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Diffstat (limited to 'webrtc/video_engine/payload_router.cc')
-rw-r--r-- | webrtc/video_engine/payload_router.cc | 101 |
1 files changed, 0 insertions, 101 deletions
diff --git a/webrtc/video_engine/payload_router.cc b/webrtc/video_engine/payload_router.cc deleted file mode 100644 index 3af3d4829e..0000000000 --- a/webrtc/video_engine/payload_router.cc +++ /dev/null @@ -1,101 +0,0 @@ -/* - * Copyright (c) 2015 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 "webrtc/video_engine/payload_router.h" - -#include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" -#include "webrtc/system_wrappers/include/critical_section_wrapper.h" - -namespace webrtc { - -PayloadRouter::PayloadRouter() - : crit_(CriticalSectionWrapper::CreateCriticalSection()), - active_(false) {} - -PayloadRouter::~PayloadRouter() {} - -size_t PayloadRouter::DefaultMaxPayloadLength() { - const size_t kIpUdpSrtpLength = 44; - return IP_PACKET_SIZE - kIpUdpSrtpLength; -} - -void PayloadRouter::SetSendingRtpModules( - const std::list<RtpRtcp*>& rtp_modules) { - CriticalSectionScoped cs(crit_.get()); - rtp_modules_.clear(); - rtp_modules_.reserve(rtp_modules.size()); - for (auto* rtp_module : rtp_modules) { - rtp_modules_.push_back(rtp_module); - } -} - -void PayloadRouter::set_active(bool active) { - CriticalSectionScoped cs(crit_.get()); - active_ = active; -} - -bool PayloadRouter::active() { - CriticalSectionScoped cs(crit_.get()); - return active_ && !rtp_modules_.empty(); -} - -bool PayloadRouter::RoutePayload(FrameType frame_type, - int8_t payload_type, - uint32_t time_stamp, - int64_t capture_time_ms, - const uint8_t* payload_data, - size_t payload_length, - const RTPFragmentationHeader* fragmentation, - const RTPVideoHeader* rtp_video_hdr) { - CriticalSectionScoped cs(crit_.get()); - if (!active_ || rtp_modules_.empty()) - return false; - - // The simulcast index might actually be larger than the number of modules in - // case the encoder was processing a frame during a codec reconfig. - if (rtp_video_hdr != NULL && - rtp_video_hdr->simulcastIdx >= rtp_modules_.size()) - return false; - - int stream_idx = 0; - if (rtp_video_hdr != NULL) - stream_idx = rtp_video_hdr->simulcastIdx; - return rtp_modules_[stream_idx]->SendOutgoingData( - frame_type, payload_type, time_stamp, capture_time_ms, payload_data, - payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false; -} - -void PayloadRouter::SetTargetSendBitrates( - const std::vector<uint32_t>& stream_bitrates) { - CriticalSectionScoped cs(crit_.get()); - if (stream_bitrates.size() < rtp_modules_.size()) { - // There can be a size mis-match during codec reconfiguration. - return; - } - int idx = 0; - for (auto* rtp_module : rtp_modules_) { - rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]); - } -} - -size_t PayloadRouter::MaxPayloadLength() const { - size_t min_payload_length = DefaultMaxPayloadLength(); - CriticalSectionScoped cs(crit_.get()); - for (auto* rtp_module : rtp_modules_) { - size_t module_payload_length = rtp_module->MaxDataPayloadLength(); - if (module_payload_length < min_payload_length) - min_payload_length = module_payload_length; - } - return min_payload_length; -} - -} // namespace webrtc |