| /* |
| * Copyright (c) 2013 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/video_send_stream.h" |
| |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/video_engine/include/vie_base.h" |
| #include "webrtc/video_engine/include/vie_capture.h" |
| #include "webrtc/video_engine/include/vie_codec.h" |
| #include "webrtc/video_engine/include/vie_external_codec.h" |
| #include "webrtc/video_engine/include/vie_image_process.h" |
| #include "webrtc/video_engine/include/vie_network.h" |
| #include "webrtc/video_engine/include/vie_rtp_rtcp.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace webrtc { |
| namespace internal { |
| |
| VideoSendStream::VideoSendStream(newapi::Transport* transport, |
| CpuOveruseObserver* overuse_observer, |
| webrtc::VideoEngine* video_engine, |
| const VideoSendStream::Config& config, |
| int base_channel) |
| : transport_adapter_(transport), |
| encoded_frame_proxy_(config.post_encode_callback), |
| codec_lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| config_(config), |
| external_codec_(NULL), |
| channel_(-1) { |
| video_engine_base_ = ViEBase::GetInterface(video_engine); |
| video_engine_base_->CreateChannel(channel_, base_channel); |
| assert(channel_ != -1); |
| |
| rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine); |
| assert(rtp_rtcp_ != NULL); |
| |
| assert(config_.rtp.ssrcs.size() > 0); |
| if (config_.suspend_below_min_bitrate) |
| config_.pacing = true; |
| rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing); |
| |
| for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { |
| const std::string& extension = config_.rtp.extensions[i].name; |
| int id = config_.rtp.extensions[i].id; |
| if (extension == RtpExtension::kTOffset) { |
| if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0) |
| abort(); |
| } else if (extension == RtpExtension::kAbsSendTime) { |
| if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0) |
| abort(); |
| } else { |
| abort(); // Unsupported extension. |
| } |
| } |
| |
| rtp_rtcp_->SetRembStatus(channel_, true, false); |
| |
| // Enable NACK, FEC or both. |
| if (config_.rtp.fec.red_payload_type != -1) { |
| assert(config_.rtp.fec.ulpfec_payload_type != -1); |
| if (config_.rtp.nack.rtp_history_ms > 0) { |
| rtp_rtcp_->SetHybridNACKFECStatus( |
| channel_, |
| true, |
| static_cast<unsigned char>(config_.rtp.fec.red_payload_type), |
| static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type)); |
| } else { |
| rtp_rtcp_->SetFECStatus( |
| channel_, |
| true, |
| static_cast<unsigned char>(config_.rtp.fec.red_payload_type), |
| static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type)); |
| } |
| } else { |
| rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0); |
| } |
| |
| char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength]; |
| assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength); |
| strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1); |
| rtcp_cname[sizeof(rtcp_cname) - 1] = '\0'; |
| |
| rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname); |
| |
| capture_ = ViECapture::GetInterface(video_engine); |
| capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_); |
| capture_->ConnectCaptureDevice(capture_id_, channel_); |
| |
| network_ = ViENetwork::GetInterface(video_engine); |
| assert(network_ != NULL); |
| |
| network_->RegisterSendTransport(channel_, transport_adapter_); |
| // 28 to match packet overhead in ModuleRtpRtcpImpl. |
| network_->SetMTU(channel_, |
| static_cast<unsigned int>(config_.rtp.max_packet_size + 28)); |
| |
| if (config.encoder) { |
| external_codec_ = ViEExternalCodec::GetInterface(video_engine); |
| if (external_codec_->RegisterExternalSendCodec(channel_, |
| config.codec.plType, |
| config.encoder, |
| config.internal_source) != |
| 0) { |
| abort(); |
| } |
| } |
| |
| codec_ = ViECodec::GetInterface(video_engine); |
| if (!SetCodec(config_.codec)) |
| abort(); |
| |
| if (overuse_observer) { |
| video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer); |
| } |
| |
| image_process_ = ViEImageProcess::GetInterface(video_engine); |
| image_process_->RegisterPreEncodeCallback(channel_, |
| config_.pre_encode_callback); |
| if (config_.post_encode_callback) { |
| image_process_->RegisterPostEncodeImageCallback(channel_, |
| &encoded_frame_proxy_); |
| } |
| |
| if (config.suspend_below_min_bitrate) { |
| codec_->SuspendBelowMinBitrate(channel_); |
| } |
| |
| stats_proxy_.reset(new SendStatisticsProxy(config, this)); |
| |
| rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_, |
| stats_proxy_.get()); |
| rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_, |
| stats_proxy_.get()); |
| rtp_rtcp_->RegisterSendBitrateObserver(channel_, stats_proxy_.get()); |
| rtp_rtcp_->RegisterSendFrameCountObserver(channel_, stats_proxy_.get()); |
| |
| codec_->RegisterEncoderObserver(channel_, *stats_proxy_); |
| capture_->RegisterObserver(capture_id_, *stats_proxy_); |
| } |
| |
| VideoSendStream::~VideoSendStream() { |
| capture_->DeregisterObserver(capture_id_); |
| codec_->DeregisterEncoderObserver(channel_); |
| |
| rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, stats_proxy_.get()); |
| rtp_rtcp_->DeregisterSendBitrateObserver(channel_, stats_proxy_.get()); |
| rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_, |
| stats_proxy_.get()); |
| rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_, |
| stats_proxy_.get()); |
| |
| image_process_->DeRegisterPreEncodeCallback(channel_); |
| |
| network_->DeregisterSendTransport(channel_); |
| |
| capture_->DisconnectCaptureDevice(channel_); |
| capture_->ReleaseCaptureDevice(capture_id_); |
| |
| if (external_codec_) { |
| external_codec_->DeRegisterExternalSendCodec(channel_, |
| config_.codec.plType); |
| } |
| |
| video_engine_base_->DeleteChannel(channel_); |
| |
| image_process_->Release(); |
| video_engine_base_->Release(); |
| capture_->Release(); |
| codec_->Release(); |
| if (external_codec_) |
| external_codec_->Release(); |
| network_->Release(); |
| rtp_rtcp_->Release(); |
| } |
| |
| void VideoSendStream::PutFrame(const I420VideoFrame& frame) { |
| input_frame_.CopyFrame(frame); |
| SwapFrame(&input_frame_); |
| } |
| |
| void VideoSendStream::SwapFrame(I420VideoFrame* frame) { |
| // TODO(pbos): Warn if frame is "too far" into the future, or too old. This |
| // would help detect if frame's being used without NTP. |
| // TO REVIEWER: Is there any good check for this? Should it be |
| // skipped? |
| if (frame != &input_frame_) |
| input_frame_.SwapFrame(frame); |
| |
| // TODO(pbos): Local rendering should not be done on the capture thread. |
| if (config_.local_renderer != NULL) |
| config_.local_renderer->RenderFrame(input_frame_, 0); |
| |
| external_capture_->SwapFrame(&input_frame_); |
| } |
| |
| VideoSendStreamInput* VideoSendStream::Input() { return this; } |
| |
| void VideoSendStream::StartSending() { |
| transport_adapter_.Enable(); |
| video_engine_base_->StartSend(channel_); |
| video_engine_base_->StartReceive(channel_); |
| } |
| |
| void VideoSendStream::StopSending() { |
| video_engine_base_->StopSend(channel_); |
| video_engine_base_->StopReceive(channel_); |
| transport_adapter_.Disable(); |
| } |
| |
| bool VideoSendStream::SetCodec(const VideoCodec& codec) { |
| assert(config_.rtp.ssrcs.size() >= codec.numberOfSimulcastStreams); |
| |
| CriticalSectionScoped crit(codec_lock_.get()); |
| if (codec_->SetSendCodec(channel_, codec) != 0) |
| return false; |
| |
| for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) { |
| rtp_rtcp_->SetLocalSSRC(channel_, |
| config_.rtp.ssrcs[i], |
| kViEStreamTypeNormal, |
| static_cast<unsigned char>(i)); |
| } |
| |
| if (&config_.codec != &codec) |
| config_.codec = codec; |
| |
| if (config_.rtp.rtx.ssrcs.empty()) |
| return true; |
| |
| // Set up RTX. |
| assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size()); |
| for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) { |
| rtp_rtcp_->SetLocalSSRC(channel_, |
| config_.rtp.rtx.ssrcs[i], |
| kViEStreamTypeRtx, |
| static_cast<unsigned char>(i)); |
| } |
| |
| if (config_.rtp.rtx.payload_type != 0) |
| rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type); |
| |
| return true; |
| } |
| |
| VideoCodec VideoSendStream::GetCodec() { |
| CriticalSectionScoped crit(codec_lock_.get()); |
| return config_.codec; |
| } |
| |
| bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| return network_->ReceivedRTCPPacket( |
| channel_, packet, static_cast<int>(length)) == 0; |
| } |
| |
| VideoSendStream::Stats VideoSendStream::GetStats() const { |
| return stats_proxy_->GetStats(); |
| } |
| |
| bool VideoSendStream::GetSendSideDelay(VideoSendStream::Stats* stats) { |
| return codec_->GetSendSideDelay( |
| channel_, &stats->avg_delay_ms, &stats->max_delay_ms); |
| } |
| |
| std::string VideoSendStream::GetCName() { |
| char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength]; |
| rtp_rtcp_->GetRTCPCName(channel_, rtcp_cname); |
| return rtcp_cname; |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |