diff options
Diffstat (limited to 'third_party/libwebrtc/media/base/media_engine.cc')
| -rw-r--r-- | third_party/libwebrtc/media/base/media_engine.cc | 291 |
1 files changed, 291 insertions, 0 deletions
diff --git a/third_party/libwebrtc/media/base/media_engine.cc b/third_party/libwebrtc/media/base/media_engine.cc new file mode 100644 index 0000000000..7304ab03d7 --- /dev/null +++ b/third_party/libwebrtc/media/base/media_engine.cc @@ -0,0 +1,291 @@ +/* + * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "media/base/media_engine.h" + +#include <stddef.h> + +#include <cstdint> +#include <string> +#include <utility> + +#include "absl/algorithm/container.h" +#include "api/video/video_bitrate_allocation.h" +#include "rtc_base/checks.h" +#include "rtc_base/string_encode.h" + +namespace cricket { + +RtpCapabilities::RtpCapabilities() = default; +RtpCapabilities::~RtpCapabilities() = default; + +webrtc::RtpParameters CreateRtpParametersWithOneEncoding() { + webrtc::RtpParameters parameters; + webrtc::RtpEncodingParameters encoding; + parameters.encodings.push_back(encoding); + return parameters; +} + +webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) { + std::vector<uint32_t> primary_ssrcs; + sp.GetPrimarySsrcs(&primary_ssrcs); + size_t encoding_count = primary_ssrcs.size(); + + std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count); + for (size_t i = 0; i < encodings.size(); ++i) { + encodings[i].ssrc = primary_ssrcs[i]; + } + + const std::vector<RidDescription>& rids = sp.rids(); + RTC_DCHECK(rids.size() == 0 || rids.size() == encoding_count); + for (size_t i = 0; i < rids.size(); ++i) { + encodings[i].rid = rids[i].rid; + } + + webrtc::RtpParameters parameters; + parameters.encodings = encodings; + parameters.rtcp.cname = sp.cname; + return parameters; +} + +std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions( + const RtpHeaderExtensionQueryInterface& query_interface) { + std::vector<webrtc::RtpExtension> extensions; + for (const auto& entry : query_interface.GetRtpHeaderExtensions()) { + if (entry.direction != webrtc::RtpTransceiverDirection::kStopped) + extensions.emplace_back(entry.uri, *entry.preferred_id); + } + return extensions; +} + +webrtc::RTCError CheckScalabilityModeValues( + const webrtc::RtpParameters& rtp_parameters, + rtc::ArrayView<cricket::Codec> codec_preferences, + absl::optional<cricket::Codec> send_codec) { + using webrtc::RTCErrorType; + + if (codec_preferences.empty()) { + // This is an audio sender or an extra check in the stack where the codec + // list is not available and we can't check the scalability_mode values. + return webrtc::RTCError::OK(); + } + + for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { + if (rtp_parameters.encodings[i].codec) { + bool codecFound = false; + for (const cricket::VideoCodec& codec : codec_preferences) { + if (codec.MatchesRtpCodec(*rtp_parameters.encodings[i].codec)) { + codecFound = true; + send_codec = codec; + break; + } + } + if (!codecFound) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to use an unsupported codec for layer " + + std::to_string(i)); + } + } + if (rtp_parameters.encodings[i].scalability_mode) { + if (!send_codec) { + bool scalabilityModeFound = false; + for (const cricket::VideoCodec& codec : codec_preferences) { + for (const auto& scalability_mode : codec.scalability_modes) { + if (ScalabilityModeToString(scalability_mode) == + *rtp_parameters.encodings[i].scalability_mode) { + scalabilityModeFound = true; + break; + } + } + if (scalabilityModeFound) + break; + } + + if (!scalabilityModeFound) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters scalabilityMode " + "to an unsupported value for the current codecs."); + } + } else { + bool scalabilityModeFound = false; + for (const auto& scalability_mode : send_codec->scalability_modes) { + if (ScalabilityModeToString(scalability_mode) == + *rtp_parameters.encodings[i].scalability_mode) { + scalabilityModeFound = true; + break; + } + } + if (!scalabilityModeFound) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters scalabilityMode " + "to an unsupported value for the current codecs."); + } + } + } + } + + return webrtc::RTCError::OK(); +} + +webrtc::RTCError CheckRtpParametersValues( + const webrtc::RtpParameters& rtp_parameters, + rtc::ArrayView<cricket::Codec> codec_preferences, + absl::optional<cricket::Codec> send_codec) { + using webrtc::RTCErrorType; + + for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { + if (rtp_parameters.encodings[i].bitrate_priority <= 0) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, + "Attempted to set RtpParameters bitrate_priority to " + "an invalid number. bitrate_priority must be > 0."); + } + if (rtp_parameters.encodings[i].scale_resolution_down_by && + *rtp_parameters.encodings[i].scale_resolution_down_by < 1.0) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_RANGE, + "Attempted to set RtpParameters scale_resolution_down_by to an " + "invalid value. scale_resolution_down_by must be >= 1.0"); + } + if (rtp_parameters.encodings[i].max_framerate && + *rtp_parameters.encodings[i].max_framerate < 0.0) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, + "Attempted to set RtpParameters max_framerate to an " + "invalid value. max_framerate must be >= 0.0"); + } + if (rtp_parameters.encodings[i].min_bitrate_bps && + rtp_parameters.encodings[i].max_bitrate_bps) { + if (*rtp_parameters.encodings[i].max_bitrate_bps < + *rtp_parameters.encodings[i].min_bitrate_bps) { + LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_RANGE, + "Attempted to set RtpParameters min bitrate " + "larger than max bitrate."); + } + } + if (rtp_parameters.encodings[i].num_temporal_layers) { + if (*rtp_parameters.encodings[i].num_temporal_layers < 1 || + *rtp_parameters.encodings[i].num_temporal_layers > + webrtc::kMaxTemporalStreams) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, + "Attempted to set RtpParameters " + "num_temporal_layers to an invalid number."); + } + } + + if (rtp_parameters.encodings[i].requested_resolution && + rtp_parameters.encodings[i].scale_resolution_down_by) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, + "Attempted to set scale_resolution_down_by and " + "requested_resolution simultaniously."); + } + + if (i > 0 && rtp_parameters.encodings[i - 1].codec != + rtp_parameters.encodings[i].codec) { + LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION, + "Attempted to use different codec values for " + "different encodings."); + } + } + + return CheckScalabilityModeValues(rtp_parameters, codec_preferences, + send_codec); +} + +webrtc::RTCError CheckRtpParametersInvalidModificationAndValues( + const webrtc::RtpParameters& old_rtp_parameters, + const webrtc::RtpParameters& rtp_parameters) { + return CheckRtpParametersInvalidModificationAndValues( + old_rtp_parameters, rtp_parameters, {}, absl::nullopt); +} + +webrtc::RTCError CheckRtpParametersInvalidModificationAndValues( + const webrtc::RtpParameters& old_rtp_parameters, + const webrtc::RtpParameters& rtp_parameters, + rtc::ArrayView<cricket::Codec> codec_preferences, + absl::optional<cricket::Codec> send_codec) { + using webrtc::RTCErrorType; + if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters with different encoding count"); + } + if (rtp_parameters.rtcp != old_rtp_parameters.rtcp) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters with modified RTCP parameters"); + } + if (rtp_parameters.header_extensions != + old_rtp_parameters.header_extensions) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters with modified header extensions"); + } + if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings, + [](const webrtc::RtpEncodingParameters& encoding1, + const webrtc::RtpEncodingParameters& encoding2) { + return encoding1.rid == encoding2.rid; + })) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION, + "Attempted to change RID values in the encodings."); + } + if (!absl::c_equal(old_rtp_parameters.encodings, rtp_parameters.encodings, + [](const webrtc::RtpEncodingParameters& encoding1, + const webrtc::RtpEncodingParameters& encoding2) { + return encoding1.ssrc == encoding2.ssrc; + })) { + LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION, + "Attempted to set RtpParameters with modified SSRC"); + } + + return CheckRtpParametersValues(rtp_parameters, codec_preferences, + send_codec); +} + +CompositeMediaEngine::CompositeMediaEngine( + std::unique_ptr<webrtc::FieldTrialsView> trials, + std::unique_ptr<VoiceEngineInterface> audio_engine, + std::unique_ptr<VideoEngineInterface> video_engine) + : trials_(std::move(trials)), + voice_engine_(std::move(audio_engine)), + video_engine_(std::move(video_engine)) {} + +CompositeMediaEngine::CompositeMediaEngine( + std::unique_ptr<VoiceEngineInterface> audio_engine, + std::unique_ptr<VideoEngineInterface> video_engine) + : CompositeMediaEngine(nullptr, + std::move(audio_engine), + std::move(video_engine)) {} + +CompositeMediaEngine::~CompositeMediaEngine() = default; + +bool CompositeMediaEngine::Init() { + voice().Init(); + return true; +} + +VoiceEngineInterface& CompositeMediaEngine::voice() { + return *voice_engine_.get(); +} + +VideoEngineInterface& CompositeMediaEngine::video() { + return *video_engine_.get(); +} + +const VoiceEngineInterface& CompositeMediaEngine::voice() const { + return *voice_engine_.get(); +} + +const VideoEngineInterface& CompositeMediaEngine::video() const { + return *video_engine_.get(); +} + +} // namespace cricket |