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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
pbos@webrtc.orgb581c902013-10-28 16:32:01 +000011#include "webrtc/video/video_send_stream.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000012
henrik.lundin@webrtc.orgce21c822013-10-23 11:04:57 +000013#include <string>
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000014#include <vector>
15
16#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
17#include "webrtc/video_engine/include/vie_base.h"
18#include "webrtc/video_engine/include/vie_capture.h"
19#include "webrtc/video_engine/include/vie_codec.h"
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +000020#include "webrtc/video_engine/include/vie_external_codec.h"
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +000021#include "webrtc/video_engine/include/vie_image_process.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"
pbos@webrtc.orgf39df522014-03-19 08:43:57 +000024#include "webrtc/video_engine/vie_defines.h"
pbos@webrtc.orgb581c902013-10-28 16:32:01 +000025#include "webrtc/video_send_stream.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000026
27namespace webrtc {
28namespace internal {
29
pbos@webrtc.org12d5ede2013-07-09 08:02:33 +000030VideoSendStream::VideoSendStream(newapi::Transport* transport,
asapersson@webrtc.org4b1817f2014-01-31 10:05:07 +000031 CpuOveruseObserver* overuse_observer,
pbos@webrtc.org12d5ede2013-07-09 08:02:33 +000032 webrtc::VideoEngine* video_engine,
mflodman@webrtc.orge4d538a2013-12-13 09:40:45 +000033 const VideoSendStream::Config& config,
34 int base_channel)
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +000035 : transport_adapter_(transport),
sprang@webrtc.org4a9843f2013-11-26 11:41:59 +000036 encoded_frame_proxy_(config.post_encode_callback),
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +000037 codec_lock_(CriticalSectionWrapper::CreateCriticalSection()),
38 config_(config),
mflodman@webrtc.orge4d538a2013-12-13 09:40:45 +000039 external_codec_(NULL),
pbos@webrtc.orgc54ff692014-04-28 13:00:21 +000040 channel_(-1),
41 stats_proxy_(new SendStatisticsProxy(config, this)) {
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000042 video_engine_base_ = ViEBase::GetInterface(video_engine);
mflodman@webrtc.orge4d538a2013-12-13 09:40:45 +000043 video_engine_base_->CreateChannel(channel_, base_channel);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000044 assert(channel_ != -1);
45
46 rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
47 assert(rtp_rtcp_ != NULL);
48
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +000049 assert(config_.rtp.ssrcs.size() > 0);
henrik.lundin@webrtc.orgd7d60c82013-11-21 14:05:40 +000050 if (config_.suspend_below_min_bitrate)
51 config_.pacing = true;
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +000052 rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
pbos@webrtc.org905cebd2013-09-11 10:14:56 +000053
pbos@webrtc.orgbef6e622014-03-19 10:59:52 +000054 assert(config_.rtp.min_transmit_bitrate_bps >= 0);
pbos@webrtc.org9420a1f2014-03-13 12:52:27 +000055 rtp_rtcp_->SetMinTransmitBitrate(channel_,
pbos@webrtc.orgbef6e622014-03-19 10:59:52 +000056 config_.rtp.min_transmit_bitrate_bps / 1000);
pbos@webrtc.org9420a1f2014-03-13 12:52:27 +000057
pbos@webrtc.org905cebd2013-09-11 10:14:56 +000058 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
59 const std::string& extension = config_.rtp.extensions[i].name;
60 int id = config_.rtp.extensions[i].id;
pbos@webrtc.org60108c22013-11-20 11:48:56 +000061 if (extension == RtpExtension::kTOffset) {
pbos@webrtc.org905cebd2013-09-11 10:14:56 +000062 if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
63 abort();
pbos@webrtc.org60108c22013-11-20 11:48:56 +000064 } else if (extension == RtpExtension::kAbsSendTime) {
pbos@webrtc.orge22b7612013-09-11 19:00:39 +000065 if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
66 abort();
pbos@webrtc.org905cebd2013-09-11 10:14:56 +000067 } else {
68 abort(); // Unsupported extension.
69 }
70 }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000071
mflodman@webrtc.orgab6ccbc2013-12-13 16:36:28 +000072 rtp_rtcp_->SetRembStatus(channel_, true, false);
73
pbos@webrtc.orgaa693dd2013-09-20 11:56:26 +000074 // Enable NACK, FEC or both.
75 if (config_.rtp.fec.red_payload_type != -1) {
76 assert(config_.rtp.fec.ulpfec_payload_type != -1);
77 if (config_.rtp.nack.rtp_history_ms > 0) {
78 rtp_rtcp_->SetHybridNACKFECStatus(
79 channel_,
80 true,
81 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
82 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
83 } else {
84 rtp_rtcp_->SetFECStatus(
85 channel_,
86 true,
87 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
88 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
89 }
90 } else {
91 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
92 }
93
pbos@webrtc.orgdebc6722013-08-22 09:42:17 +000094 char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
95 assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
96 strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
97 rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
98
99 rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname);
100
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000101 capture_ = ViECapture::GetInterface(video_engine);
102 capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
103 capture_->ConnectCaptureDevice(capture_id_, channel_);
104
105 network_ = ViENetwork::GetInterface(video_engine);
106 assert(network_ != NULL);
107
pbos@webrtc.org26d75f32013-09-18 11:52:42 +0000108 network_->RegisterSendTransport(channel_, transport_adapter_);
sprang@webrtc.org6133dd52013-10-16 13:29:14 +0000109 // 28 to match packet overhead in ModuleRtpRtcpImpl.
pbos@webrtc.orgb581c902013-10-28 16:32:01 +0000110 network_->SetMTU(channel_,
111 static_cast<unsigned int>(config_.rtp.max_packet_size + 28));
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000112
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000113 assert(config.encoder_settings.encoder != NULL);
114 assert(config.encoder_settings.payload_type >= 0);
115 assert(config.encoder_settings.payload_type <= 127);
116 external_codec_ = ViEExternalCodec::GetInterface(video_engine);
117 if (external_codec_->RegisterExternalSendCodec(
118 channel_,
119 config.encoder_settings.payload_type,
120 config.encoder_settings.encoder,
121 false) != 0) {
122 abort();
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000123 }
124
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000125 codec_ = ViECodec::GetInterface(video_engine);
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000126 if (!ReconfigureVideoEncoder(config_.encoder_settings.streams,
127 config_.encoder_settings.encoder_settings)) {
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000128 abort();
mflodman@webrtc.orgecbeb2b2013-07-23 11:35:00 +0000129 }
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000130
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000131 if (overuse_observer)
132 video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
133
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000134 image_process_ = ViEImageProcess::GetInterface(video_engine);
135 image_process_->RegisterPreEncodeCallback(channel_,
136 config_.pre_encode_callback);
sprang@webrtc.org4a9843f2013-11-26 11:41:59 +0000137 if (config_.post_encode_callback) {
138 image_process_->RegisterPostEncodeImageCallback(channel_,
139 &encoded_frame_proxy_);
140 }
henrik.lundin@webrtc.orgce21c822013-10-23 11:04:57 +0000141
henrik.lundin@webrtc.org0bf5a2f2014-03-06 07:19:28 +0000142 if (config_.suspend_below_min_bitrate) {
henrik.lundin@webrtc.org8fdf1912013-11-18 12:18:43 +0000143 codec_->SuspendBelowMinBitrate(channel_);
henrik.lundin@webrtc.orgce21c822013-10-23 11:04:57 +0000144 }
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000145
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000146 rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
147 stats_proxy_.get());
148 rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
149 stats_proxy_.get());
150 rtp_rtcp_->RegisterSendBitrateObserver(channel_, stats_proxy_.get());
151 rtp_rtcp_->RegisterSendFrameCountObserver(channel_, stats_proxy_.get());
152
153 codec_->RegisterEncoderObserver(channel_, *stats_proxy_);
154 capture_->RegisterObserver(capture_id_, *stats_proxy_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000155}
156
157VideoSendStream::~VideoSendStream() {
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000158 capture_->DeregisterObserver(capture_id_);
159 codec_->DeregisterEncoderObserver(channel_);
160
161 rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, stats_proxy_.get());
162 rtp_rtcp_->DeregisterSendBitrateObserver(channel_, stats_proxy_.get());
163 rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
164 stats_proxy_.get());
165 rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
166 stats_proxy_.get());
167
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000168 image_process_->DeRegisterPreEncodeCallback(channel_);
169
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000170 network_->DeregisterSendTransport(channel_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000171
172 capture_->DisconnectCaptureDevice(channel_);
173 capture_->ReleaseCaptureDevice(capture_id_);
174
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000175 external_codec_->DeRegisterExternalSendCodec(
176 channel_, config_.encoder_settings.payload_type);
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000177
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000178 video_engine_base_->DeleteChannel(channel_);
179
180 image_process_->Release();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000181 video_engine_base_->Release();
182 capture_->Release();
183 codec_->Release();
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000184 if (external_codec_)
185 external_codec_->Release();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000186 network_->Release();
187 rtp_rtcp_->Release();
188}
189
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000190void VideoSendStream::PutFrame(const I420VideoFrame& frame) {
191 input_frame_.CopyFrame(frame);
192 SwapFrame(&input_frame_);
193}
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000194
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000195void VideoSendStream::SwapFrame(I420VideoFrame* frame) {
196 // TODO(pbos): Warn if frame is "too far" into the future, or too old. This
197 // would help detect if frame's being used without NTP.
198 // TO REVIEWER: Is there any good check for this? Should it be
199 // skipped?
200 if (frame != &input_frame_)
201 input_frame_.SwapFrame(frame);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000202
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000203 // TODO(pbos): Local rendering should not be done on the capture thread.
204 if (config_.local_renderer != NULL)
205 config_.local_renderer->RenderFrame(input_frame_, 0);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000206
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000207 external_capture_->SwapFrame(&input_frame_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000208}
209
pbos@webrtc.orgd8e92c92013-08-23 09:19:30 +0000210VideoSendStreamInput* VideoSendStream::Input() { return this; }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000211
pbos@webrtc.org9d0f79f2014-04-24 11:13:21 +0000212void VideoSendStream::Start() {
sprang@webrtc.org48ac0da2014-01-27 13:03:02 +0000213 transport_adapter_.Enable();
pbos@webrtc.orgdf9f0992014-01-10 18:47:32 +0000214 video_engine_base_->StartSend(channel_);
215 video_engine_base_->StartReceive(channel_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000216}
217
pbos@webrtc.org9d0f79f2014-04-24 11:13:21 +0000218void VideoSendStream::Stop() {
pbos@webrtc.orgdf9f0992014-01-10 18:47:32 +0000219 video_engine_base_->StopSend(channel_);
220 video_engine_base_->StopReceive(channel_);
sprang@webrtc.org48ac0da2014-01-27 13:03:02 +0000221 transport_adapter_.Disable();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000222}
223
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000224bool VideoSendStream::ReconfigureVideoEncoder(
225 const std::vector<VideoStream>& streams,
226 void* encoder_settings) {
227 assert(!streams.empty());
228 assert(config_.rtp.ssrcs.size() >= streams.size());
229 // TODO(pbos): Wire encoder_settings.
230 assert(encoder_settings == NULL);
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000231
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000232 // VideoStreams in config_.encoder_settings need to be locked.
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000233 CriticalSectionScoped crit(codec_lock_.get());
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000234
235 VideoCodec video_codec;
236 memset(&video_codec, 0, sizeof(video_codec));
237 video_codec.codecType =
238 (config_.encoder_settings.payload_name == "VP8" ? kVideoCodecVP8
239 : kVideoCodecGeneric);
240
241 if (video_codec.codecType == kVideoCodecVP8) {
242 video_codec.codecSpecific.VP8.resilience = kResilientStream;
243 video_codec.codecSpecific.VP8.numberOfTemporalLayers = 1;
244 video_codec.codecSpecific.VP8.denoisingOn = true;
245 video_codec.codecSpecific.VP8.errorConcealmentOn = false;
246 video_codec.codecSpecific.VP8.automaticResizeOn = false;
247 video_codec.codecSpecific.VP8.frameDroppingOn = true;
248 video_codec.codecSpecific.VP8.keyFrameInterval = 3000;
249 }
250
251 strncpy(video_codec.plName,
252 config_.encoder_settings.payload_name.c_str(),
253 kPayloadNameSize - 1);
254 video_codec.plName[kPayloadNameSize - 1] = '\0';
255 video_codec.plType = config_.encoder_settings.payload_type;
256 video_codec.numberOfSimulcastStreams =
257 static_cast<unsigned char>(streams.size());
258 video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
259 assert(streams.size() <= kMaxSimulcastStreams);
260 for (size_t i = 0; i < streams.size(); ++i) {
261 SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
262 assert(streams[i].width > 0);
263 assert(streams[i].height > 0);
264 assert(streams[i].max_framerate > 0);
265 // Different framerates not supported per stream at the moment.
266 assert(streams[i].max_framerate == streams[0].max_framerate);
267 assert(streams[i].min_bitrate_bps >= 0);
268 assert(streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps);
269 assert(streams[i].max_bitrate_bps >= streams[i].target_bitrate_bps);
270 assert(streams[i].max_qp >= 0);
271
272 sim_stream->width = static_cast<unsigned short>(streams[i].width);
273 sim_stream->height = static_cast<unsigned short>(streams[i].height);
274 sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
275 sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
276 sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
277 sim_stream->qpMax = streams[i].max_qp;
278 // TODO(pbos): Implement mapping for temporal layers.
279 assert(streams[i].temporal_layers.empty());
280
281 video_codec.width = std::max(video_codec.width,
282 static_cast<unsigned short>(streams[i].width));
283 video_codec.height = std::max(
284 video_codec.height, static_cast<unsigned short>(streams[i].height));
285 video_codec.minBitrate =
286 std::min(video_codec.minBitrate,
287 static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
288 video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
289 video_codec.qpMax = std::max(video_codec.qpMax,
290 static_cast<unsigned int>(streams[i].max_qp));
291 }
292
293 if (video_codec.minBitrate < kViEMinCodecBitrate)
294 video_codec.minBitrate = kViEMinCodecBitrate;
295 if (video_codec.maxBitrate < kViEMinCodecBitrate)
296 video_codec.maxBitrate = kViEMinCodecBitrate;
297
298 video_codec.startBitrate = 300;
299
300 if (video_codec.startBitrate < video_codec.minBitrate)
301 video_codec.startBitrate = video_codec.minBitrate;
302 if (video_codec.startBitrate > video_codec.maxBitrate)
303 video_codec.startBitrate = video_codec.maxBitrate;
304
305 assert(config_.encoder_settings.streams[0].max_framerate > 0);
306 video_codec.maxFramerate = config_.encoder_settings.streams[0].max_framerate;
307
308 if (codec_->SetSendCodec(channel_, video_codec) != 0)
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000309 return false;
310
pbos@webrtc.org9105cbd2013-11-28 11:59:31 +0000311 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
312 rtp_rtcp_->SetLocalSSRC(channel_,
313 config_.rtp.ssrcs[i],
314 kViEStreamTypeNormal,
315 static_cast<unsigned char>(i));
316 }
317
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000318 config_.encoder_settings.streams = streams;
319 config_.encoder_settings.encoder_settings = encoder_settings;
pbos@webrtc.orgc7667752014-01-24 09:30:53 +0000320
pbos@webrtc.org9105cbd2013-11-28 11:59:31 +0000321 if (config_.rtp.rtx.ssrcs.empty())
322 return true;
323
324 // Set up RTX.
325 assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
326 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
327 rtp_rtcp_->SetLocalSSRC(channel_,
328 config_.rtp.rtx.ssrcs[i],
329 kViEStreamTypeRtx,
330 static_cast<unsigned char>(i));
331 }
332
pbos@webrtc.orgc7667752014-01-24 09:30:53 +0000333 if (config_.rtp.rtx.payload_type != 0)
334 rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
pbos@webrtc.org9105cbd2013-11-28 11:59:31 +0000335
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000336 return true;
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000337}
338
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000339bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
340 return network_->ReceivedRTCPPacket(
pbos@webrtc.org30c741a2013-08-05 13:25:51 +0000341 channel_, packet, static_cast<int>(length)) == 0;
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000342}
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000343
344VideoSendStream::Stats VideoSendStream::GetStats() const {
345 return stats_proxy_->GetStats();
346}
347
348bool VideoSendStream::GetSendSideDelay(VideoSendStream::Stats* stats) {
349 return codec_->GetSendSideDelay(
350 channel_, &stats->avg_delay_ms, &stats->max_delay_ms);
351}
352
353std::string VideoSendStream::GetCName() {
354 char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
355 rtp_rtcp_->GetRTCPCName(channel_, rtcp_cname);
356 return rtcp_cname;
357}
358
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000359} // namespace internal
360} // namespace webrtc