New VideoEngine API implementation on top of old one, first steps.

BUG=1668
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1360004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4044 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/internal/video_send_stream.cc b/video_engine/internal/video_send_stream.cc
new file mode 100644
index 0000000..c818258
--- /dev/null
+++ b/video_engine/internal/video_send_stream.cc
@@ -0,0 +1,145 @@
+/*
+ *  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_engine/internal/video_send_stream.h"
+
+#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_network.h"
+#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
+#include "webrtc/video_engine/new_include/video_send_stream.h"
+
+namespace webrtc {
+namespace internal {
+
+VideoSendStream::VideoSendStream(
+    newapi::Transport* transport, webrtc::VideoEngine* video_engine,
+    const newapi::VideoSendStreamConfig& send_stream_config)
+    : transport_(transport), config_(send_stream_config) {
+
+  if (config_.codec.numberOfSimulcastStreams > 0) {
+    assert(config_.rtp.ssrcs.size() == config_.codec.numberOfSimulcastStreams);
+  } else {
+    assert(config_.rtp.ssrcs.size() == 1);
+  }
+
+  video_engine_base_ = ViEBase::GetInterface(video_engine);
+  video_engine_base_->CreateChannel(channel_);
+  assert(channel_ != -1);
+
+  rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
+  assert(rtp_rtcp_ != NULL);
+
+  assert(config_.rtp.ssrcs.size() == 1);
+  rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs[0]);
+
+  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_, *this);
+
+  codec_ = ViECodec::GetInterface(video_engine);
+  if (codec_->SetSendCodec(channel_, config_.codec) != 0) {
+    abort();
+  }
+}
+
+VideoSendStream::~VideoSendStream() {
+  network_->DeregisterSendTransport(channel_);
+  video_engine_base_->DeleteChannel(channel_);
+
+  capture_->DisconnectCaptureDevice(channel_);
+  capture_->ReleaseCaptureDevice(capture_id_);
+
+  video_engine_base_->Release();
+  capture_->Release();
+  codec_->Release();
+  network_->Release();
+  rtp_rtcp_->Release();
+}
+
+void VideoSendStream::PutFrame(const I420VideoFrame& frame,
+                               int32_t delta_capture_time) {
+  I420VideoFrame frame_copy;
+  frame_copy.CopyFrame(frame);
+
+  if (config_.pre_encode_callback != NULL) {
+    config_.pre_encode_callback->FrameCallback(&frame_copy);
+  }
+
+  ViEVideoFrameI420 vf;
+
+  // TODO(pbos): This represents a memcpy step and is only required because
+  //             external_capture_ only takes ViEVideoFrameI420s.
+  vf.y_plane = frame_copy.buffer(kYPlane);
+  vf.u_plane = frame_copy.buffer(kUPlane);
+  vf.v_plane = frame_copy.buffer(kVPlane);
+  vf.y_pitch = frame.stride(kYPlane);
+  vf.u_pitch = frame.stride(kUPlane);
+  vf.v_pitch = frame.stride(kVPlane);
+  vf.width = frame.width();
+  vf.height = frame.height();
+
+  external_capture_->IncomingFrameI420(vf, frame.timestamp());
+
+  if (config_.local_renderer != NULL) {
+    config_.local_renderer->RenderFrame(frame, 0);
+  }
+}
+
+newapi::VideoSendStreamInput* VideoSendStream::Input() { return this; }
+
+void VideoSendStream::StartSend() {
+  if (video_engine_base_->StartSend(channel_) != 0) abort();
+}
+
+void VideoSendStream::StopSend() {
+  if (video_engine_base_->StopSend(channel_) != 0) abort();
+}
+
+void VideoSendStream::GetSendStatistics(
+    std::vector<newapi::SendStatistics>* statistics) {
+  // TODO(pbos): Implement
+}
+
+bool VideoSendStream::SetTargetBitrate(
+    int min_bitrate, int max_bitrate,
+    const std::vector<SimulcastStream>& streams) {
+  return false;
+}
+
+void VideoSendStream::GetSendCodec(VideoCodec* send_codec) {
+  *send_codec = config_.codec;
+}
+
+int VideoSendStream::SendPacket(int /*channel*/, const void* packet,
+                                int length) {
+  // TODO(pbos): Lock these methods and the destructor so it can't be processing
+  //             a packet when the destructor has been called.
+  assert(length >= 0);
+  return transport_->SendRTP(packet, static_cast<size_t>(length)) ? 0 : -1;
+}
+
+int VideoSendStream::SendRTCPPacket(int /*channel*/, const void* packet,
+                                    int length) {
+  assert(length >= 0);
+  return transport_->SendRTCP(packet, static_cast<size_t>(length)) ? 0 : -1;
+}
+
+}  // namespace internal
+}  // namespace webrtc