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pbos@webrtc.org29d58392013-05-16 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.org16e03b72013-10-28 16:32:01 +000011#include "webrtc/video/video_send_stream.h"
pbos@webrtc.org29d58392013-05-16 12:08:03 +000012
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +000013#include <algorithm>
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000014#include <sstream>
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +000015#include <string>
pbos@webrtc.org29d58392013-05-16 12:08:03 +000016#include <vector>
17
18#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
pbos@webrtc.org6ae48c62014-06-06 10:49:19 +000019#include "webrtc/system_wrappers/interface/logging.h"
pbos@webrtc.org29d58392013-05-16 12:08:03 +000020#include "webrtc/video_engine/include/vie_base.h"
21#include "webrtc/video_engine/include/vie_capture.h"
22#include "webrtc/video_engine/include/vie_codec.h"
stefan@webrtc.org360e3762013-08-22 09:29:56 +000023#include "webrtc/video_engine/include/vie_external_codec.h"
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +000024#include "webrtc/video_engine/include/vie_image_process.h"
pbos@webrtc.org29d58392013-05-16 12:08:03 +000025#include "webrtc/video_engine/include/vie_network.h"
26#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +000027#include "webrtc/video_engine/vie_defines.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000028#include "webrtc/video_send_stream.h"
pbos@webrtc.org29d58392013-05-16 12:08:03 +000029
30namespace webrtc {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000031std::string
pbos@webrtc.org024e4d52014-05-15 10:03:24 +000032VideoSendStream::Config::EncoderSettings::ToString() const {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000033 std::stringstream ss;
34 ss << "{payload_name: " << payload_name;
35 ss << ", payload_type: " << payload_type;
36 if (encoder != NULL)
pbos@webrtc.org6ae48c62014-06-06 10:49:19 +000037 ss << ", encoder: " << (encoder != NULL ? "(encoder)" : "NULL");
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000038 ss << '}';
39 return ss.str();
40}
41
pbos@webrtc.org024e4d52014-05-15 10:03:24 +000042std::string VideoSendStream::Config::Rtp::Rtx::ToString()
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000043 const {
44 std::stringstream ss;
45 ss << "{ssrcs: {";
46 for (size_t i = 0; i < ssrcs.size(); ++i) {
47 ss << ssrcs[i];
48 if (i != ssrcs.size() - 1)
49 ss << "}, {";
50 }
51 ss << '}';
52
53 ss << ", payload_type: " << payload_type;
54 ss << '}';
55 return ss.str();
56}
57
pbos@webrtc.org024e4d52014-05-15 10:03:24 +000058std::string VideoSendStream::Config::Rtp::ToString() const {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000059 std::stringstream ss;
60 ss << "{ssrcs: {";
61 for (size_t i = 0; i < ssrcs.size(); ++i) {
62 ss << ssrcs[i];
63 if (i != ssrcs.size() - 1)
64 ss << "}, {";
65 }
66 ss << '}';
67
68 ss << ", max_packet_size: " << max_packet_size;
69 if (min_transmit_bitrate_bps != 0)
70 ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
71
72 ss << ", extensions: {";
73 for (size_t i = 0; i < extensions.size(); ++i) {
74 ss << extensions[i].ToString();
75 if (i != extensions.size() - 1)
76 ss << "}, {";
77 }
78 ss << '}';
79
80 if (nack.rtp_history_ms != 0)
81 ss << ", nack.rtp_history_ms: " << nack.rtp_history_ms;
82 if (fec.ulpfec_payload_type != -1 || fec.red_payload_type != -1)
83 ss << ", fec: " << fec.ToString();
84 if (rtx.payload_type != 0 || !rtx.ssrcs.empty())
85 ss << ", rtx: " << rtx.ToString();
86 if (c_name != "")
87 ss << ", c_name: " << c_name;
88 ss << '}';
89 return ss.str();
90}
91
pbos@webrtc.org024e4d52014-05-15 10:03:24 +000092std::string VideoSendStream::Config::ToString() const {
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000093 std::stringstream ss;
94 ss << "{encoder_settings: " << encoder_settings.ToString();
95 ss << ", rtp: " << rtp.ToString();
96 if (pre_encode_callback != NULL)
97 ss << ", (pre_encode_callback)";
98 if (post_encode_callback != NULL)
99 ss << ", (post_encode_callback)";
100 if (local_renderer != NULL) {
101 ss << ", (local_renderer, render_delay_ms: " << render_delay_ms << ")";
102 }
103 if (target_delay_ms > 0)
104 ss << ", target_delay_ms: " << target_delay_ms;
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000105 if (suspend_below_min_bitrate)
106 ss << ", suspend_below_min_bitrate: on";
107 ss << '}';
108 return ss.str();
109}
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000110
pbos@webrtc.org024e4d52014-05-15 10:03:24 +0000111namespace internal {
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000112VideoSendStream::VideoSendStream(
113 newapi::Transport* transport,
114 CpuOveruseObserver* overuse_observer,
115 webrtc::VideoEngine* video_engine,
116 const VideoSendStream::Config& config,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000117 const VideoEncoderConfig& encoder_config,
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000118 const std::map<uint32_t, RtpState>& suspended_ssrcs,
119 int base_channel,
120 int start_bitrate_bps)
pbos@webrtc.org64887612013-11-14 08:58:14 +0000121 : transport_adapter_(transport),
sprang@webrtc.org40709352013-11-26 11:41:59 +0000122 encoded_frame_proxy_(config.post_encode_callback),
pbos@webrtc.org64887612013-11-14 08:58:14 +0000123 config_(config),
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000124 start_bitrate_bps_(start_bitrate_bps),
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000125 suspended_ssrcs_(suspended_ssrcs),
mflodman@webrtc.orgf3973e82013-12-13 09:40:45 +0000126 external_codec_(NULL),
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000127 channel_(-1),
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000128 stats_proxy_(config) {
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000129 video_engine_base_ = ViEBase::GetInterface(video_engine);
mflodman@webrtc.orgf3973e82013-12-13 09:40:45 +0000130 video_engine_base_->CreateChannel(channel_, base_channel);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000131 assert(channel_ != -1);
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000132 assert(start_bitrate_bps_ > 0);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000133
134 rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
135 assert(rtp_rtcp_ != NULL);
136
pbos@webrtc.org64887612013-11-14 08:58:14 +0000137 assert(config_.rtp.ssrcs.size() > 0);
pbos@webrtc.org29023282013-09-11 10:14:56 +0000138
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000139 assert(config_.rtp.min_transmit_bitrate_bps >= 0);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000140 rtp_rtcp_->SetMinTransmitBitrate(channel_,
pbos@webrtc.org709e2972014-03-19 10:59:52 +0000141 config_.rtp.min_transmit_bitrate_bps / 1000);
pbos@webrtc.org3349ae02014-03-13 12:52:27 +0000142
pbos@webrtc.org29023282013-09-11 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.orgce90eff2013-11-20 11:48:56 +0000146 if (extension == RtpExtension::kTOffset) {
pbos@webrtc.org29023282013-09-11 10:14:56 +0000147 if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
148 abort();
pbos@webrtc.orgce90eff2013-11-20 11:48:56 +0000149 } else if (extension == RtpExtension::kAbsSendTime) {
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000150 if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
151 abort();
pbos@webrtc.org29023282013-09-11 10:14:56 +0000152 } else {
153 abort(); // Unsupported extension.
154 }
155 }
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000156
mflodman@webrtc.org92c27932013-12-13 16:36:28 +0000157 rtp_rtcp_->SetRembStatus(channel_, true, false);
158
pbos@webrtc.org0e63e762013-09-20 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.orgbe9d2a42014-06-30 13:19:09 +0000179 ConfigureSsrcs();
180
pbos@webrtc.org013d9942013-08-22 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.org29d58392013-05-16 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.orge75a1bf2013-09-18 11:52:42 +0000195 network_->RegisterSendTransport(channel_, transport_adapter_);
sprang@webrtc.org25fce9a2013-10-16 13:29:14 +0000196 // 28 to match packet overhead in ModuleRtpRtcpImpl.
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000197 network_->SetMTU(channel_,
198 static_cast<unsigned int>(config_.rtp.max_packet_size + 28));
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000199
pbos@webrtc.orgf577ae92014-03-19 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.org360e3762013-08-22 09:29:56 +0000210 }
211
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000212 codec_ = ViECodec::GetInterface(video_engine);
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000213 if (!ReconfigureVideoEncoder(encoder_config))
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000214 abort();
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000215
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000216 if (overuse_observer)
217 video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
218
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000219 video_engine_base_->RegisterSendSideDelayObserver(channel_, &stats_proxy_);
220
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000221 image_process_ = ViEImageProcess::GetInterface(video_engine);
222 image_process_->RegisterPreEncodeCallback(channel_,
223 config_.pre_encode_callback);
sprang@webrtc.org40709352013-11-26 11:41:59 +0000224 if (config_.post_encode_callback) {
225 image_process_->RegisterPostEncodeImageCallback(channel_,
226 &encoded_frame_proxy_);
227 }
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000228
pbos@webrtc.org6ae48c62014-06-06 10:49:19 +0000229 if (config_.suspend_below_min_bitrate)
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000230 codec_->SuspendBelowMinBitrate(channel_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000231
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000232 rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000233 &stats_proxy_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000234 rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000235 &stats_proxy_);
236 rtp_rtcp_->RegisterSendBitrateObserver(channel_, &stats_proxy_);
237 rtp_rtcp_->RegisterSendFrameCountObserver(channel_, &stats_proxy_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000238
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000239 codec_->RegisterEncoderObserver(channel_, stats_proxy_);
240 capture_->RegisterObserver(capture_id_, stats_proxy_);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000241}
242
243VideoSendStream::~VideoSendStream() {
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000244 capture_->DeregisterObserver(capture_id_);
245 codec_->DeregisterEncoderObserver(channel_);
246
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000247 rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, &stats_proxy_);
248 rtp_rtcp_->DeregisterSendBitrateObserver(channel_, &stats_proxy_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000249 rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000250 &stats_proxy_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000251 rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000252 &stats_proxy_);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000253
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000254 image_process_->DeRegisterPreEncodeCallback(channel_);
255
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000256 network_->DeregisterSendTransport(channel_);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000257
258 capture_->DisconnectCaptureDevice(channel_);
259 capture_->ReleaseCaptureDevice(capture_id_);
260
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000261 external_codec_->DeRegisterExternalSendCodec(
262 channel_, config_.encoder_settings.payload_type);
stefan@webrtc.org360e3762013-08-22 09:29:56 +0000263
pbos@webrtc.orgfe1ef932013-10-21 10:34:43 +0000264 video_engine_base_->DeleteChannel(channel_);
265
266 image_process_->Release();
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000267 video_engine_base_->Release();
268 capture_->Release();
269 codec_->Release();
stefan@webrtc.org360e3762013-08-22 09:29:56 +0000270 if (external_codec_)
271 external_codec_->Release();
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000272 network_->Release();
273 rtp_rtcp_->Release();
274}
275
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000276void VideoSendStream::SwapFrame(I420VideoFrame* frame) {
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000277 // TODO(pbos): Local rendering should not be done on the capture thread.
278 if (config_.local_renderer != NULL)
pbos@webrtc.org1566ee22014-05-23 13:03:45 +0000279 config_.local_renderer->RenderFrame(*frame, 0);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000280
pbos@webrtc.org1566ee22014-05-23 13:03:45 +0000281 external_capture_->SwapFrame(frame);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000282}
283
pbos@webrtc.org74fa4892013-08-23 09:19:30 +0000284VideoSendStreamInput* VideoSendStream::Input() { return this; }
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000285
pbos@webrtc.orga5c8d2c2014-04-24 11:13:21 +0000286void VideoSendStream::Start() {
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000287 transport_adapter_.Enable();
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +0000288 video_engine_base_->StartSend(channel_);
289 video_engine_base_->StartReceive(channel_);
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000290}
291
pbos@webrtc.orga5c8d2c2014-04-24 11:13:21 +0000292void VideoSendStream::Stop() {
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +0000293 video_engine_base_->StopSend(channel_);
294 video_engine_base_->StopReceive(channel_);
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000295 transport_adapter_.Disable();
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000296}
297
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000298bool VideoSendStream::ReconfigureVideoEncoder(
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000299 const VideoEncoderConfig& config) {
300 const std::vector<VideoStream>& streams = config.streams;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000301 assert(!streams.empty());
302 assert(config_.rtp.ssrcs.size() >= streams.size());
pbos@webrtc.org64887612013-11-14 08:58:14 +0000303
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000304 VideoCodec video_codec;
305 memset(&video_codec, 0, sizeof(video_codec));
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000306 if (config_.encoder_settings.payload_name == "VP8") {
307 video_codec.codecType = kVideoCodecVP8;
308 } else if (config_.encoder_settings.payload_name == "H264") {
309 video_codec.codecType = kVideoCodecH264;
310 } else {
311 video_codec.codecType = kVideoCodecGeneric;
312 }
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000313 switch (config.content_type) {
314 case VideoEncoderConfig::kRealtimeVideo:
315 video_codec.mode = kRealtimeVideo;
316 break;
317 case VideoEncoderConfig::kScreenshare:
318 video_codec.mode = kScreensharing;
319 break;
320 }
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000321
322 if (video_codec.codecType == kVideoCodecVP8) {
pbos@webrtc.org6cd6ba82014-09-18 12:42:28 +0000323 video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000324 } else if (video_codec.codecType == kVideoCodecH264) {
pbos@webrtc.org6cd6ba82014-09-18 12:42:28 +0000325 video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000326 }
327
pbos@webrtc.org91f17522014-07-10 10:13:37 +0000328 if (video_codec.codecType == kVideoCodecVP8) {
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000329 if (config.encoder_specific_settings != NULL) {
330 video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>(
331 config.encoder_specific_settings);
pbos@webrtc.org91f17522014-07-10 10:13:37 +0000332 }
333 } else {
334 // TODO(pbos): Support encoder_settings codec-agnostically.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000335 assert(config.encoder_specific_settings == NULL);
pbos@webrtc.org91f17522014-07-10 10:13:37 +0000336 }
337
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000338 strncpy(video_codec.plName,
339 config_.encoder_settings.payload_name.c_str(),
340 kPayloadNameSize - 1);
341 video_codec.plName[kPayloadNameSize - 1] = '\0';
342 video_codec.plType = config_.encoder_settings.payload_type;
343 video_codec.numberOfSimulcastStreams =
344 static_cast<unsigned char>(streams.size());
345 video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
346 assert(streams.size() <= kMaxSimulcastStreams);
347 for (size_t i = 0; i < streams.size(); ++i) {
348 SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
349 assert(streams[i].width > 0);
350 assert(streams[i].height > 0);
351 assert(streams[i].max_framerate > 0);
352 // Different framerates not supported per stream at the moment.
353 assert(streams[i].max_framerate == streams[0].max_framerate);
354 assert(streams[i].min_bitrate_bps >= 0);
355 assert(streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps);
356 assert(streams[i].max_bitrate_bps >= streams[i].target_bitrate_bps);
357 assert(streams[i].max_qp >= 0);
358
359 sim_stream->width = static_cast<unsigned short>(streams[i].width);
360 sim_stream->height = static_cast<unsigned short>(streams[i].height);
361 sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
362 sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
363 sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
364 sim_stream->qpMax = streams[i].max_qp;
365 // TODO(pbos): Implement mapping for temporal layers.
366 assert(streams[i].temporal_layers.empty());
367
368 video_codec.width = std::max(video_codec.width,
369 static_cast<unsigned short>(streams[i].width));
370 video_codec.height = std::max(
371 video_codec.height, static_cast<unsigned short>(streams[i].height));
372 video_codec.minBitrate =
373 std::min(video_codec.minBitrate,
374 static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
375 video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
376 video_codec.qpMax = std::max(video_codec.qpMax,
377 static_cast<unsigned int>(streams[i].max_qp));
378 }
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000379 video_codec.startBitrate =
380 static_cast<unsigned int>(start_bitrate_bps_) / 1000;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000381
382 if (video_codec.minBitrate < kViEMinCodecBitrate)
383 video_codec.minBitrate = kViEMinCodecBitrate;
384 if (video_codec.maxBitrate < kViEMinCodecBitrate)
385 video_codec.maxBitrate = kViEMinCodecBitrate;
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000386 if (video_codec.startBitrate < video_codec.minBitrate)
387 video_codec.startBitrate = video_codec.minBitrate;
388 if (video_codec.startBitrate > video_codec.maxBitrate)
389 video_codec.startBitrate = video_codec.maxBitrate;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000390
391 if (video_codec.startBitrate < video_codec.minBitrate)
392 video_codec.startBitrate = video_codec.minBitrate;
393 if (video_codec.startBitrate > video_codec.maxBitrate)
394 video_codec.startBitrate = video_codec.maxBitrate;
395
pbos@webrtc.org6ae48c62014-06-06 10:49:19 +0000396 assert(streams[0].max_framerate > 0);
397 video_codec.maxFramerate = streams[0].max_framerate;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000398
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000399 return codec_->SetSendCodec(channel_, video_codec) == 0;
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000400}
401
pbos@webrtc.orgbbb07e62013-08-05 12:01:36 +0000402bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
403 return network_->ReceivedRTCPPacket(
pbos@webrtc.orgccdcbae2013-08-05 13:25:51 +0000404 channel_, packet, static_cast<int>(length)) == 0;
pbos@webrtc.orgbbb07e62013-08-05 12:01:36 +0000405}
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000406
407VideoSendStream::Stats VideoSendStream::GetStats() const {
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000408 return stats_proxy_.GetStats();
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000409}
410
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000411void VideoSendStream::ConfigureSsrcs() {
412 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
413 uint32_t ssrc = config_.rtp.ssrcs[i];
414 rtp_rtcp_->SetLocalSSRC(
415 channel_, ssrc, kViEStreamTypeNormal, static_cast<unsigned char>(i));
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000416 RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
417 if (it != suspended_ssrcs_.end())
418 rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000419 }
420
421 if (config_.rtp.rtx.ssrcs.empty()) {
422 assert(!config_.rtp.rtx.pad_with_redundant_payloads);
423 return;
424 }
425
426 // Set up RTX.
427 assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000428 for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
429 uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000430 rtp_rtcp_->SetLocalSSRC(channel_,
431 config_.rtp.rtx.ssrcs[i],
432 kViEStreamTypeRtx,
433 static_cast<unsigned char>(i));
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000434 RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
435 if (it != suspended_ssrcs_.end())
436 rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000437 }
438
439 if (config_.rtp.rtx.pad_with_redundant_payloads) {
440 rtp_rtcp_->SetPadWithRedundantPayloads(channel_, true);
441 }
442
443 assert(config_.rtp.rtx.payload_type >= 0);
444 rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
445}
446
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000447std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
448 std::map<uint32_t, RtpState> rtp_states;
449 for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
450 uint32_t ssrc = config_.rtp.ssrcs[i];
451 rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
452 }
453
454 for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
455 uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
456 rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
457 }
458
459 return rtp_states;
460}
461
pbos@webrtc.org26c0c412014-09-03 16:17:12 +0000462void VideoSendStream::SignalNetworkState(Call::NetworkState state) {
463 // When network goes up, enable RTCP status before setting transmission state.
464 // When it goes down, disable RTCP afterwards. This ensures that any packets
465 // sent due to the network state changed will not be dropped.
466 if (state == Call::kNetworkUp)
467 rtp_rtcp_->SetRTCPStatus(channel_, kRtcpCompound_RFC4585);
468 network_->SetNetworkTransmissionState(channel_, state == Call::kNetworkUp);
469 if (state == Call::kNetworkDown)
470 rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
471}
472
pbos@webrtc.org29d58392013-05-16 12:08:03 +0000473} // namespace internal
474} // namespace webrtc