| /* |
| * 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/mediaengine.h" |
| |
| #include "api/video/video_bitrate_allocation.h" |
| #include "rtc_base/stringencode.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]; |
| } |
| webrtc::RtpParameters parameters; |
| parameters.encodings = encodings; |
| parameters.rtcp.cname = sp.cname; |
| return parameters; |
| } |
| |
| webrtc::RTCError ValidateRtpParameters( |
| const webrtc::RtpParameters& old_rtp_parameters, |
| const webrtc::RtpParameters& rtp_parameters) { |
| 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"); |
| } |
| |
| for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { |
| if (rtp_parameters.encodings[i].ssrc != |
| old_rtp_parameters.encodings[i].ssrc) { |
| LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION, |
| "Attempted to set RtpParameters with modified SSRC"); |
| } |
| 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].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 (i > 0 && (rtp_parameters.encodings[i].num_temporal_layers != |
| rtp_parameters.encodings[i - 1].num_temporal_layers)) { |
| LOG_AND_RETURN_ERROR( |
| RTCErrorType::INVALID_MODIFICATION, |
| "Attempted to set RtpParameters num_temporal_layers " |
| "at encoding layer i: " + |
| rtc::ToString(i) + |
| " to a different value than other encoding layers."); |
| } |
| } |
| return webrtc::RTCError::OK(); |
| } |
| |
| }; // namespace cricket |