blob: 55921c2ca7455882d4d205405eab85c879d9cd09 [file] [log] [blame]
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +00001/*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/video_engine/internal/video_send_stream.h"
12
pbos@webrtc.orgdebc6722013-08-22 09:42:17 +000013#include <string.h>
14
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000015#include <vector>
16
17#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
18#include "webrtc/video_engine/include/vie_base.h"
19#include "webrtc/video_engine/include/vie_capture.h"
20#include "webrtc/video_engine/include/vie_codec.h"
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +000021#include "webrtc/video_engine/include/vie_external_codec.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000022#include "webrtc/video_engine/include/vie_network.h"
23#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
24#include "webrtc/video_engine/new_include/video_send_stream.h"
25
26namespace webrtc {
27namespace internal {
28
mflodman@webrtc.orgecbeb2b2013-07-23 11:35:00 +000029// Super simple and temporary overuse logic. This will move to the application
30// as soon as the new API allows changing send codec on the fly.
31class ResolutionAdaptor : public webrtc::CpuOveruseObserver {
32 public:
33 ResolutionAdaptor(ViECodec* codec, int channel, size_t width, size_t height)
34 : codec_(codec),
35 channel_(channel),
36 max_width_(width),
37 max_height_(height) {}
38
39 virtual ~ResolutionAdaptor() {}
40
41 virtual void OveruseDetected() OVERRIDE {
42 VideoCodec codec;
43 if (codec_->GetSendCodec(channel_, codec) != 0)
44 return;
45
46 if (codec.width / 2 < min_width || codec.height / 2 < min_height)
47 return;
48
49 codec.width /= 2;
50 codec.height /= 2;
51 codec_->SetSendCodec(channel_, codec);
52 }
53
54 virtual void NormalUsage() OVERRIDE {
55 VideoCodec codec;
56 if (codec_->GetSendCodec(channel_, codec) != 0)
57 return;
58
59 if (codec.width * 2u > max_width_ || codec.height * 2u > max_height_)
60 return;
61
62 codec.width *= 2;
63 codec.height *= 2;
64 codec_->SetSendCodec(channel_, codec);
65 }
66
67 private:
68 // Temporary and arbitrary chosen minimum resolution.
69 static const size_t min_width = 160;
70 static const size_t min_height = 120;
71
72 ViECodec* codec_;
73 const int channel_;
74
75 const size_t max_width_;
76 const size_t max_height_;
77};
78
pbos@webrtc.org12d5ede2013-07-09 08:02:33 +000079VideoSendStream::VideoSendStream(newapi::Transport* transport,
mflodman@webrtc.orgecbeb2b2013-07-23 11:35:00 +000080 bool overuse_detection,
pbos@webrtc.org12d5ede2013-07-09 08:02:33 +000081 webrtc::VideoEngine* video_engine,
pbos@webrtc.orgd8e92c92013-08-23 09:19:30 +000082 const VideoSendStream::Config& config)
pbos@webrtc.org26d75f32013-09-18 11:52:42 +000083 : transport_adapter_(transport), config_(config), external_codec_(NULL) {
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000084
85 if (config_.codec.numberOfSimulcastStreams > 0) {
86 assert(config_.rtp.ssrcs.size() == config_.codec.numberOfSimulcastStreams);
87 } else {
88 assert(config_.rtp.ssrcs.size() == 1);
89 }
90
91 video_engine_base_ = ViEBase::GetInterface(video_engine);
92 video_engine_base_->CreateChannel(channel_);
93 assert(channel_ != -1);
94
95 rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
96 assert(rtp_rtcp_ != NULL);
97
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +000098 if (config_.rtp.ssrcs.size() == 1) {
99 rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs[0]);
100 } else {
101 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
102 rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs[i],
103 kViEStreamTypeNormal, i);
104 }
105 }
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000106 rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
pbos@webrtc.orgf952fce2013-09-16 13:01:47 +0000107 if (!config_.rtp.rtx.ssrcs.empty()) {
108 assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
109 for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
110 rtp_rtcp_->SetLocalSSRC(
111 channel_, config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx, i);
112 }
113
114 if (config_.rtp.rtx.rtx_payload_type != 0) {
115 rtp_rtcp_->SetRtxSendPayloadType(channel_,
116 config_.rtp.rtx.rtx_payload_type);
117 }
118 }
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000119
120 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
121 const std::string& extension = config_.rtp.extensions[i].name;
122 int id = config_.rtp.extensions[i].id;
123 if (extension == "toffset") {
124 if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
125 abort();
pbos@webrtc.orge22b7612013-09-11 19:00:39 +0000126 } else if (extension == "abs-send-time") {
127 if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
128 abort();
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000129 } else {
130 abort(); // Unsupported extension.
131 }
132 }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000133
pbos@webrtc.orgaa693dd2013-09-20 11:56:26 +0000134 // Enable NACK, FEC or both.
135 if (config_.rtp.fec.red_payload_type != -1) {
136 assert(config_.rtp.fec.ulpfec_payload_type != -1);
137 if (config_.rtp.nack.rtp_history_ms > 0) {
138 rtp_rtcp_->SetHybridNACKFECStatus(
139 channel_,
140 true,
141 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
142 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
143 } else {
144 rtp_rtcp_->SetFECStatus(
145 channel_,
146 true,
147 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
148 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
149 }
150 } else {
151 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
152 }
153
pbos@webrtc.orgdebc6722013-08-22 09:42:17 +0000154 char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
155 assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
156 strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
157 rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
158
159 rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname);
160
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000161 capture_ = ViECapture::GetInterface(video_engine);
162 capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
163 capture_->ConnectCaptureDevice(capture_id_, channel_);
164
165 network_ = ViENetwork::GetInterface(video_engine);
166 assert(network_ != NULL);
167
pbos@webrtc.org26d75f32013-09-18 11:52:42 +0000168 network_->RegisterSendTransport(channel_, transport_adapter_);
sprang@webrtc.org6133dd52013-10-16 13:29:14 +0000169 // 28 to match packet overhead in ModuleRtpRtcpImpl.
170 network_->SetMTU(channel_, config_.rtp.max_packet_size + 28);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000171
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000172 if (config.encoder) {
173 external_codec_ = ViEExternalCodec::GetInterface(video_engine);
174 if (external_codec_->RegisterExternalSendCodec(
175 channel_, config.codec.plType, config.encoder,
176 config.internal_source) != 0) {
177 abort();
178 }
179 }
180
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000181 codec_ = ViECodec::GetInterface(video_engine);
182 if (codec_->SetSendCodec(channel_, config_.codec) != 0) {
183 abort();
184 }
mflodman@webrtc.orgecbeb2b2013-07-23 11:35:00 +0000185
186 if (overuse_detection) {
187 overuse_observer_.reset(
188 new ResolutionAdaptor(codec_, channel_, config_.codec.width,
189 config_.codec.height));
190 video_engine_base_->RegisterCpuOveruseObserver(channel_,
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000191 overuse_observer_.get());
mflodman@webrtc.orgecbeb2b2013-07-23 11:35:00 +0000192 }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000193}
194
195VideoSendStream::~VideoSendStream() {
196 network_->DeregisterSendTransport(channel_);
197 video_engine_base_->DeleteChannel(channel_);
198
199 capture_->DisconnectCaptureDevice(channel_);
200 capture_->ReleaseCaptureDevice(capture_id_);
201
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000202 if (external_codec_) {
203 external_codec_->DeRegisterExternalSendCodec(channel_,
204 config_.codec.plType);
205 }
206
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000207 video_engine_base_->Release();
208 capture_->Release();
209 codec_->Release();
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000210 if (external_codec_)
211 external_codec_->Release();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000212 network_->Release();
213 rtp_rtcp_->Release();
214}
215
216void VideoSendStream::PutFrame(const I420VideoFrame& frame,
pbos@webrtc.orgd9f91852013-05-23 12:37:11 +0000217 uint32_t time_since_capture_ms) {
218 // TODO(pbos): frame_copy should happen after the VideoProcessingModule has
219 // resized the frame.
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000220 I420VideoFrame frame_copy;
221 frame_copy.CopyFrame(frame);
222
223 if (config_.pre_encode_callback != NULL) {
224 config_.pre_encode_callback->FrameCallback(&frame_copy);
225 }
226
227 ViEVideoFrameI420 vf;
228
229 // TODO(pbos): This represents a memcpy step and is only required because
230 // external_capture_ only takes ViEVideoFrameI420s.
231 vf.y_plane = frame_copy.buffer(kYPlane);
232 vf.u_plane = frame_copy.buffer(kUPlane);
233 vf.v_plane = frame_copy.buffer(kVPlane);
234 vf.y_pitch = frame.stride(kYPlane);
235 vf.u_pitch = frame.stride(kUPlane);
236 vf.v_pitch = frame.stride(kVPlane);
237 vf.width = frame.width();
238 vf.height = frame.height();
239
pbos@webrtc.orgd9f91852013-05-23 12:37:11 +0000240 external_capture_->IncomingFrameI420(vf, frame.render_time_ms());
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000241
242 if (config_.local_renderer != NULL) {
243 config_.local_renderer->RenderFrame(frame, 0);
244 }
245}
246
pbos@webrtc.orgd8e92c92013-08-23 09:19:30 +0000247VideoSendStreamInput* VideoSendStream::Input() { return this; }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000248
249void VideoSendStream::StartSend() {
pbos@webrtc.orgd9f91852013-05-23 12:37:11 +0000250 if (video_engine_base_->StartSend(channel_) != 0)
251 abort();
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000252 if (video_engine_base_->StartReceive(channel_) != 0)
253 abort();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000254}
255
256void VideoSendStream::StopSend() {
pbos@webrtc.orgd9f91852013-05-23 12:37:11 +0000257 if (video_engine_base_->StopSend(channel_) != 0)
258 abort();
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000259 if (video_engine_base_->StopReceive(channel_) != 0)
260 abort();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000261}
262
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000263bool VideoSendStream::SetTargetBitrate(
pbos@webrtc.orgd9f91852013-05-23 12:37:11 +0000264 int min_bitrate,
265 int max_bitrate,
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000266 const std::vector<SimulcastStream>& streams) {
267 return false;
268}
269
270void VideoSendStream::GetSendCodec(VideoCodec* send_codec) {
271 *send_codec = config_.codec;
272}
273
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000274bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
275 return network_->ReceivedRTCPPacket(
pbos@webrtc.org30c741a2013-08-05 13:25:51 +0000276 channel_, packet, static_cast<int>(length)) == 0;
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000277}
278
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000279} // namespace internal
280} // namespace webrtc