BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1561006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
149 lines
4.6 KiB
C++
149 lines
4.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/internal/video_send_stream.h"
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#include <vector>
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/video_engine/include/vie_base.h"
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#include "webrtc/video_engine/include/vie_capture.h"
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#include "webrtc/video_engine/include/vie_codec.h"
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#include "webrtc/video_engine/include/vie_network.h"
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#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
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#include "webrtc/video_engine/new_include/video_send_stream.h"
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namespace webrtc {
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namespace internal {
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VideoSendStream::VideoSendStream(
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newapi::Transport* transport,
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webrtc::VideoEngine* video_engine,
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const newapi::VideoSendStream::Config& config)
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: transport_(transport), config_(config) {
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if (config_.codec.numberOfSimulcastStreams > 0) {
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assert(config_.rtp.ssrcs.size() == config_.codec.numberOfSimulcastStreams);
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} else {
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assert(config_.rtp.ssrcs.size() == 1);
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}
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video_engine_base_ = ViEBase::GetInterface(video_engine);
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video_engine_base_->CreateChannel(channel_);
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assert(channel_ != -1);
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rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
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assert(rtp_rtcp_ != NULL);
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assert(config_.rtp.ssrcs.size() == 1);
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rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs[0]);
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capture_ = ViECapture::GetInterface(video_engine);
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capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
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capture_->ConnectCaptureDevice(capture_id_, channel_);
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network_ = ViENetwork::GetInterface(video_engine);
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assert(network_ != NULL);
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network_->RegisterSendTransport(channel_, *this);
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codec_ = ViECodec::GetInterface(video_engine);
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if (codec_->SetSendCodec(channel_, config_.codec) != 0) {
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abort();
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}
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}
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VideoSendStream::~VideoSendStream() {
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network_->DeregisterSendTransport(channel_);
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video_engine_base_->DeleteChannel(channel_);
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capture_->DisconnectCaptureDevice(channel_);
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capture_->ReleaseCaptureDevice(capture_id_);
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video_engine_base_->Release();
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capture_->Release();
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codec_->Release();
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network_->Release();
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rtp_rtcp_->Release();
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}
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void VideoSendStream::PutFrame(const I420VideoFrame& frame,
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uint32_t time_since_capture_ms) {
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// TODO(pbos): frame_copy should happen after the VideoProcessingModule has
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// resized the frame.
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I420VideoFrame frame_copy;
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frame_copy.CopyFrame(frame);
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if (config_.pre_encode_callback != NULL) {
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config_.pre_encode_callback->FrameCallback(&frame_copy);
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}
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ViEVideoFrameI420 vf;
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// TODO(pbos): This represents a memcpy step and is only required because
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// external_capture_ only takes ViEVideoFrameI420s.
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vf.y_plane = frame_copy.buffer(kYPlane);
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vf.u_plane = frame_copy.buffer(kUPlane);
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vf.v_plane = frame_copy.buffer(kVPlane);
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vf.y_pitch = frame.stride(kYPlane);
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vf.u_pitch = frame.stride(kUPlane);
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vf.v_pitch = frame.stride(kVPlane);
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vf.width = frame.width();
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vf.height = frame.height();
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external_capture_->IncomingFrameI420(vf, frame.render_time_ms());
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if (config_.local_renderer != NULL) {
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config_.local_renderer->RenderFrame(frame, 0);
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}
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}
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newapi::VideoSendStreamInput* VideoSendStream::Input() { return this; }
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void VideoSendStream::StartSend() {
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if (video_engine_base_->StartSend(channel_) != 0)
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abort();
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}
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void VideoSendStream::StopSend() {
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if (video_engine_base_->StopSend(channel_) != 0)
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abort();
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}
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bool VideoSendStream::SetTargetBitrate(
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int min_bitrate,
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int max_bitrate,
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const std::vector<SimulcastStream>& streams) {
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return false;
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}
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void VideoSendStream::GetSendCodec(VideoCodec* send_codec) {
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*send_codec = config_.codec;
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}
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int VideoSendStream::SendPacket(int /*channel*/,
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const void* packet,
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int length) {
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// TODO(pbos): Lock these methods and the destructor so it can't be processing
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// a packet when the destructor has been called.
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assert(length >= 0);
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return transport_->SendRTP(packet, static_cast<size_t>(length)) ? 0 : -1;
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}
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int VideoSendStream::SendRTCPPacket(int /*channel*/,
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const void* packet,
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int length) {
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assert(length >= 0);
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return transport_->SendRTCP(packet, static_cast<size_t>(length)) ? 0 : -1;
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}
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} // namespace internal
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} // namespace webrtc
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