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streams_.push_back(sp);
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}
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// Sets the CNAME of all StreamParams if it have not been set.
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void SetCnameIfEmpty(const std::string& cname) {
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for (cricket::StreamParamsVec::iterator it = streams_.begin();
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it != streams_.end(); ++it) {
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if (it->cname.empty())
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it->cname = cname;
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}
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}
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uint32_t first_ssrc() const {
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if (streams_.empty()) {
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return 0;
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}
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return streams_[0].first_ssrc();
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}
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bool has_ssrcs() const {
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if (streams_.empty()) {
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return false;
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}
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return streams_[0].has_ssrcs();
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}
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void set_conference_mode(bool enable) { conference_mode_ = enable; }
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bool conference_mode() const { return conference_mode_; }
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// https://tools.ietf.org/html/rfc4566#section-5.7
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// May be present at the media or session level of SDP. If present at both
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// levels, the media-level attribute overwrites the session-level one.
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void set_connection_address(const rtc::SocketAddress& address) {
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connection_address_ = address;
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}
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const rtc::SocketAddress& connection_address() const {
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return connection_address_;
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}
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protected:
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bool rtcp_mux_ = false;
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bool rtcp_reduced_size_ = false;
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int bandwidth_ = kAutoBandwidth;
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std::string protocol_;
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std::vector<CryptoParams> cryptos_;
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std::vector<webrtc::RtpExtension> rtp_header_extensions_;
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bool rtp_header_extensions_set_ = false;
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StreamParamsVec streams_;
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bool conference_mode_ = false;
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webrtc::RtpTransceiverDirection direction_ =
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webrtc::RtpTransceiverDirection::kSendRecv;
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rtc::SocketAddress connection_address_;
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};
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template <class C>
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class MediaContentDescriptionImpl : public MediaContentDescription {
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public:
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typedef C CodecType;
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// Codecs should be in preference order (most preferred codec first).
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const std::vector<C>& codecs() const { return codecs_; }
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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() {}
|
||||
|
||||
virtual ContentDescription* Copy() const {
|
||||
return new AudioContentDescription(*this);
|
||||
}
|
||||
virtual MediaType type() const { return MEDIA_TYPE_AUDIO; }
|
||||
};
|
||||
|
||||
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;
|
||||
};
|
||||
|
||||
// Analagous to a <jingle><content> or <session><description>.
|
||||
// name = name of <content name="...">
|
||||
// type = xmlns of <content>
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user