blob: be3fa5c4d44b3441383eb22f005d5125bfff6342 [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
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
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000013#include <sstream>
henrik.lundin@webrtc.orgce21c822013-10-23 11:04:57 +000014#include <string>
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000015#include <vector>
16
17#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
pbos@webrtc.org15536882014-06-06 10:49:19 +000018#include "webrtc/system_wrappers/interface/logging.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000019#include "webrtc/video_engine/include/vie_base.h"
20#include "webrtc/video_engine/include/vie_capture.h"
21#include "webrtc/video_engine/include/vie_codec.h"
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +000022#include "webrtc/video_engine/include/vie_external_codec.h"
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +000023#include "webrtc/video_engine/include/vie_image_process.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000024#include "webrtc/video_engine/include/vie_network.h"
25#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
pbos@webrtc.orgf39df522014-03-19 08:43:57 +000026#include "webrtc/video_engine/vie_defines.h"
pbos@webrtc.orgb581c902013-10-28 16:32:01 +000027#include "webrtc/video_send_stream.h"
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +000028
29namespace webrtc {
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000030std::string
pbos@webrtc.orgc23ed472014-05-15 10:03:24 +000031VideoSendStream::Config::EncoderSettings::ToString() const {
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000032 std::stringstream ss;
33 ss << "{payload_name: " << payload_name;
34 ss << ", payload_type: " << payload_type;
35 if (encoder != NULL)
pbos@webrtc.org15536882014-06-06 10:49:19 +000036 ss << ", encoder: " << (encoder != NULL ? "(encoder)" : "NULL");
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000037 ss << '}';
38 return ss.str();
39}
40
pbos@webrtc.orgc23ed472014-05-15 10:03:24 +000041std::string VideoSendStream::Config::Rtp::Rtx::ToString()
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000042 const {
43 std::stringstream ss;
44 ss << "{ssrcs: {";
45 for (size_t i = 0; i < ssrcs.size(); ++i) {
46 ss << ssrcs[i];
47 if (i != ssrcs.size() - 1)
48 ss << "}, {";
49 }
50 ss << '}';
51
52 ss << ", payload_type: " << payload_type;
53 ss << '}';
54 return ss.str();
55}
56
pbos@webrtc.orgc23ed472014-05-15 10:03:24 +000057std::string VideoSendStream::Config::Rtp::ToString() const {
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000058 std::stringstream ss;
59 ss << "{ssrcs: {";
60 for (size_t i = 0; i < ssrcs.size(); ++i) {
61 ss << ssrcs[i];
62 if (i != ssrcs.size() - 1)
63 ss << "}, {";
64 }
65 ss << '}';
66
67 ss << ", max_packet_size: " << max_packet_size;
68 if (min_transmit_bitrate_bps != 0)
69 ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
70
71 ss << ", extensions: {";
72 for (size_t i = 0; i < extensions.size(); ++i) {
73 ss << extensions[i].ToString();
74 if (i != extensions.size() - 1)
75 ss << "}, {";
76 }
77 ss << '}';
78
79 if (nack.rtp_history_ms != 0)
80 ss << ", nack.rtp_history_ms: " << nack.rtp_history_ms;
81 if (fec.ulpfec_payload_type != -1 || fec.red_payload_type != -1)
82 ss << ", fec: " << fec.ToString();
83 if (rtx.payload_type != 0 || !rtx.ssrcs.empty())
84 ss << ", rtx: " << rtx.ToString();
85 if (c_name != "")
86 ss << ", c_name: " << c_name;
87 ss << '}';
88 return ss.str();
89}
90
pbos@webrtc.orgc23ed472014-05-15 10:03:24 +000091std::string VideoSendStream::Config::ToString() const {
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +000092 std::stringstream ss;
93 ss << "{encoder_settings: " << encoder_settings.ToString();
94 ss << ", rtp: " << rtp.ToString();
95 if (pre_encode_callback != NULL)
96 ss << ", (pre_encode_callback)";
97 if (post_encode_callback != NULL)
98 ss << ", (post_encode_callback)";
99 if (local_renderer != NULL) {
100 ss << ", (local_renderer, render_delay_ms: " << render_delay_ms << ")";
101 }
102 if (target_delay_ms > 0)
103 ss << ", target_delay_ms: " << target_delay_ms;
pbos@webrtc.orgb4bc1a62014-05-15 09:35:06 +0000104 if (suspend_below_min_bitrate)
105 ss << ", suspend_below_min_bitrate: on";
106 ss << '}';
107 return ss.str();
108}
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000109
pbos@webrtc.orgc23ed472014-05-15 10:03:24 +0000110namespace internal {
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000111VideoSendStream::VideoSendStream(
112 newapi::Transport* transport,
113 CpuOveruseObserver* overuse_observer,
114 webrtc::VideoEngine* video_engine,
115 const VideoSendStream::Config& config,
116 const std::vector<VideoStream> video_streams,
117 const void* encoder_settings,
118 const std::map<uint32_t, RtpState>& suspended_ssrcs,
119 int base_channel,
120 int start_bitrate_bps)
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000121 : transport_adapter_(transport),
sprang@webrtc.org4a9843f2013-11-26 11:41:59 +0000122 encoded_frame_proxy_(config.post_encode_callback),
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000123 config_(config),
mflodman@webrtc.org212705c2014-06-16 08:57:39 +0000124 start_bitrate_bps_(start_bitrate_bps),
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000125 suspended_ssrcs_(suspended_ssrcs),
mflodman@webrtc.orge4d538a2013-12-13 09:40:45 +0000126 external_codec_(NULL),
pbos@webrtc.orgc54ff692014-04-28 13:00:21 +0000127 channel_(-1),
128 stats_proxy_(new SendStatisticsProxy(config, this)) {
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000129 video_engine_base_ = ViEBase::GetInterface(video_engine);
mflodman@webrtc.orge4d538a2013-12-13 09:40:45 +0000130 video_engine_base_->CreateChannel(channel_, base_channel);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000131 assert(channel_ != -1);
mflodman@webrtc.org212705c2014-06-16 08:57:39 +0000132 assert(start_bitrate_bps_ > 0);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000133
134 rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
135 assert(rtp_rtcp_ != NULL);
136
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000137 assert(config_.rtp.ssrcs.size() > 0);
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000138
pbos@webrtc.orgbef6e622014-03-19 10:59:52 +0000139 assert(config_.rtp.min_transmit_bitrate_bps >= 0);
pbos@webrtc.org9420a1f2014-03-13 12:52:27 +0000140 rtp_rtcp_->SetMinTransmitBitrate(channel_,
pbos@webrtc.orgbef6e622014-03-19 10:59:52 +0000141 config_.rtp.min_transmit_bitrate_bps / 1000);
pbos@webrtc.org9420a1f2014-03-13 12:52:27 +0000142
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000143 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
144 const std::string& extension = config_.rtp.extensions[i].name;
145 int id = config_.rtp.extensions[i].id;
pbos@webrtc.org60108c22013-11-20 11:48:56 +0000146 if (extension == RtpExtension::kTOffset) {
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000147 if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
148 abort();
pbos@webrtc.org60108c22013-11-20 11:48:56 +0000149 } else if (extension == RtpExtension::kAbsSendTime) {
pbos@webrtc.orge22b7612013-09-11 19:00:39 +0000150 if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
151 abort();
pbos@webrtc.org905cebd2013-09-11 10:14:56 +0000152 } else {
153 abort(); // Unsupported extension.
154 }
155 }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000156
mflodman@webrtc.orgab6ccbc2013-12-13 16:36:28 +0000157 rtp_rtcp_->SetRembStatus(channel_, true, false);
158
pbos@webrtc.orgaa693dd2013-09-20 11:56:26 +0000159 // Enable NACK, FEC or both.
160 if (config_.rtp.fec.red_payload_type != -1) {
161 assert(config_.rtp.fec.ulpfec_payload_type != -1);
162 if (config_.rtp.nack.rtp_history_ms > 0) {
163 rtp_rtcp_->SetHybridNACKFECStatus(
164 channel_,
165 true,
166 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
167 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
168 } else {
169 rtp_rtcp_->SetFECStatus(
170 channel_,
171 true,
172 static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
173 static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
174 }
175 } else {
176 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
177 }
178
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000179 ConfigureSsrcs();
180
pbos@webrtc.orgdebc6722013-08-22 09:42:17 +0000181 char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
182 assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
183 strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
184 rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
185
186 rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname);
187
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000188 capture_ = ViECapture::GetInterface(video_engine);
189 capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
190 capture_->ConnectCaptureDevice(capture_id_, channel_);
191
192 network_ = ViENetwork::GetInterface(video_engine);
193 assert(network_ != NULL);
194
pbos@webrtc.org26d75f32013-09-18 11:52:42 +0000195 network_->RegisterSendTransport(channel_, transport_adapter_);
sprang@webrtc.org6133dd52013-10-16 13:29:14 +0000196 // 28 to match packet overhead in ModuleRtpRtcpImpl.
pbos@webrtc.orgb581c902013-10-28 16:32:01 +0000197 network_->SetMTU(channel_,
198 static_cast<unsigned int>(config_.rtp.max_packet_size + 28));
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000199
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000200 assert(config.encoder_settings.encoder != NULL);
201 assert(config.encoder_settings.payload_type >= 0);
202 assert(config.encoder_settings.payload_type <= 127);
203 external_codec_ = ViEExternalCodec::GetInterface(video_engine);
204 if (external_codec_->RegisterExternalSendCodec(
205 channel_,
206 config.encoder_settings.payload_type,
207 config.encoder_settings.encoder,
208 false) != 0) {
209 abort();
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000210 }
211
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000212 codec_ = ViECodec::GetInterface(video_engine);
pbos@webrtc.org15536882014-06-06 10:49:19 +0000213 if (!ReconfigureVideoEncoder(video_streams, encoder_settings))
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000214 abort();
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000215
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000216 if (overuse_observer)
217 video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
218
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000219 image_process_ = ViEImageProcess::GetInterface(video_engine);
220 image_process_->RegisterPreEncodeCallback(channel_,
221 config_.pre_encode_callback);
sprang@webrtc.org4a9843f2013-11-26 11:41:59 +0000222 if (config_.post_encode_callback) {
223 image_process_->RegisterPostEncodeImageCallback(channel_,
224 &encoded_frame_proxy_);
225 }
henrik.lundin@webrtc.orgce21c822013-10-23 11:04:57 +0000226
pbos@webrtc.org15536882014-06-06 10:49:19 +0000227 if (config_.suspend_below_min_bitrate)
henrik.lundin@webrtc.org8fdf1912013-11-18 12:18:43 +0000228 codec_->SuspendBelowMinBitrate(channel_);
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000229
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000230 rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
231 stats_proxy_.get());
232 rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
233 stats_proxy_.get());
234 rtp_rtcp_->RegisterSendBitrateObserver(channel_, stats_proxy_.get());
235 rtp_rtcp_->RegisterSendFrameCountObserver(channel_, stats_proxy_.get());
236
237 codec_->RegisterEncoderObserver(channel_, *stats_proxy_);
238 capture_->RegisterObserver(capture_id_, *stats_proxy_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000239}
240
241VideoSendStream::~VideoSendStream() {
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000242 capture_->DeregisterObserver(capture_id_);
243 codec_->DeregisterEncoderObserver(channel_);
244
245 rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, stats_proxy_.get());
246 rtp_rtcp_->DeregisterSendBitrateObserver(channel_, stats_proxy_.get());
247 rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
248 stats_proxy_.get());
249 rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
250 stats_proxy_.get());
251
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000252 image_process_->DeRegisterPreEncodeCallback(channel_);
253
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000254 network_->DeregisterSendTransport(channel_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000255
256 capture_->DisconnectCaptureDevice(channel_);
257 capture_->ReleaseCaptureDevice(capture_id_);
258
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000259 external_codec_->DeRegisterExternalSendCodec(
260 channel_, config_.encoder_settings.payload_type);
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000261
pbos@webrtc.org3ba57eb2013-10-21 10:34:43 +0000262 video_engine_base_->DeleteChannel(channel_);
263
264 image_process_->Release();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000265 video_engine_base_->Release();
266 capture_->Release();
267 codec_->Release();
stefan@webrtc.orga0a91d82013-08-22 09:29:56 +0000268 if (external_codec_)
269 external_codec_->Release();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000270 network_->Release();
271 rtp_rtcp_->Release();
272}
273
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000274void VideoSendStream::SwapFrame(I420VideoFrame* frame) {
pbos@webrtc.org7123a802013-12-11 16:26:16 +0000275 // TODO(pbos): Local rendering should not be done on the capture thread.
276 if (config_.local_renderer != NULL)
pbos@webrtc.org18781922014-05-23 13:03:45 +0000277 config_.local_renderer->RenderFrame(*frame, 0);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000278
pbos@webrtc.org18781922014-05-23 13:03:45 +0000279 external_capture_->SwapFrame(frame);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000280}
281
pbos@webrtc.orgd8e92c92013-08-23 09:19:30 +0000282VideoSendStreamInput* VideoSendStream::Input() { return this; }
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000283
pbos@webrtc.org9d0f79f2014-04-24 11:13:21 +0000284void VideoSendStream::Start() {
sprang@webrtc.org48ac0da2014-01-27 13:03:02 +0000285 transport_adapter_.Enable();
pbos@webrtc.orgdf9f0992014-01-10 18:47:32 +0000286 video_engine_base_->StartSend(channel_);
287 video_engine_base_->StartReceive(channel_);
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000288}
289
pbos@webrtc.org9d0f79f2014-04-24 11:13:21 +0000290void VideoSendStream::Stop() {
pbos@webrtc.orgdf9f0992014-01-10 18:47:32 +0000291 video_engine_base_->StopSend(channel_);
292 video_engine_base_->StopReceive(channel_);
sprang@webrtc.org48ac0da2014-01-27 13:03:02 +0000293 transport_adapter_.Disable();
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000294}
295
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000296bool VideoSendStream::ReconfigureVideoEncoder(
297 const std::vector<VideoStream>& streams,
pbos@webrtc.org15536882014-06-06 10:49:19 +0000298 const void* encoder_settings) {
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000299 assert(!streams.empty());
300 assert(config_.rtp.ssrcs.size() >= streams.size());
pbos@webrtc.org8f2997c2013-11-14 08:58:14 +0000301
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000302 VideoCodec video_codec;
303 memset(&video_codec, 0, sizeof(video_codec));
304 video_codec.codecType =
305 (config_.encoder_settings.payload_name == "VP8" ? kVideoCodecVP8
306 : kVideoCodecGeneric);
307
308 if (video_codec.codecType == kVideoCodecVP8) {
309 video_codec.codecSpecific.VP8.resilience = kResilientStream;
310 video_codec.codecSpecific.VP8.numberOfTemporalLayers = 1;
311 video_codec.codecSpecific.VP8.denoisingOn = true;
312 video_codec.codecSpecific.VP8.errorConcealmentOn = false;
313 video_codec.codecSpecific.VP8.automaticResizeOn = false;
314 video_codec.codecSpecific.VP8.frameDroppingOn = true;
315 video_codec.codecSpecific.VP8.keyFrameInterval = 3000;
316 }
317
pbos@webrtc.orga7651f82014-07-10 10:13:37 +0000318 if (video_codec.codecType == kVideoCodecVP8) {
319 if (encoder_settings != NULL) {
320 video_codec.codecSpecific.VP8 =
321 *reinterpret_cast<const VideoCodecVP8*>(encoder_settings);
322 }
323 } else {
324 // TODO(pbos): Support encoder_settings codec-agnostically.
325 assert(encoder_settings == NULL);
326 }
327
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000328 strncpy(video_codec.plName,
329 config_.encoder_settings.payload_name.c_str(),
330 kPayloadNameSize - 1);
331 video_codec.plName[kPayloadNameSize - 1] = '\0';
332 video_codec.plType = config_.encoder_settings.payload_type;
333 video_codec.numberOfSimulcastStreams =
334 static_cast<unsigned char>(streams.size());
335 video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
336 assert(streams.size() <= kMaxSimulcastStreams);
337 for (size_t i = 0; i < streams.size(); ++i) {
338 SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
339 assert(streams[i].width > 0);
340 assert(streams[i].height > 0);
341 assert(streams[i].max_framerate > 0);
342 // Different framerates not supported per stream at the moment.
343 assert(streams[i].max_framerate == streams[0].max_framerate);
344 assert(streams[i].min_bitrate_bps >= 0);
345 assert(streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps);
346 assert(streams[i].max_bitrate_bps >= streams[i].target_bitrate_bps);
347 assert(streams[i].max_qp >= 0);
348
349 sim_stream->width = static_cast<unsigned short>(streams[i].width);
350 sim_stream->height = static_cast<unsigned short>(streams[i].height);
351 sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
352 sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
353 sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
354 sim_stream->qpMax = streams[i].max_qp;
355 // TODO(pbos): Implement mapping for temporal layers.
356 assert(streams[i].temporal_layers.empty());
357
358 video_codec.width = std::max(video_codec.width,
359 static_cast<unsigned short>(streams[i].width));
360 video_codec.height = std::max(
361 video_codec.height, static_cast<unsigned short>(streams[i].height));
362 video_codec.minBitrate =
363 std::min(video_codec.minBitrate,
364 static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
365 video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
366 video_codec.qpMax = std::max(video_codec.qpMax,
367 static_cast<unsigned int>(streams[i].max_qp));
368 }
mflodman@webrtc.org212705c2014-06-16 08:57:39 +0000369 video_codec.startBitrate =
370 static_cast<unsigned int>(start_bitrate_bps_) / 1000;
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000371
372 if (video_codec.minBitrate < kViEMinCodecBitrate)
373 video_codec.minBitrate = kViEMinCodecBitrate;
374 if (video_codec.maxBitrate < kViEMinCodecBitrate)
375 video_codec.maxBitrate = kViEMinCodecBitrate;
mflodman@webrtc.org212705c2014-06-16 08:57:39 +0000376 if (video_codec.startBitrate < video_codec.minBitrate)
377 video_codec.startBitrate = video_codec.minBitrate;
378 if (video_codec.startBitrate > video_codec.maxBitrate)
379 video_codec.startBitrate = video_codec.maxBitrate;
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000380
381 if (video_codec.startBitrate < video_codec.minBitrate)
382 video_codec.startBitrate = video_codec.minBitrate;
383 if (video_codec.startBitrate > video_codec.maxBitrate)
384 video_codec.startBitrate = video_codec.maxBitrate;
385
pbos@webrtc.org15536882014-06-06 10:49:19 +0000386 assert(streams[0].max_framerate > 0);
387 video_codec.maxFramerate = streams[0].max_framerate;
pbos@webrtc.orgf39df522014-03-19 08:43:57 +0000388
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000389 return codec_->SetSendCodec(channel_, video_codec) == 0;
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000390}
391
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000392bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
393 return network_->ReceivedRTCPPacket(
pbos@webrtc.org30c741a2013-08-05 13:25:51 +0000394 channel_, packet, static_cast<int>(length)) == 0;
pbos@webrtc.orgbf9bc322013-08-05 12:01:36 +0000395}
sprang@webrtc.orgca723002014-01-07 09:54:34 +0000396
397VideoSendStream::Stats VideoSendStream::GetStats() const {
398 return stats_proxy_->GetStats();
399}
400
401bool VideoSendStream::GetSendSideDelay(VideoSendStream::Stats* stats) {
402 return codec_->GetSendSideDelay(
403 channel_, &stats->avg_delay_ms, &stats->max_delay_ms);
404}
405
406std::string VideoSendStream::GetCName() {
407 char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
408 rtp_rtcp_->GetRTCPCName(channel_, rtcp_cname);
409 return rtcp_cname;
410}
411
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000412void VideoSendStream::ConfigureSsrcs() {
413 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
414 uint32_t ssrc = config_.rtp.ssrcs[i];
415 rtp_rtcp_->SetLocalSSRC(
416 channel_, ssrc, kViEStreamTypeNormal, static_cast<unsigned char>(i));
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000417 RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
418 if (it != suspended_ssrcs_.end())
419 rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000420 }
421
422 if (config_.rtp.rtx.ssrcs.empty()) {
423 assert(!config_.rtp.rtx.pad_with_redundant_payloads);
424 return;
425 }
426
427 // Set up RTX.
428 assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000429 for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
430 uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000431 rtp_rtcp_->SetLocalSSRC(channel_,
432 config_.rtp.rtx.ssrcs[i],
433 kViEStreamTypeRtx,
434 static_cast<unsigned char>(i));
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000435 RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
436 if (it != suspended_ssrcs_.end())
437 rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
pbos@webrtc.orgb96302f2014-06-30 13:19:09 +0000438 }
439
440 if (config_.rtp.rtx.pad_with_redundant_payloads) {
441 rtp_rtcp_->SetPadWithRedundantPayloads(channel_, true);
442 }
443
444 assert(config_.rtp.rtx.payload_type >= 0);
445 rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
446}
447
pbos@webrtc.org49e63062014-07-07 13:06:48 +0000448std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
449 std::map<uint32_t, RtpState> rtp_states;
450 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
451 uint32_t ssrc = config_.rtp.ssrcs[i];
452 rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
453 }
454
455 for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
456 uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
457 rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
458 }
459
460 return rtp_states;
461}
462
pbos@webrtc.orgdc8c8832013-05-16 12:08:03 +0000463} // namespace internal
464} // namespace webrtc