Move static functions in media_session into anonymous namespace
and clean up methods that are now detected as unused. BUG=None Change-Id: If5dac3d43d4df6c7c108504c202c2383fe4a3f27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321580 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40823}
This commit is contained in:
Philipp Hancke
WebRTC LUCI CQ
parent
21b5ec1cc7
commit
f97058e9ed
@ -115,15 +115,17 @@ cricket::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
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namespace cricket {
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static bool IsRtxCodec(const Codec& codec) {
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namespace {
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bool IsRtxCodec(const Codec& codec) {
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return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
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}
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static bool IsRtxCodec(const webrtc::RtpCodecCapability& capability) {
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bool IsRtxCodec(const webrtc::RtpCodecCapability& capability) {
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return absl::EqualsIgnoreCase(capability.name, kRtxCodecName);
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}
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static bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
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bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
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for (const auto& codec : codecs) {
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if (IsRtxCodec(codec)) {
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return true;
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@ -132,19 +134,19 @@ static bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
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return false;
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}
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static bool IsRedCodec(const Codec& codec) {
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bool IsRedCodec(const Codec& codec) {
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return absl::EqualsIgnoreCase(codec.name, kRedCodecName);
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}
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static bool IsRedCodec(const webrtc::RtpCodecCapability& capability) {
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bool IsRedCodec(const webrtc::RtpCodecCapability& capability) {
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return absl::EqualsIgnoreCase(capability.name, kRedCodecName);
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}
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static bool IsFlexfecCodec(const Codec& codec) {
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bool IsFlexfecCodec(const Codec& codec) {
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return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
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}
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static bool ContainsFlexfecCodec(const std::vector<Codec>& codecs) {
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bool ContainsFlexfecCodec(const std::vector<Codec>& codecs) {
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for (const auto& codec : codecs) {
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if (IsFlexfecCodec(codec)) {
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return true;
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@ -153,15 +155,15 @@ static bool ContainsFlexfecCodec(const std::vector<Codec>& codecs) {
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return false;
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}
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static bool IsUlpfecCodec(const Codec& codec) {
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bool IsUlpfecCodec(const Codec& codec) {
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return absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName);
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}
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static bool IsComfortNoiseCodec(const Codec& codec) {
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bool IsComfortNoiseCodec(const Codec& codec) {
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return absl::EqualsIgnoreCase(codec.name, kComfortNoiseCodecName);
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}
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static RtpTransceiverDirection NegotiateRtpTransceiverDirection(
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RtpTransceiverDirection NegotiateRtpTransceiverDirection(
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RtpTransceiverDirection offer,
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RtpTransceiverDirection wants) {
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bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer);
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@ -172,17 +174,16 @@ static RtpTransceiverDirection NegotiateRtpTransceiverDirection(
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offer_send && wants_recv);
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}
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static bool IsMediaContentOfType(const ContentInfo* content,
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MediaType media_type) {
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bool IsMediaContentOfType(const ContentInfo* content, MediaType media_type) {
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if (!content || !content->media_description()) {
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return false;
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}
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return content->media_description()->type() == media_type;
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}
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static bool CreateCryptoParams(int tag,
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const std::string& cipher,
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CryptoParams* crypto_out) {
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bool CreateCryptoParams(int tag,
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const std::string& cipher,
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CryptoParams* crypto_out) {
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int key_len;
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int salt_len;
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if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher),
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@ -206,8 +207,8 @@ static bool CreateCryptoParams(int tag,
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return true;
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}
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static bool AddCryptoParams(const std::string& crypto_suite,
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CryptoParamsVec* cryptos_out) {
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bool AddCryptoParams(const std::string& crypto_suite,
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CryptoParamsVec* cryptos_out) {
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int size = static_cast<int>(cryptos_out->size());
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cryptos_out->resize(size + 1);
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@ -291,31 +292,14 @@ void GetSupportedVideoSdesCryptoSuiteNames(
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crypto_options, crypto_suite_names);
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}
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void GetSupportedDataSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites) {
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crypto_suites->push_back(rtc::kSrtpAes128CmSha1_80);
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if (crypto_options.srtp.enable_gcm_crypto_suites) {
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crypto_suites->push_back(rtc::kSrtpAeadAes256Gcm);
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crypto_suites->push_back(rtc::kSrtpAeadAes128Gcm);
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}
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}
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void GetSupportedDataSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
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crypto_options, crypto_suite_names);
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}
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// Support any GCM cipher (if enabled through options). For video support only
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// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless
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// bundle is enabled because it is low overhead.
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// Pick the crypto in the list that is supported.
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static bool SelectCrypto(const MediaContentDescription* offer,
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bool bundle,
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const webrtc::CryptoOptions& crypto_options,
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CryptoParams* crypto_out) {
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bool SelectCrypto(const MediaContentDescription* offer,
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bool bundle,
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const webrtc::CryptoOptions& crypto_options,
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CryptoParams* crypto_out) {
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bool audio = offer->type() == MEDIA_TYPE_AUDIO;
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const CryptoParamsVec& cryptos = offer->cryptos();
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@ -332,7 +316,7 @@ static bool SelectCrypto(const MediaContentDescription* offer,
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}
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// Finds all StreamParams of all media types and attach them to stream_params.
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static StreamParamsVec GetCurrentStreamParams(
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StreamParamsVec GetCurrentStreamParams(
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const std::vector<const ContentInfo*>& active_local_contents) {
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StreamParamsVec stream_params;
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for (const ContentInfo* content : active_local_contents) {
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@ -343,7 +327,7 @@ static StreamParamsVec GetCurrentStreamParams(
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return stream_params;
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}
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static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
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StreamParams CreateStreamParamsForNewSenderWithSsrcs(
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const SenderOptions& sender,
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const std::string& rtcp_cname,
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bool include_rtx_streams,
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@ -376,9 +360,8 @@ static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
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return result;
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}
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static bool ValidateSimulcastLayers(
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const std::vector<RidDescription>& rids,
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const SimulcastLayerList& simulcast_layers) {
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bool ValidateSimulcastLayers(const std::vector<RidDescription>& rids,
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const SimulcastLayerList& simulcast_layers) {
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return absl::c_all_of(
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simulcast_layers.GetAllLayers(), [&rids](const SimulcastLayer& layer) {
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return absl::c_any_of(rids, [&layer](const RidDescription& rid) {
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@ -387,7 +370,7 @@ static bool ValidateSimulcastLayers(
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});
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}
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static StreamParams CreateStreamParamsForNewSenderWithRids(
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StreamParams CreateStreamParamsForNewSenderWithRids(
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const SenderOptions& sender,
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const std::string& rtcp_cname) {
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RTC_DCHECK(!sender.rids.empty());
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@ -409,7 +392,7 @@ static StreamParams CreateStreamParamsForNewSenderWithRids(
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// Adds SimulcastDescription if indicated by the media description options.
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// MediaContentDescription should already be set up with the send rids.
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static void AddSimulcastToMediaDescription(
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void AddSimulcastToMediaDescription(
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const MediaDescriptionOptions& media_description_options,
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MediaContentDescription* description) {
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RTC_DCHECK(description);
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@ -442,12 +425,12 @@ static void AddSimulcastToMediaDescription(
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// Adds a StreamParams for each SenderOptions in `sender_options` to
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// content_description.
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// `current_params` - All currently known StreamParams of any media type.
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static bool AddStreamParams(const std::vector<SenderOptions>& sender_options,
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const std::string& rtcp_cname,
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UniqueRandomIdGenerator* ssrc_generator,
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StreamParamsVec* current_streams,
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MediaContentDescription* content_description,
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const webrtc::FieldTrialsView& field_trials) {
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bool AddStreamParams(const std::vector<SenderOptions>& sender_options,
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const std::string& rtcp_cname,
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UniqueRandomIdGenerator* ssrc_generator,
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StreamParamsVec* current_streams,
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MediaContentDescription* content_description,
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const webrtc::FieldTrialsView& field_trials) {
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// SCTP streams are not negotiated using SDP/ContentDescriptions.
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if (IsSctpProtocol(content_description->protocol())) {
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return true;
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@ -494,8 +477,8 @@ static bool AddStreamParams(const std::vector<SenderOptions>& sender_options,
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// `bundle_group`. The transport infos of the content names within the
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// `bundle_group` should be updated to use the ufrag, pwd and DTLS role of the
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// first content within the `bundle_group`.
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static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
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SessionDescription* sdesc) {
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bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
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SessionDescription* sdesc) {
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// The bundle should not be empty.
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if (!sdesc || !bundle_group.FirstContentName()) {
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return false;
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@ -529,9 +512,9 @@ static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
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// Gets the CryptoParamsVec of the given `content_name` from `sdesc`, and
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// sets it to `cryptos`.
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static bool GetCryptosByName(const SessionDescription* sdesc,
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const std::string& content_name,
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CryptoParamsVec* cryptos) {
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bool GetCryptosByName(const SessionDescription* sdesc,
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const std::string& content_name,
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CryptoParamsVec* cryptos) {
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if (!sdesc || !cryptos) {
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return false;
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}
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@ -545,8 +528,8 @@ static bool GetCryptosByName(const SessionDescription* sdesc,
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// Prunes the `target_cryptos` by removing the crypto params (crypto_suite)
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// which are not available in `filter`.
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static void PruneCryptos(const CryptoParamsVec& filter,
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CryptoParamsVec* target_cryptos) {
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void PruneCryptos(const CryptoParamsVec& filter,
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CryptoParamsVec* target_cryptos) {
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if (!target_cryptos) {
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return;
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}
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@ -565,8 +548,7 @@ static void PruneCryptos(const CryptoParamsVec& filter,
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target_cryptos->end());
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}
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static bool IsRtpContent(SessionDescription* sdesc,
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const std::string& content_name) {
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bool IsRtpContent(SessionDescription* sdesc, const std::string& content_name) {
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bool is_rtp = false;
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ContentInfo* content = sdesc->GetContentByName(content_name);
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if (content && content->media_description()) {
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@ -579,8 +561,8 @@ static bool IsRtpContent(SessionDescription* sdesc,
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// `bundle_group`. The crypto parameters of all the contents within the
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// `bundle_group` should be updated to use the common subset of the
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// available cryptos.
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static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
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SessionDescription* sdesc) {
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bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
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SessionDescription* sdesc) {
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// The bundle should not be empty.
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if (!sdesc || !bundle_group.FirstContentName()) {
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return false;
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@ -640,7 +622,7 @@ static bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
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return true;
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}
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static std::vector<const ContentInfo*> GetActiveContents(
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std::vector<const ContentInfo*> GetActiveContents(
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const SessionDescription& description,
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const MediaSessionOptions& session_options) {
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std::vector<const ContentInfo*> active_contents;
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@ -663,7 +645,7 @@ static std::vector<const ContentInfo*> GetActiveContents(
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// crypto (in current_cryptos) and it is enabled (in secure_policy), crypto is
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// created (according to crypto_suites). The created content is added to the
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// offer.
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static RTCError CreateContentOffer(
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RTCError CreateContentOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const SecurePolicy& secure_policy,
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@ -716,7 +698,7 @@ static RTCError CreateContentOffer(
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return RTCError::OK();
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}
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static RTCError CreateMediaContentOffer(
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RTCError CreateMediaContentOffer(
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const MediaDescriptionOptions& media_description_options,
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const MediaSessionOptions& session_options,
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const std::vector<Codec>& codecs,
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@ -743,11 +725,11 @@ static RTCError CreateMediaContentOffer(
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}
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template <class C>
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static bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
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const int codec1_id,
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const std::vector<C>& codecs2,
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const int codec2_id,
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const webrtc::FieldTrialsView* field_trials) {
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bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
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const int codec1_id,
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const std::vector<C>& codecs2,
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const int codec2_id,
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const webrtc::FieldTrialsView* field_trials) {
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const C* codec1 = FindCodecById(codecs1, codec1_id);
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const C* codec2 = FindCodecById(codecs2, codec2_id);
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return codec1 != nullptr && codec2 != nullptr &&
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@ -755,9 +737,9 @@ static bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
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}
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template <class C>
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static void NegotiatePacketization(const C& local_codec,
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const C& remote_codec,
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C* negotiated_codec) {}
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void NegotiatePacketization(const C& local_codec,
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const C& remote_codec,
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C* negotiated_codec) {}
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template <>
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void NegotiatePacketization(const VideoCodec& local_codec,
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@ -769,72 +751,11 @@ void NegotiatePacketization(const VideoCodec& local_codec,
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: absl::nullopt;
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}
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template <class C>
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static void NegotiateCodecs(const std::vector<C>& local_codecs,
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const std::vector<C>& offered_codecs,
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std::vector<C>* negotiated_codecs,
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bool keep_offer_order,
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const webrtc::FieldTrialsView* field_trials) {
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for (const C& ours : local_codecs) {
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absl::optional<C> theirs =
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FindMatchingCodec(local_codecs, offered_codecs, ours, field_trials);
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// Note that we intentionally only find one matching codec for each of our
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// local codecs, in case the remote offer contains duplicate codecs.
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if (theirs) {
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C negotiated = ours;
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NegotiatePacketization(ours, *theirs, &negotiated);
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negotiated.IntersectFeedbackParams(*theirs);
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if (IsRtxCodec(negotiated)) {
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const auto apt_it =
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theirs->params.find(kCodecParamAssociatedPayloadType);
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// FindMatchingCodec shouldn't return something with no apt value.
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RTC_DCHECK(apt_it != theirs->params.end());
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negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
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// We support parsing the declarative rtx-time parameter.
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const auto rtx_time_it = theirs->params.find(kCodecParamRtxTime);
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if (rtx_time_it != theirs->params.end()) {
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negotiated.SetParam(kCodecParamRtxTime, rtx_time_it->second);
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}
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} else if (IsRedCodec(negotiated)) {
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const auto red_it =
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theirs->params.find(kCodecParamNotInNameValueFormat);
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if (red_it != theirs->params.end()) {
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negotiated.SetParam(kCodecParamNotInNameValueFormat, red_it->second);
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}
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}
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if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
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webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params,
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&negotiated.params);
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}
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negotiated.id = theirs->id;
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negotiated.name = theirs->name;
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negotiated_codecs->push_back(std::move(negotiated));
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}
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}
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if (keep_offer_order) {
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// RFC3264: Although the answerer MAY list the formats in their desired
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// order of preference, it is RECOMMENDED that unless there is a
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// specific reason, the answerer list formats in the same relative order
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// they were present in the offer.
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// This can be skipped when the transceiver has any codec preferences.
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std::unordered_map<int, int> payload_type_preferences;
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int preference = static_cast<int>(offered_codecs.size() + 1);
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for (const C& codec : offered_codecs) {
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payload_type_preferences[codec.id] = preference--;
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}
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absl::c_sort(*negotiated_codecs, [&payload_type_preferences](const C& a,
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const C& b) {
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return payload_type_preferences[a.id] > payload_type_preferences[b.id];
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});
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}
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}
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// Finds a codec in `codecs2` that matches `codec_to_match`, which is
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// a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
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// the codecs themselves and their associated codecs must match.
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template <class C>
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static absl::optional<C> FindMatchingCodec(
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absl::optional<C> FindMatchingCodec(
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const std::vector<C>& codecs1,
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const std::vector<C>& codecs2,
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const C& codec_to_match,
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@ -916,10 +837,71 @@ static absl::optional<C> FindMatchingCodec(
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return absl::nullopt;
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}
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template <class C>
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void NegotiateCodecs(const std::vector<C>& local_codecs,
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const std::vector<C>& offered_codecs,
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std::vector<C>* negotiated_codecs,
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bool keep_offer_order,
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const webrtc::FieldTrialsView* field_trials) {
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for (const C& ours : local_codecs) {
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absl::optional<C> theirs =
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FindMatchingCodec(local_codecs, offered_codecs, ours, field_trials);
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// Note that we intentionally only find one matching codec for each of our
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// local codecs, in case the remote offer contains duplicate codecs.
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if (theirs) {
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C negotiated = ours;
|
||||
NegotiatePacketization(ours, *theirs, &negotiated);
|
||||
negotiated.IntersectFeedbackParams(*theirs);
|
||||
if (IsRtxCodec(negotiated)) {
|
||||
const auto apt_it =
|
||||
theirs->params.find(kCodecParamAssociatedPayloadType);
|
||||
// FindMatchingCodec shouldn't return something with no apt value.
|
||||
RTC_DCHECK(apt_it != theirs->params.end());
|
||||
negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
|
||||
|
||||
// We support parsing the declarative rtx-time parameter.
|
||||
const auto rtx_time_it = theirs->params.find(kCodecParamRtxTime);
|
||||
if (rtx_time_it != theirs->params.end()) {
|
||||
negotiated.SetParam(kCodecParamRtxTime, rtx_time_it->second);
|
||||
}
|
||||
} else if (IsRedCodec(negotiated)) {
|
||||
const auto red_it =
|
||||
theirs->params.find(kCodecParamNotInNameValueFormat);
|
||||
if (red_it != theirs->params.end()) {
|
||||
negotiated.SetParam(kCodecParamNotInNameValueFormat, red_it->second);
|
||||
}
|
||||
}
|
||||
if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
|
||||
webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params,
|
||||
&negotiated.params);
|
||||
}
|
||||
negotiated.id = theirs->id;
|
||||
negotiated.name = theirs->name;
|
||||
negotiated_codecs->push_back(std::move(negotiated));
|
||||
}
|
||||
}
|
||||
if (keep_offer_order) {
|
||||
// RFC3264: Although the answerer MAY list the formats in their desired
|
||||
// order of preference, it is RECOMMENDED that unless there is a
|
||||
// specific reason, the answerer list formats in the same relative order
|
||||
// they were present in the offer.
|
||||
// This can be skipped when the transceiver has any codec preferences.
|
||||
std::unordered_map<int, int> payload_type_preferences;
|
||||
int preference = static_cast<int>(offered_codecs.size() + 1);
|
||||
for (const C& codec : offered_codecs) {
|
||||
payload_type_preferences[codec.id] = preference--;
|
||||
}
|
||||
absl::c_sort(*negotiated_codecs, [&payload_type_preferences](const C& a,
|
||||
const C& b) {
|
||||
return payload_type_preferences[a.id] > payload_type_preferences[b.id];
|
||||
});
|
||||
}
|
||||
}
|
||||
|
||||
// Find the codec in `codec_list` that `rtx_codec` is associated with.
|
||||
template <class C>
|
||||
static const C* GetAssociatedCodecForRtx(const std::vector<C>& codec_list,
|
||||
const C& rtx_codec) {
|
||||
const C* GetAssociatedCodecForRtx(const std::vector<C>& codec_list,
|
||||
const C& rtx_codec) {
|
||||
std::string associated_pt_str;
|
||||
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
|
||||
&associated_pt_str)) {
|
||||
@ -948,8 +930,8 @@ static const C* GetAssociatedCodecForRtx(const std::vector<C>& codec_list,
|
||||
|
||||
// Find the codec in `codec_list` that `red_codec` is associated with.
|
||||
template <class C>
|
||||
static const C* GetAssociatedCodecForRed(const std::vector<C>& codec_list,
|
||||
const C& red_codec) {
|
||||
const C* GetAssociatedCodecForRed(const std::vector<C>& codec_list,
|
||||
const C& red_codec) {
|
||||
std::string fmtp;
|
||||
if (!red_codec.GetParam(kCodecParamNotInNameValueFormat, &fmtp)) {
|
||||
// Normal for video/RED.
|
||||
@ -988,10 +970,10 @@ static const C* GetAssociatedCodecForRed(const std::vector<C>& codec_list,
|
||||
// already exist in `offered_codecs` and ensure the payload types don't
|
||||
// collide.
|
||||
template <class C>
|
||||
static void MergeCodecs(const std::vector<C>& reference_codecs,
|
||||
std::vector<C>* offered_codecs,
|
||||
UsedPayloadTypes* used_pltypes,
|
||||
const webrtc::FieldTrialsView* field_trials) {
|
||||
void MergeCodecs(const std::vector<C>& reference_codecs,
|
||||
std::vector<C>* offered_codecs,
|
||||
UsedPayloadTypes* used_pltypes,
|
||||
const webrtc::FieldTrialsView* field_trials) {
|
||||
// Add all new codecs that are not RTX/RED codecs.
|
||||
// The two-pass splitting of the loops means preferring payload types
|
||||
// of actual codecs with respect to collisions.
|
||||
@ -1060,7 +1042,7 @@ static void MergeCodecs(const std::vector<C>& reference_codecs,
|
||||
// a list filtered for the media section`s direction but with default payload
|
||||
// types.
|
||||
template <typename Codecs>
|
||||
static Codecs MatchCodecPreference(
|
||||
Codecs MatchCodecPreference(
|
||||
const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
|
||||
const Codecs& codecs,
|
||||
const Codecs& supported_codecs,
|
||||
@ -1163,11 +1145,11 @@ std::vector<C> ComputeCodecsUnion(const std::vector<C>& codecs1,
|
||||
// `offered_extensions` is for either audio or video while `regular_extensions`
|
||||
// and `encrypted_extensions` are used for both audio and video. There could be
|
||||
// overlap between audio extensions and video extensions.
|
||||
static void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
|
||||
RtpHeaderExtensions* offered_extensions,
|
||||
RtpHeaderExtensions* regular_extensions,
|
||||
RtpHeaderExtensions* encrypted_extensions,
|
||||
UsedRtpHeaderExtensionIds* used_ids) {
|
||||
void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
|
||||
RtpHeaderExtensions* offered_extensions,
|
||||
RtpHeaderExtensions* regular_extensions,
|
||||
RtpHeaderExtensions* encrypted_extensions,
|
||||
UsedRtpHeaderExtensionIds* used_ids) {
|
||||
for (auto reference_extension : reference_extensions) {
|
||||
if (!webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
|
||||
*offered_extensions, reference_extension.uri,
|
||||
@ -1201,10 +1183,9 @@ static void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
|
||||
}
|
||||
}
|
||||
|
||||
static void AddEncryptedVersionsOfHdrExts(
|
||||
RtpHeaderExtensions* offered_extensions,
|
||||
RtpHeaderExtensions* encrypted_extensions,
|
||||
UsedRtpHeaderExtensionIds* used_ids) {
|
||||
void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* offered_extensions,
|
||||
RtpHeaderExtensions* encrypted_extensions,
|
||||
UsedRtpHeaderExtensionIds* used_ids) {
|
||||
RtpHeaderExtensions encrypted_extensions_to_add;
|
||||
for (const auto& extension : *offered_extensions) {
|
||||
// Skip existing encrypted offered extension
|
||||
@ -1253,7 +1234,7 @@ static void AddEncryptedVersionsOfHdrExts(
|
||||
|
||||
// Mostly identical to RtpExtension::FindHeaderExtensionByUri but discards any
|
||||
// encrypted extensions that this implementation cannot encrypt.
|
||||
static const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported(
|
||||
const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported(
|
||||
const std::vector<webrtc::RtpExtension>& extensions,
|
||||
absl::string_view uri,
|
||||
webrtc::RtpExtension::Filter filter) {
|
||||
@ -1277,11 +1258,10 @@ static const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported(
|
||||
filter);
|
||||
}
|
||||
|
||||
static void NegotiateRtpHeaderExtensions(
|
||||
const RtpHeaderExtensions& local_extensions,
|
||||
const RtpHeaderExtensions& offered_extensions,
|
||||
webrtc::RtpExtension::Filter filter,
|
||||
RtpHeaderExtensions* negotiated_extensions) {
|
||||
void NegotiateRtpHeaderExtensions(const RtpHeaderExtensions& local_extensions,
|
||||
const RtpHeaderExtensions& offered_extensions,
|
||||
webrtc::RtpExtension::Filter filter,
|
||||
RtpHeaderExtensions* negotiated_extensions) {
|
||||
// TransportSequenceNumberV2 is not offered by default. The special logic for
|
||||
// the TransportSequenceNumber extensions works as follows:
|
||||
// Offer Answer
|
||||
@ -1361,7 +1341,7 @@ static void NegotiateRtpHeaderExtensions(
|
||||
}
|
||||
}
|
||||
|
||||
static void StripCNCodecs(AudioCodecs* audio_codecs) {
|
||||
void StripCNCodecs(AudioCodecs* audio_codecs) {
|
||||
audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
|
||||
[](const AudioCodec& codec) {
|
||||
return IsComfortNoiseCodec(codec);
|
||||
@ -1369,15 +1349,15 @@ static void StripCNCodecs(AudioCodecs* audio_codecs) {
|
||||
audio_codecs->end());
|
||||
}
|
||||
|
||||
static bool SetCodecsInAnswer(
|
||||
const MediaContentDescription* offer,
|
||||
const std::vector<Codec>& local_codecs,
|
||||
const MediaDescriptionOptions& media_description_options,
|
||||
const MediaSessionOptions& session_options,
|
||||
UniqueRandomIdGenerator* ssrc_generator,
|
||||
StreamParamsVec* current_streams,
|
||||
MediaContentDescription* answer,
|
||||
const webrtc::FieldTrialsView& field_trials) {
|
||||
template <class C>
|
||||
bool SetCodecsInAnswer(const MediaContentDescriptionImpl<C>* offer,
|
||||
const std::vector<C>& local_codecs,
|
||||
const MediaDescriptionOptions& media_description_options,
|
||||
const MediaSessionOptions& session_options,
|
||||
UniqueRandomIdGenerator* ssrc_generator,
|
||||
StreamParamsVec* current_streams,
|
||||
MediaContentDescription* answer,
|
||||
const webrtc::FieldTrialsView& field_trials) {
|
||||
std::vector<Codec> negotiated_codecs;
|
||||
NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
|
||||
media_description_options.codec_preferences.empty(),
|
||||
@ -1399,7 +1379,7 @@ static bool SetCodecsInAnswer(
|
||||
// (according to crypto_suites). The codecs, rtcp_mux, and crypto are all
|
||||
// negotiated with the offer. If the negotiation fails, this method returns
|
||||
// false. The created content is added to the offer.
|
||||
static bool CreateMediaContentAnswer(
|
||||
bool CreateMediaContentAnswer(
|
||||
const MediaContentDescription* offer,
|
||||
const MediaDescriptionOptions& media_description_options,
|
||||
const MediaSessionOptions& session_options,
|
||||
@ -1468,9 +1448,9 @@ static bool CreateMediaContentAnswer(
|
||||
return true;
|
||||
}
|
||||
|
||||
static bool IsMediaProtocolSupported(MediaType type,
|
||||
const std::string& protocol,
|
||||
bool secure_transport) {
|
||||
bool IsMediaProtocolSupported(MediaType type,
|
||||
const std::string& protocol,
|
||||
bool secure_transport) {
|
||||
// Since not all applications serialize and deserialize the media protocol,
|
||||
// we will have to accept `protocol` to be empty.
|
||||
if (protocol.empty()) {
|
||||
@ -1497,8 +1477,7 @@ static bool IsMediaProtocolSupported(MediaType type,
|
||||
}
|
||||
}
|
||||
|
||||
static void SetMediaProtocol(bool secure_transport,
|
||||
MediaContentDescription* desc) {
|
||||
void SetMediaProtocol(bool secure_transport, MediaContentDescription* desc) {
|
||||
if (!desc->cryptos().empty())
|
||||
desc->set_protocol(kMediaProtocolSavpf);
|
||||
else if (secure_transport)
|
||||
@ -1509,7 +1488,7 @@ static void SetMediaProtocol(bool secure_transport,
|
||||
|
||||
// Gets the TransportInfo of the given `content_name` from the
|
||||
// `current_description`. If doesn't exist, returns a new one.
|
||||
static const TransportDescription* GetTransportDescription(
|
||||
const TransportDescription* GetTransportDescription(
|
||||
const std::string& content_name,
|
||||
const SessionDescription* current_description) {
|
||||
const TransportDescription* desc = NULL;
|
||||
@ -1524,8 +1503,8 @@ static const TransportDescription* GetTransportDescription(
|
||||
}
|
||||
|
||||
// Gets the current DTLS state from the transport description.
|
||||
static bool IsDtlsActive(const ContentInfo* content,
|
||||
const SessionDescription* current_description) {
|
||||
bool IsDtlsActive(const ContentInfo* content,
|
||||
const SessionDescription* current_description) {
|
||||
if (!content) {
|
||||
return false;
|
||||
}
|
||||
@ -1540,6 +1519,8 @@ static bool IsDtlsActive(const ContentInfo* content,
|
||||
.description.secure();
|
||||
}
|
||||
|
||||
} // namespace
|
||||
|
||||
void MediaDescriptionOptions::AddAudioSender(
|
||||
const std::string& track_id,
|
||||
const std::vector<std::string>& stream_ids) {
|
||||
@ -1648,25 +1629,20 @@ void MediaSessionDescriptionFactory::set_video_codecs(
|
||||
ComputeVideoCodecsIntersectionAndUnion();
|
||||
}
|
||||
|
||||
static void RemoveUnifiedPlanExtensions(RtpHeaderExtensions* extensions) {
|
||||
RTC_DCHECK(extensions);
|
||||
|
||||
extensions->erase(
|
||||
std::remove_if(extensions->begin(), extensions->end(),
|
||||
[](auto extension) {
|
||||
return extension.uri == webrtc::RtpExtension::kMidUri ||
|
||||
extension.uri == webrtc::RtpExtension::kRidUri ||
|
||||
extension.uri ==
|
||||
webrtc::RtpExtension::kRepairedRidUri;
|
||||
}),
|
||||
extensions->end());
|
||||
}
|
||||
|
||||
RtpHeaderExtensions
|
||||
MediaSessionDescriptionFactory::filtered_rtp_header_extensions(
|
||||
RtpHeaderExtensions extensions) const {
|
||||
if (!is_unified_plan_) {
|
||||
RemoveUnifiedPlanExtensions(&extensions);
|
||||
// Remove extensions only supported with unified-plan.
|
||||
extensions.erase(
|
||||
std::remove_if(
|
||||
extensions.begin(), extensions.end(),
|
||||
[](const webrtc::RtpExtension& extension) {
|
||||
return extension.uri == webrtc::RtpExtension::kMidUri ||
|
||||
extension.uri == webrtc::RtpExtension::kRidUri ||
|
||||
extension.uri == webrtc::RtpExtension::kRepairedRidUri;
|
||||
}),
|
||||
extensions.end());
|
||||
}
|
||||
return extensions;
|
||||
}
|
||||
|
||||
@ -400,26 +400,6 @@ VideoContentDescription* GetFirstVideoContentDescription(
|
||||
SctpDataContentDescription* GetFirstSctpDataContentDescription(
|
||||
SessionDescription* sdesc);
|
||||
|
||||
// Helper functions to return crypto suites used for SDES.
|
||||
void GetSupportedAudioSdesCryptoSuites(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<int>* crypto_suites);
|
||||
void GetSupportedVideoSdesCryptoSuites(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<int>* crypto_suites);
|
||||
void GetSupportedDataSdesCryptoSuites(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<int>* crypto_suites);
|
||||
void GetSupportedAudioSdesCryptoSuiteNames(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<std::string>* crypto_suite_names);
|
||||
void GetSupportedVideoSdesCryptoSuiteNames(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<std::string>* crypto_suite_names);
|
||||
void GetSupportedDataSdesCryptoSuiteNames(
|
||||
const webrtc::CryptoOptions& crypto_options,
|
||||
std::vector<std::string>* crypto_suite_names);
|
||||
|
||||
} // namespace cricket
|
||||
|
||||
#endif // PC_MEDIA_SESSION_H_
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user