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Refactor audio/video offer/answer creation helpers

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BUG=webrtc:15214

Change-Id: I35dcac465221760e54b09bc6c5e4126df4193289
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326141
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41093}
This commit is contained in:
Philipp Hancke 2023-11-06 09:52:46 +01:00 committed by WebRTC LUCI CQ
parent 5dbfe9621a
commit 7946be7429
2 changed files with 209 additions and 123 deletions
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305
pc/media_session.cc
View File

@ -2237,37 +2237,19 @@ RTCError MediaSessionDescriptionFactory::AddTransportAnswer(
return RTCError::OK();
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
webrtc::RTCErrorOr<AudioCodecs>
MediaSessionDescriptionFactory::GetNegotiatedAudioCodecsForOffer(
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,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
const webrtc::FieldTrialsView* field_trials) const {
// Filter audio_codecs (which includes all codecs, with correctly remapped
// payload types) based on transceiver direction.
const AudioCodecs& supported_audio_codecs =
GetAudioCodecsForOffer(media_description_options.direction);
AudioCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
// Add the codecs from the current transceiver's codec preferences.
// They override any existing codecs from previous negotiations.
@ -2311,6 +2293,37 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
// If application doesn't want CN codecs in offer.
StripCNCodecs(&filtered_codecs);
}
return filtered_codecs;
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpHeaderExtensions& audio_rtp_extensions,
const AudioCodecs& audio_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
auto error_or_filtered_codecs = GetNegotiatedAudioCodecsForOffer(
media_description_options, session_options, current_content, audio_codecs,
&transport_desc_factory_->trials());
if (!error_or_filtered_codecs.ok()) {
return error_or_filtered_codecs.MoveError();
}
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
@ -2321,7 +2334,8 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
auto error = CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs, sdes_policy,
media_description_options, session_options,
error_or_filtered_codecs.MoveValue(), sdes_policy,
GetCryptos(current_content), crypto_suites, audio_rtp_extensions,
ssrc_generator(), current_streams, audio.get(),
transport_desc_factory_->trials());
@ -2346,20 +2360,13 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
return RTCError::OK();
}
// TODO(kron): This function is very similar to AddAudioContentForOffer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
webrtc::RTCErrorOr<VideoCodecs>
MediaSessionDescriptionFactory::GetNegotiatedVideoCodecsForOffer(
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,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
const webrtc::FieldTrialsView* field_trials) const {
// Filter video_codecs (which includes all codecs, with correctly remapped
// payload types) based on transceiver direction.
const VideoCodecs& supported_video_codecs =
@ -2431,6 +2438,28 @@ RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
}
}
return filtered_codecs;
}
// TODO(kron): This function is very similar to AddAudioContentForOffer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpHeaderExtensions& video_rtp_extensions,
const VideoCodecs& video_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
auto error_or_filtered_codecs = GetNegotiatedVideoCodecsForOffer(
media_description_options, session_options, current_content, video_codecs,
&transport_desc_factory_->trials());
if (!error_or_filtered_codecs.ok()) {
return error_or_filtered_codecs.MoveError();
}
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
@ -2439,7 +2468,8 @@ RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
auto error = CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs, sdes_policy,
media_description_options, session_options,
error_or_filtered_codecs.MoveValue(), sdes_policy,
GetCryptos(current_content), crypto_suites, video_rtp_extensions,
ssrc_generator(), current_streams, video.get(),
transport_desc_factory_->trials());
@ -2526,56 +2556,14 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForOffer(
current_description, desc, ice_credentials);
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
webrtc::RTCErrorOr<AudioCodecs>
MediaSessionDescriptionFactory::GetNegotiatedAudioCodecsForAnswer(
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,
const RtpHeaderExtensions& rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
const AudioContentDescription* offer_audio_description =
offer_content->media_description()->as_audio();
std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!audio_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, audio transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_audio_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
AudioCodecs supported_audio_codecs =
GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
const AudioCodecs& supported_audio_codecs,
const webrtc::FieldTrialsView* field_trials) const {
AudioCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
@ -2618,6 +2606,64 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
// If application doesn't want CN codecs in answer.
StripCNCodecs(&filtered_codecs);
}
return filtered_codecs;
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
const AudioCodecs& audio_codecs,
const RtpHeaderExtensions& rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
const AudioContentDescription* offer_audio_description =
offer_content->media_description()->as_audio();
std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!audio_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, audio transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_audio_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
AudioCodecs supported_audio_codecs =
GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
auto error_or_filtered_codecs = GetNegotiatedAudioCodecsForAnswer(
media_description_options, session_options, current_content, audio_codecs,
supported_audio_codecs, &transport_desc_factory_->trials());
if (!error_or_filtered_codecs.ok()) {
return error_or_filtered_codecs.MoveError();
}
auto filtered_codecs = error_or_filtered_codecs.MoveValue();
// Determine if we have media codecs in common.
bool has_common_media_codecs =
@ -2671,46 +2717,14 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
return RTCError::OK();
}
// TODO(kron): This function is very similar to AddAudioContentForAnswer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
webrtc::RTCErrorOr<AudioCodecs>
MediaSessionDescriptionFactory::GetNegotiatedVideoCodecsForAnswer(
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,
const RtpHeaderExtensions& default_video_rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
const VideoContentDescription* offer_video_description =
offer_content->media_description()->as_video();
std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!video_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, video transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_video_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
VideoCodecs supported_video_codecs =
GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
const VideoCodecs& supported_video_codecs,
const webrtc::FieldTrialsView* field_trials) const {
VideoCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
@ -2755,13 +2769,6 @@ RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
filtered_codecs =
ComputeCodecsUnion(filtered_codecs, other_video_codecs, field_trials);
}
// Determine if we have media codecs in common.
bool has_common_media_codecs =
std::find_if(
filtered_codecs.begin(), filtered_codecs.end(), [](const Codec& c) {
return !(IsRedCodec(c) || IsUlpfecCodec(c) || IsFlexfecCodec(c));
}) != filtered_codecs.end();
if (session_options.raw_packetization_for_video) {
for (Codec& codec : filtered_codecs) {
if (codec.IsMediaCodec()) {
@ -2769,6 +2776,60 @@ RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
}
}
}
return filtered_codecs;
}
// TODO(kron): This function is very similar to AddAudioContentForAnswer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
const VideoCodecs& video_codecs,
const RtpHeaderExtensions& default_video_rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
const VideoContentDescription* offer_video_description =
offer_content->media_description()->as_video();
std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!video_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, video transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_video_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
VideoCodecs supported_video_codecs =
GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
auto error_or_filtered_codecs = GetNegotiatedVideoCodecsForAnswer(
media_description_options, session_options, current_content, video_codecs,
supported_video_codecs, &transport_desc_factory_->trials());
if (!error_or_filtered_codecs.ok()) {
return error_or_filtered_codecs.MoveError();
}
auto filtered_codecs = error_or_filtered_codecs.MoveValue();
// Determine if we have media codecs in common.
bool has_common_media_codecs =
std::find_if(
filtered_codecs.begin(), filtered_codecs.end(), [](const Codec& c) {
return !(IsRedCodec(c) || IsUlpfecCodec(c) || IsFlexfecCodec(c));
}) != filtered_codecs.end();
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;

27
pc/media_session.h
View File

@ -233,7 +233,12 @@ class MediaSessionDescriptionFactory {
// 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.
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedAudioCodecsForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const AudioCodecs& audio_codecs,
const webrtc::FieldTrialsView* field_trials) const;
webrtc::RTCError AddAudioContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
@ -245,6 +250,12 @@ class MediaSessionDescriptionFactory {
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedVideoCodecsForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const VideoCodecs& video_codecs,
const webrtc::FieldTrialsView* field_trials) const;
webrtc::RTCError AddVideoContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
@ -273,6 +284,13 @@ class MediaSessionDescriptionFactory {
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedAudioCodecsForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const AudioCodecs& audio_codecs,
const AudioCodecs& supported_audio_codecs,
const webrtc::FieldTrialsView* field_trials) const;
webrtc::RTCError AddAudioContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
@ -287,6 +305,13 @@ class MediaSessionDescriptionFactory {
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const;
webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedVideoCodecsForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const VideoCodecs& video_codecs,
const VideoCodecs& supported_video_codecs,
const webrtc::FieldTrialsView* field_trials) const;
webrtc::RTCError AddVideoContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,