webrtcvideoengine2.cc revision 09c77b95bb62566be64da662f0b3b6a838ec6553 
1/*
2 * libjingle
3 * Copyright 2014 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 *  1. Redistributions of source code must retain the above copyright notice,
9 *     this list of conditions and the following disclaimer.
10 *  2. Redistributions in binary form must reproduce the above copyright notice,
11 *     this list of conditions and the following disclaimer in the documentation
12 *     and/or other materials provided with the distribution.
13 *  3. The name of the author may not be used to endorse or promote products
14 *     derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifdef HAVE_WEBRTC_VIDEO
29#include "talk/media/webrtc/webrtcvideoengine2.h"
30
31#include <algorithm>
32#include <set>
33#include <string>
34
35#include "libyuv/convert_from.h"
36#include "talk/media/base/videocapturer.h"
37#include "talk/media/base/videorenderer.h"
38#include "talk/media/webrtc/constants.h"
39#include "talk/media/webrtc/simulcast.h"
40#include "talk/media/webrtc/webrtcvideocapturer.h"
41#include "talk/media/webrtc/webrtcvideoengine.h"
42#include "talk/media/webrtc/webrtcvideoframe.h"
43#include "talk/media/webrtc/webrtcvoiceengine.h"
44#include "webrtc/base/buffer.h"
45#include "webrtc/base/logging.h"
46#include "webrtc/base/stringutils.h"
47#include "webrtc/call.h"
48#include "webrtc/system_wrappers/interface/trace_event.h"
49#include "webrtc/video_decoder.h"
50#include "webrtc/video_encoder.h"
51
52#define UNIMPLEMENTED                                                 \
53  LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
54  ASSERT(false)
55
56namespace cricket {
57namespace {
58static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
59  std::stringstream out;
60  out << '{';
61  for (size_t i = 0; i < codecs.size(); ++i) {
62    out << codecs[i].ToString();
63    if (i != codecs.size() - 1) {
64      out << ", ";
65    }
66  }
67  out << '}';
68  return out.str();
69}
70
71static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) {
72  bool has_video = false;
73  for (size_t i = 0; i < codecs.size(); ++i) {
74    if (!codecs[i].ValidateCodecFormat()) {
75      return false;
76    }
77    if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) {
78      has_video = true;
79    }
80  }
81  if (!has_video) {
82    LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: "
83                  << CodecVectorToString(codecs);
84    return false;
85  }
86  return true;
87}
88
89static std::string RtpExtensionsToString(
90    const std::vector<RtpHeaderExtension>& extensions) {
91  std::stringstream out;
92  out << '{';
93  for (size_t i = 0; i < extensions.size(); ++i) {
94    out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
95    if (i != extensions.size() - 1) {
96      out << ", ";
97    }
98  }
99  out << '}';
100  return out.str();
101}
102
103// Merges two fec configs and logs an error if a conflict arises
104// such that merging in diferent order would trigger a diferent output.
105static void MergeFecConfig(const webrtc::FecConfig& other,
106                           webrtc::FecConfig* output) {
107  if (other.ulpfec_payload_type != -1) {
108    if (output->ulpfec_payload_type != -1 &&
109        output->ulpfec_payload_type != other.ulpfec_payload_type) {
110      LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
111                      << output->ulpfec_payload_type << " and "
112                      << other.ulpfec_payload_type;
113    }
114    output->ulpfec_payload_type = other.ulpfec_payload_type;
115  }
116  if (other.red_payload_type != -1) {
117    if (output->red_payload_type != -1 &&
118        output->red_payload_type != other.red_payload_type) {
119      LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
120                      << output->red_payload_type << " and "
121                      << other.red_payload_type;
122    }
123    output->red_payload_type = other.red_payload_type;
124  }
125}
126}  // namespace
127
128// This constant is really an on/off, lower-level configurable NACK history
129// duration hasn't been implemented.
130static const int kNackHistoryMs = 1000;
131
132static const int kDefaultQpMax = 56;
133
134static const int kDefaultRtcpReceiverReportSsrc = 1;
135
136const char kH264CodecName[] = "H264";
137
138static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs,
139                                   const VideoCodec& requested_codec,
140                                   VideoCodec* matching_codec) {
141  for (size_t i = 0; i < codecs.size(); ++i) {
142    if (requested_codec.Matches(codecs[i])) {
143      *matching_codec = codecs[i];
144      return true;
145    }
146  }
147  return false;
148}
149
150static bool ValidateRtpHeaderExtensionIds(
151    const std::vector<RtpHeaderExtension>& extensions) {
152  std::set<int> extensions_used;
153  for (size_t i = 0; i < extensions.size(); ++i) {
154    if (extensions[i].id < 0 || extensions[i].id >= 15 ||
155        !extensions_used.insert(extensions[i].id).second) {
156      LOG(LS_ERROR) << "RTP extensions are with incorrect or duplicate ids.";
157      return false;
158    }
159  }
160  return true;
161}
162
163static bool CompareRtpHeaderExtensionIds(
164    const webrtc::RtpExtension& extension1,
165    const webrtc::RtpExtension& extension2) {
166  // Sorting on ID is sufficient, more than one extension per ID is unsupported.
167  return extension1.id > extension2.id;
168}
169
170static std::vector<webrtc::RtpExtension> FilterRtpExtensions(
171    const std::vector<RtpHeaderExtension>& extensions) {
172  std::vector<webrtc::RtpExtension> webrtc_extensions;
173  for (size_t i = 0; i < extensions.size(); ++i) {
174    // Unsupported extensions will be ignored.
175    if (webrtc::RtpExtension::IsSupported(extensions[i].uri)) {
176      webrtc_extensions.push_back(webrtc::RtpExtension(
177          extensions[i].uri, extensions[i].id));
178    } else {
179      LOG(LS_WARNING) << "Unsupported RTP extension: " << extensions[i].uri;
180    }
181  }
182
183  // Sort filtered headers to make sure that they can later be compared
184  // regardless of in which order they were entered.
185  std::sort(webrtc_extensions.begin(), webrtc_extensions.end(),
186            CompareRtpHeaderExtensionIds);
187  return webrtc_extensions;
188}
189
190static bool RtpExtensionsHaveChanged(
191    const std::vector<webrtc::RtpExtension>& before,
192    const std::vector<webrtc::RtpExtension>& after) {
193  if (before.size() != after.size())
194    return true;
195  for (size_t i = 0; i < before.size(); ++i) {
196    if (before[i].id != after[i].id)
197      return true;
198    if (before[i].name != after[i].name)
199      return true;
200  }
201  return false;
202}
203
204std::vector<webrtc::VideoStream>
205WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
206    const VideoCodec& codec,
207    const VideoOptions& options,
208    size_t num_streams) {
209  // Use default factory for non-simulcast.
210  int max_qp = kDefaultQpMax;
211  codec.GetParam(kCodecParamMaxQuantization, &max_qp);
212
213  int min_bitrate_kbps;
214  if (!codec.GetParam(kCodecParamMinBitrate, &min_bitrate_kbps) ||
215      min_bitrate_kbps < kMinVideoBitrate) {
216    min_bitrate_kbps = kMinVideoBitrate;
217  }
218
219  int max_bitrate_kbps;
220  if (!codec.GetParam(kCodecParamMaxBitrate, &max_bitrate_kbps)) {
221    max_bitrate_kbps = 0;
222  }
223
224  return GetSimulcastConfig(
225      num_streams,
226      GetSimulcastBitrateMode(options),
227      codec.width,
228      codec.height,
229      max_bitrate_kbps * 1000,
230      max_qp,
231      codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate);
232}
233
234std::vector<webrtc::VideoStream>
235WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoStreams(
236    const VideoCodec& codec,
237    const VideoOptions& options,
238    size_t num_streams) {
239  if (num_streams != 1)
240    return CreateSimulcastVideoStreams(codec, options, num_streams);
241
242  webrtc::VideoStream stream;
243  stream.width = codec.width;
244  stream.height = codec.height;
245  stream.max_framerate =
246      codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
247
248  stream.min_bitrate_bps = kMinVideoBitrate * 1000;
249  stream.target_bitrate_bps = stream.max_bitrate_bps = kMaxVideoBitrate * 1000;
250
251  int max_qp = kDefaultQpMax;
252  codec.GetParam(kCodecParamMaxQuantization, &max_qp);
253  stream.max_qp = max_qp;
254  std::vector<webrtc::VideoStream> streams;
255  streams.push_back(stream);
256  return streams;
257}
258
259void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
260    const VideoCodec& codec,
261    const VideoOptions& options) {
262  if (CodecNameMatches(codec.name, kVp8CodecName)) {
263    encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
264    options.video_noise_reduction.Get(&encoder_settings_.vp8.denoisingOn);
265    return &encoder_settings_.vp8;
266  }
267  if (CodecNameMatches(codec.name, kVp9CodecName)) {
268    encoder_settings_.vp9 = webrtc::VideoEncoder::GetDefaultVp9Settings();
269    options.video_noise_reduction.Get(&encoder_settings_.vp9.denoisingOn);
270    return &encoder_settings_.vp9;
271  }
272  return NULL;
273}
274
275DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
276    : default_recv_ssrc_(0), default_renderer_(NULL) {}
277
278UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
279    VideoMediaChannel* channel,
280    uint32_t ssrc) {
281  if (default_recv_ssrc_ != 0) {  // Already one default stream.
282    LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set.";
283    return kDropPacket;
284  }
285
286  StreamParams sp;
287  sp.ssrcs.push_back(ssrc);
288  LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
289  if (!channel->AddRecvStream(sp)) {
290    LOG(LS_WARNING) << "Could not create default receive stream.";
291  }
292
293  channel->SetRenderer(ssrc, default_renderer_);
294  default_recv_ssrc_ = ssrc;
295  return kDeliverPacket;
296}
297
298WebRtcCallFactory::~WebRtcCallFactory() {
299}
300webrtc::Call* WebRtcCallFactory::CreateCall(
301    const webrtc::Call::Config& config) {
302  return webrtc::Call::Create(config);
303}
304
305VideoRenderer* DefaultUnsignalledSsrcHandler::GetDefaultRenderer() const {
306  return default_renderer_;
307}
308
309void DefaultUnsignalledSsrcHandler::SetDefaultRenderer(
310    VideoMediaChannel* channel,
311    VideoRenderer* renderer) {
312  default_renderer_ = renderer;
313  if (default_recv_ssrc_ != 0) {
314    channel->SetRenderer(default_recv_ssrc_, default_renderer_);
315  }
316}
317
318WebRtcVideoEngine2::WebRtcVideoEngine2()
319    : worker_thread_(NULL),
320      voice_engine_(NULL),
321      default_codec_format_(kDefaultVideoMaxWidth,
322                            kDefaultVideoMaxHeight,
323                            FPS_TO_INTERVAL(kDefaultVideoMaxFramerate),
324                            FOURCC_ANY),
325      initialized_(false),
326      call_factory_(&default_call_factory_),
327      external_decoder_factory_(NULL),
328      external_encoder_factory_(NULL) {
329  LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
330  video_codecs_ = GetSupportedCodecs();
331  rtp_header_extensions_.push_back(
332      RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
333                         kRtpTimestampOffsetHeaderExtensionDefaultId));
334  rtp_header_extensions_.push_back(
335      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
336                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
337}
338
339WebRtcVideoEngine2::~WebRtcVideoEngine2() {
340  LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
341
342  if (initialized_) {
343    Terminate();
344  }
345}
346
347void WebRtcVideoEngine2::SetCallFactory(WebRtcCallFactory* call_factory) {
348  assert(!initialized_);
349  call_factory_ = call_factory;
350}
351
352bool WebRtcVideoEngine2::Init(rtc::Thread* worker_thread) {
353  LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
354  worker_thread_ = worker_thread;
355  ASSERT(worker_thread_ != NULL);
356
357  initialized_ = true;
358  return true;
359}
360
361void WebRtcVideoEngine2::Terminate() {
362  LOG(LS_INFO) << "WebRtcVideoEngine2::Terminate";
363
364  initialized_ = false;
365}
366
367int WebRtcVideoEngine2::GetCapabilities() { return VIDEO_RECV | VIDEO_SEND; }
368
369bool WebRtcVideoEngine2::SetDefaultEncoderConfig(
370    const VideoEncoderConfig& config) {
371  const VideoCodec& codec = config.max_codec;
372  bool supports_codec = false;
373  for (size_t i = 0; i < video_codecs_.size(); ++i) {
374    if (CodecNameMatches(video_codecs_[i].name, codec.name)) {
375      video_codecs_[i] = codec;
376      supports_codec = true;
377      break;
378    }
379  }
380
381  if (!supports_codec) {
382    LOG(LS_ERROR) << "SetDefaultEncoderConfig, codec not supported: "
383                  << codec.ToString();
384    return false;
385  }
386
387  default_codec_format_ =
388      VideoFormat(codec.width,
389                  codec.height,
390                  VideoFormat::FpsToInterval(codec.framerate),
391                  FOURCC_ANY);
392  return true;
393}
394
395WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
396    const VideoOptions& options,
397    VoiceMediaChannel* voice_channel) {
398  assert(initialized_);
399  LOG(LS_INFO) << "CreateChannel: "
400               << (voice_channel != NULL ? "With" : "Without")
401               << " voice channel. Options: " << options.ToString();
402  WebRtcVideoChannel2* channel =
403      new WebRtcVideoChannel2(call_factory_,
404                              voice_engine_,
405                              voice_channel,
406                              options,
407                              external_encoder_factory_,
408                              external_decoder_factory_);
409  if (!channel->Init()) {
410    delete channel;
411    return NULL;
412  }
413  channel->SetRecvCodecs(video_codecs_);
414  return channel;
415}
416
417const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
418  return video_codecs_;
419}
420
421const std::vector<RtpHeaderExtension>&
422WebRtcVideoEngine2::rtp_header_extensions() const {
423  return rtp_header_extensions_;
424}
425
426void WebRtcVideoEngine2::SetLogging(int min_sev, const char* filter) {
427  // TODO(pbos): Set up logging.
428  LOG(LS_VERBOSE) << "SetLogging: " << min_sev << '"' << filter << '"';
429  // if min_sev == -1, we keep the current log level.
430  if (min_sev < 0) {
431    assert(min_sev == -1);
432    return;
433  }
434}
435
436void WebRtcVideoEngine2::SetExternalDecoderFactory(
437    WebRtcVideoDecoderFactory* decoder_factory) {
438  assert(!initialized_);
439  external_decoder_factory_ = decoder_factory;
440}
441
442void WebRtcVideoEngine2::SetExternalEncoderFactory(
443    WebRtcVideoEncoderFactory* encoder_factory) {
444  assert(!initialized_);
445  if (external_encoder_factory_ == encoder_factory)
446    return;
447
448  // No matter what happens we shouldn't hold on to a stale
449  // WebRtcSimulcastEncoderFactory.
450  simulcast_encoder_factory_.reset();
451
452  if (encoder_factory &&
453      WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory(
454          encoder_factory->codecs())) {
455    simulcast_encoder_factory_.reset(
456        new WebRtcSimulcastEncoderFactory(encoder_factory));
457    encoder_factory = simulcast_encoder_factory_.get();
458  }
459  external_encoder_factory_ = encoder_factory;
460
461  video_codecs_ = GetSupportedCodecs();
462}
463
464bool WebRtcVideoEngine2::EnableTimedRender() {
465  // TODO(pbos): Figure out whether this can be removed.
466  return true;
467}
468
469// Checks to see whether we comprehend and could receive a particular codec
470bool WebRtcVideoEngine2::FindCodec(const VideoCodec& in) {
471  // TODO(pbos): Probe encoder factory to figure out that the codec is supported
472  // if supported by the encoder factory. Add a corresponding test that fails
473  // with this code (that doesn't ask the factory).
474  for (size_t j = 0; j < video_codecs_.size(); ++j) {
475    VideoCodec codec(video_codecs_[j].id, video_codecs_[j].name, 0, 0, 0, 0);
476    if (codec.Matches(in)) {
477      return true;
478    }
479  }
480  return false;
481}
482
483// Tells whether the |requested| codec can be transmitted or not. If it can be
484// transmitted |out| is set with the best settings supported. Aspect ratio will
485// be set as close to |current|'s as possible. If not set |requested|'s
486// dimensions will be used for aspect ratio matching.
487bool WebRtcVideoEngine2::CanSendCodec(const VideoCodec& requested,
488                                      const VideoCodec& current,
489                                      VideoCodec* out) {
490  assert(out != NULL);
491
492  if (requested.width != requested.height &&
493      (requested.height == 0 || requested.width == 0)) {
494    // 0xn and nx0 are invalid resolutions.
495    return false;
496  }
497
498  VideoCodec matching_codec;
499  if (!FindFirstMatchingCodec(video_codecs_, requested, &matching_codec)) {
500    // Codec not supported.
501    return false;
502  }
503
504  out->id = requested.id;
505  out->name = requested.name;
506  out->preference = requested.preference;
507  out->params = requested.params;
508  out->framerate = std::min(requested.framerate, matching_codec.framerate);
509  out->params = requested.params;
510  out->feedback_params = requested.feedback_params;
511  out->width = requested.width;
512  out->height = requested.height;
513  if (requested.width == 0 && requested.height == 0) {
514    return true;
515  }
516
517  while (out->width > matching_codec.width) {
518    out->width /= 2;
519    out->height /= 2;
520  }
521
522  return out->width > 0 && out->height > 0;
523}
524
525bool WebRtcVideoEngine2::SetVoiceEngine(WebRtcVoiceEngine* voice_engine) {
526  if (initialized_) {
527    LOG(LS_WARNING) << "SetVoiceEngine can not be called after Init";
528    return false;
529  }
530  voice_engine_ = voice_engine;
531  return true;
532}
533
534// Ignore spammy trace messages, mostly from the stats API when we haven't
535// gotten RTCP info yet from the remote side.
536bool WebRtcVideoEngine2::ShouldIgnoreTrace(const std::string& trace) {
537  static const char* const kTracesToIgnore[] = {NULL};
538  for (const char* const* p = kTracesToIgnore; *p; ++p) {
539    if (trace.find(*p) == 0) {
540      return true;
541    }
542  }
543  return false;
544}
545
546std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
547  std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
548
549  if (external_encoder_factory_ == NULL) {
550    return supported_codecs;
551  }
552
553  const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
554      external_encoder_factory_->codecs();
555  for (size_t i = 0; i < codecs.size(); ++i) {
556    // Don't add internally-supported codecs twice.
557    if (CodecIsInternallySupported(codecs[i].name)) {
558      continue;
559    }
560
561    // External video encoders are given payloads 120-127. This also means that
562    // we only support up to 8 external payload types.
563    const int kExternalVideoPayloadTypeBase = 120;
564    size_t payload_type = kExternalVideoPayloadTypeBase + i;
565    assert(payload_type < 128);
566    VideoCodec codec(static_cast<int>(payload_type),
567                     codecs[i].name,
568                     codecs[i].max_width,
569                     codecs[i].max_height,
570                     codecs[i].max_fps,
571                     0);
572
573    AddDefaultFeedbackParams(&codec);
574    supported_codecs.push_back(codec);
575  }
576  return supported_codecs;
577}
578
579WebRtcVideoChannel2::WebRtcVideoChannel2(
580    WebRtcCallFactory* call_factory,
581    WebRtcVoiceEngine* voice_engine,
582    VoiceMediaChannel* voice_channel,
583    const VideoOptions& options,
584    WebRtcVideoEncoderFactory* external_encoder_factory,
585    WebRtcVideoDecoderFactory* external_decoder_factory)
586    : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
587      voice_channel_(voice_channel),
588      external_encoder_factory_(external_encoder_factory),
589      external_decoder_factory_(external_decoder_factory) {
590  SetDefaultOptions();
591  options_.SetAll(options);
592  webrtc::Call::Config config(this);
593  config.overuse_callback = this;
594  if (voice_engine != NULL) {
595    config.voice_engine = voice_engine->voe()->engine();
596  }
597
598  call_.reset(call_factory->CreateCall(config));
599
600  rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
601  sending_ = false;
602  default_send_ssrc_ = 0;
603}
604
605void WebRtcVideoChannel2::SetDefaultOptions() {
606  options_.cpu_overuse_detection.Set(false);
607  options_.dscp.Set(false);
608  options_.suspend_below_min_bitrate.Set(false);
609  options_.video_noise_reduction.Set(true);
610  options_.screencast_min_bitrate.Set(0);
611}
612
613WebRtcVideoChannel2::~WebRtcVideoChannel2() {
614  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
615           send_streams_.begin();
616       it != send_streams_.end();
617       ++it) {
618    delete it->second;
619  }
620
621  for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
622           receive_streams_.begin();
623       it != receive_streams_.end();
624       ++it) {
625    delete it->second;
626  }
627}
628
629bool WebRtcVideoChannel2::Init() { return true; }
630
631bool WebRtcVideoChannel2::CodecIsExternallySupported(
632    const std::string& name) const {
633  if (external_encoder_factory_ == NULL) {
634    return false;
635  }
636
637  const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
638      external_encoder_factory_->codecs();
639  for (size_t c = 0; c < external_codecs.size(); ++c) {
640    if (CodecNameMatches(name, external_codecs[c].name)) {
641      return true;
642    }
643  }
644  return false;
645}
646
647std::vector<WebRtcVideoChannel2::VideoCodecSettings>
648WebRtcVideoChannel2::FilterSupportedCodecs(
649    const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
650    const {
651  std::vector<VideoCodecSettings> supported_codecs;
652  for (size_t i = 0; i < mapped_codecs.size(); ++i) {
653    const VideoCodecSettings& codec = mapped_codecs[i];
654    if (CodecIsInternallySupported(codec.codec.name) ||
655        CodecIsExternallySupported(codec.codec.name)) {
656      supported_codecs.push_back(codec);
657    }
658  }
659  return supported_codecs;
660}
661
662bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
663  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvCodecs");
664  LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
665  if (!ValidateCodecFormats(codecs)) {
666    return false;
667  }
668
669  const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs);
670  if (mapped_codecs.empty()) {
671    LOG(LS_ERROR) << "SetRecvCodecs called without any video codecs.";
672    return false;
673  }
674
675  const std::vector<VideoCodecSettings> supported_codecs =
676      FilterSupportedCodecs(mapped_codecs);
677
678  if (mapped_codecs.size() != supported_codecs.size()) {
679    LOG(LS_ERROR) << "SetRecvCodecs called with unsupported video codecs.";
680    return false;
681  }
682
683  recv_codecs_ = supported_codecs;
684
685  rtc::CritScope stream_lock(&stream_crit_);
686  for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
687           receive_streams_.begin();
688       it != receive_streams_.end();
689       ++it) {
690    it->second->SetRecvCodecs(recv_codecs_);
691  }
692
693  return true;
694}
695
696bool WebRtcVideoChannel2::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
697  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendCodecs");
698  LOG(LS_INFO) << "SetSendCodecs: " << CodecVectorToString(codecs);
699  if (!ValidateCodecFormats(codecs)) {
700    return false;
701  }
702
703  const std::vector<VideoCodecSettings> supported_codecs =
704      FilterSupportedCodecs(MapCodecs(codecs));
705
706  if (supported_codecs.empty()) {
707    LOG(LS_ERROR) << "No video codecs supported by encoder factory.";
708    return false;
709  }
710
711  LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
712
713  VideoCodecSettings old_codec;
714  if (send_codec_.Get(&old_codec) && supported_codecs.front() == old_codec) {
715    // Using same codec, avoid reconfiguring.
716    return true;
717  }
718
719  send_codec_.Set(supported_codecs.front());
720
721  rtc::CritScope stream_lock(&stream_crit_);
722  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
723           send_streams_.begin();
724       it != send_streams_.end();
725       ++it) {
726    assert(it->second != NULL);
727    it->second->SetCodec(supported_codecs.front());
728  }
729
730  VideoCodec codec = supported_codecs.front().codec;
731  int bitrate_kbps;
732  if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
733      bitrate_kbps > 0) {
734    bitrate_config_.min_bitrate_bps = bitrate_kbps * 1000;
735  } else {
736    bitrate_config_.min_bitrate_bps = 0;
737  }
738  if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
739      bitrate_kbps > 0) {
740    bitrate_config_.start_bitrate_bps = bitrate_kbps * 1000;
741  } else {
742    // Do not reconfigure start bitrate unless it's specified and positive.
743    bitrate_config_.start_bitrate_bps = -1;
744  }
745  if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
746      bitrate_kbps > 0) {
747    bitrate_config_.max_bitrate_bps = bitrate_kbps * 1000;
748  } else {
749    bitrate_config_.max_bitrate_bps = -1;
750  }
751  call_->SetBitrateConfig(bitrate_config_);
752
753  return true;
754}
755
756bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
757  VideoCodecSettings codec_settings;
758  if (!send_codec_.Get(&codec_settings)) {
759    LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
760    return false;
761  }
762  *codec = codec_settings.codec;
763  return true;
764}
765
766bool WebRtcVideoChannel2::SetSendStreamFormat(uint32 ssrc,
767                                              const VideoFormat& format) {
768  LOG(LS_VERBOSE) << "SetSendStreamFormat:" << ssrc << " -> "
769                  << format.ToString();
770  rtc::CritScope stream_lock(&stream_crit_);
771  if (send_streams_.find(ssrc) == send_streams_.end()) {
772    return false;
773  }
774  return send_streams_[ssrc]->SetVideoFormat(format);
775}
776
777bool WebRtcVideoChannel2::SetRender(bool render) {
778  // TODO(pbos): Implement. Or refactor away as it shouldn't be needed.
779  LOG(LS_VERBOSE) << "SetRender: " << (render ? "true" : "false");
780  return true;
781}
782
783bool WebRtcVideoChannel2::SetSend(bool send) {
784  LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
785  if (send && !send_codec_.IsSet()) {
786    LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
787    return false;
788  }
789  if (send) {
790    StartAllSendStreams();
791  } else {
792    StopAllSendStreams();
793  }
794  sending_ = send;
795  return true;
796}
797
798bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
799  LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
800  if (sp.ssrcs.empty()) {
801    LOG(LS_ERROR) << "No SSRCs in stream parameters.";
802    return false;
803  }
804
805  uint32 ssrc = sp.first_ssrc();
806  assert(ssrc != 0);
807  // TODO(pbos): Make sure none of sp.ssrcs are used, not just the identifying
808  // ssrc.
809  rtc::CritScope stream_lock(&stream_crit_);
810  if (send_streams_.find(ssrc) != send_streams_.end()) {
811    LOG(LS_ERROR) << "Send stream with ssrc '" << ssrc << "' already exists.";
812    return false;
813  }
814
815  std::vector<uint32> primary_ssrcs;
816  sp.GetPrimarySsrcs(&primary_ssrcs);
817  std::vector<uint32> rtx_ssrcs;
818  sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs);
819  if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
820    LOG(LS_ERROR)
821        << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
822        << sp.ToString();
823    return false;
824  }
825
826  WebRtcVideoSendStream* stream =
827      new WebRtcVideoSendStream(call_.get(),
828                                external_encoder_factory_,
829                                options_,
830                                send_codec_,
831                                sp,
832                                send_rtp_extensions_);
833
834  send_streams_[ssrc] = stream;
835
836  if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
837    rtcp_receiver_report_ssrc_ = ssrc;
838  }
839  if (default_send_ssrc_ == 0) {
840    default_send_ssrc_ = ssrc;
841  }
842  if (sending_) {
843    stream->Start();
844  }
845
846  return true;
847}
848
849bool WebRtcVideoChannel2::RemoveSendStream(uint32 ssrc) {
850  LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
851
852  if (ssrc == 0) {
853    if (default_send_ssrc_ == 0) {
854      LOG(LS_ERROR) << "No default send stream active.";
855      return false;
856    }
857
858    LOG(LS_VERBOSE) << "Removing default stream: " << default_send_ssrc_;
859    ssrc = default_send_ssrc_;
860  }
861
862  WebRtcVideoSendStream* removed_stream;
863  {
864    rtc::CritScope stream_lock(&stream_crit_);
865    std::map<uint32, WebRtcVideoSendStream*>::iterator it =
866        send_streams_.find(ssrc);
867    if (it == send_streams_.end()) {
868      return false;
869    }
870
871    removed_stream = it->second;
872    send_streams_.erase(it);
873  }
874
875  delete removed_stream;
876
877  if (ssrc == default_send_ssrc_) {
878    default_send_ssrc_ = 0;
879  }
880
881  return true;
882}
883
884bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) {
885  LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
886  assert(sp.ssrcs.size() > 0);
887
888  uint32 ssrc = sp.first_ssrc();
889  assert(ssrc != 0);  // TODO(pbos): Is this ever valid?
890
891  // TODO(pbos): Check if any of the SSRCs overlap.
892  rtc::CritScope stream_lock(&stream_crit_);
893  if (receive_streams_.find(ssrc) != receive_streams_.end()) {
894    LOG(LS_ERROR) << "Receive stream for SSRC " << ssrc << "already exists.";
895    return false;
896  }
897
898  webrtc::VideoReceiveStream::Config config;
899  ConfigureReceiverRtp(&config, sp);
900
901  // Set up A/V sync if there is a VoiceChannel.
902  // TODO(pbos): The A/V is synched by the receiving channel. So we need to know
903  // the SSRC of the remote audio channel in order to sync the correct webrtc
904  // VoiceEngine channel. For now sync the first channel in non-conference to
905  // match existing behavior in WebRtcVideoEngine.
906  if (voice_channel_ != NULL && receive_streams_.empty() &&
907      !options_.conference_mode.GetWithDefaultIfUnset(false)) {
908    config.audio_channel_id =
909        static_cast<WebRtcVoiceMediaChannel*>(voice_channel_)->voe_channel();
910  }
911
912  receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
913      call_.get(), external_decoder_factory_, config, recv_codecs_);
914
915  return true;
916}
917
918void WebRtcVideoChannel2::ConfigureReceiverRtp(
919    webrtc::VideoReceiveStream::Config* config,
920    const StreamParams& sp) const {
921  uint32 ssrc = sp.first_ssrc();
922
923  config->rtp.remote_ssrc = ssrc;
924  config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
925
926  config->rtp.extensions = recv_rtp_extensions_;
927
928  // TODO(pbos): This protection is against setting the same local ssrc as
929  // remote which is not permitted by the lower-level API. RTCP requires a
930  // corresponding sender SSRC. Figure out what to do when we don't have
931  // (receive-only) or know a good local SSRC.
932  if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
933    if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
934      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
935    } else {
936      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
937    }
938  }
939
940  for (size_t i = 0; i < recv_codecs_.size(); ++i) {
941    MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
942  }
943
944  for (size_t i = 0; i < recv_codecs_.size(); ++i) {
945    uint32 rtx_ssrc;
946    if (recv_codecs_[i].rtx_payload_type != -1 &&
947        sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
948      webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
949          config->rtp.rtx[recv_codecs_[i].codec.id];
950      rtx.ssrc = rtx_ssrc;
951      rtx.payload_type = recv_codecs_[i].rtx_payload_type;
952    }
953  }
954}
955
956bool WebRtcVideoChannel2::RemoveRecvStream(uint32 ssrc) {
957  LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
958  if (ssrc == 0) {
959    LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
960    return false;
961  }
962
963  rtc::CritScope stream_lock(&stream_crit_);
964  std::map<uint32, WebRtcVideoReceiveStream*>::iterator stream =
965      receive_streams_.find(ssrc);
966  if (stream == receive_streams_.end()) {
967    LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
968    return false;
969  }
970  delete stream->second;
971  receive_streams_.erase(stream);
972
973  return true;
974}
975
976bool WebRtcVideoChannel2::SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
977  LOG(LS_INFO) << "SetRenderer: ssrc:" << ssrc << " "
978               << (renderer ? "(ptr)" : "NULL");
979  if (ssrc == 0) {
980    default_unsignalled_ssrc_handler_.SetDefaultRenderer(this, renderer);
981    return true;
982  }
983
984  rtc::CritScope stream_lock(&stream_crit_);
985  std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
986      receive_streams_.find(ssrc);
987  if (it == receive_streams_.end()) {
988    return false;
989  }
990
991  it->second->SetRenderer(renderer);
992  return true;
993}
994
995bool WebRtcVideoChannel2::GetRenderer(uint32 ssrc, VideoRenderer** renderer) {
996  if (ssrc == 0) {
997    *renderer = default_unsignalled_ssrc_handler_.GetDefaultRenderer();
998    return *renderer != NULL;
999  }
1000
1001  rtc::CritScope stream_lock(&stream_crit_);
1002  std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
1003      receive_streams_.find(ssrc);
1004  if (it == receive_streams_.end()) {
1005    return false;
1006  }
1007  *renderer = it->second->GetRenderer();
1008  return true;
1009}
1010
1011bool WebRtcVideoChannel2::GetStats(const StatsOptions& options,
1012                                   VideoMediaInfo* info) {
1013  info->Clear();
1014  FillSenderStats(info);
1015  FillReceiverStats(info);
1016  webrtc::Call::Stats stats = call_->GetStats();
1017  FillBandwidthEstimationStats(stats, info);
1018  if (stats.rtt_ms != -1) {
1019    for (size_t i = 0; i < info->senders.size(); ++i) {
1020      info->senders[i].rtt_ms = stats.rtt_ms;
1021    }
1022  }
1023  return true;
1024}
1025
1026void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) {
1027  rtc::CritScope stream_lock(&stream_crit_);
1028  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1029           send_streams_.begin();
1030       it != send_streams_.end();
1031       ++it) {
1032    video_media_info->senders.push_back(it->second->GetVideoSenderInfo());
1033  }
1034}
1035
1036void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) {
1037  rtc::CritScope stream_lock(&stream_crit_);
1038  for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
1039           receive_streams_.begin();
1040       it != receive_streams_.end();
1041       ++it) {
1042    video_media_info->receivers.push_back(it->second->GetVideoReceiverInfo());
1043  }
1044}
1045
1046void WebRtcVideoChannel2::FillBandwidthEstimationStats(
1047    const webrtc::Call::Stats& stats,
1048    VideoMediaInfo* video_media_info) {
1049  BandwidthEstimationInfo bwe_info;
1050  bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
1051  bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
1052  bwe_info.bucket_delay = stats.pacer_delay_ms;
1053
1054  // Get send stream bitrate stats.
1055  rtc::CritScope stream_lock(&stream_crit_);
1056  for (std::map<uint32, WebRtcVideoSendStream*>::iterator stream =
1057           send_streams_.begin();
1058       stream != send_streams_.end();
1059       ++stream) {
1060    stream->second->FillBandwidthEstimationInfo(&bwe_info);
1061  }
1062  video_media_info->bw_estimations.push_back(bwe_info);
1063}
1064
1065bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
1066  LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
1067               << (capturer != NULL ? "(capturer)" : "NULL");
1068  assert(ssrc != 0);
1069  rtc::CritScope stream_lock(&stream_crit_);
1070  if (send_streams_.find(ssrc) == send_streams_.end()) {
1071    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
1072    return false;
1073  }
1074  return send_streams_[ssrc]->SetCapturer(capturer);
1075}
1076
1077bool WebRtcVideoChannel2::SendIntraFrame() {
1078  // TODO(pbos): Implement.
1079  LOG(LS_VERBOSE) << "SendIntraFrame().";
1080  return true;
1081}
1082
1083bool WebRtcVideoChannel2::RequestIntraFrame() {
1084  // TODO(pbos): Implement.
1085  LOG(LS_VERBOSE) << "SendIntraFrame().";
1086  return true;
1087}
1088
1089void WebRtcVideoChannel2::OnPacketReceived(
1090    rtc::Buffer* packet,
1091    const rtc::PacketTime& packet_time) {
1092  const webrtc::PacketReceiver::DeliveryStatus delivery_result =
1093      call_->Receiver()->DeliverPacket(
1094          reinterpret_cast<const uint8_t*>(packet->data()), packet->length());
1095  switch (delivery_result) {
1096    case webrtc::PacketReceiver::DELIVERY_OK:
1097      return;
1098    case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
1099      return;
1100    case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
1101      break;
1102  }
1103
1104  uint32 ssrc = 0;
1105  if (!GetRtpSsrc(packet->data(), packet->length(), &ssrc)) {
1106    return;
1107  }
1108
1109  // TODO(pbos): Make sure that the unsignalled SSRC uses the video payload.
1110  // Also figure out whether RTX needs to be handled.
1111  switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
1112    case UnsignalledSsrcHandler::kDropPacket:
1113      return;
1114    case UnsignalledSsrcHandler::kDeliverPacket:
1115      break;
1116  }
1117
1118  if (call_->Receiver()->DeliverPacket(
1119          reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) !=
1120      webrtc::PacketReceiver::DELIVERY_OK) {
1121    LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
1122    return;
1123  }
1124}
1125
1126void WebRtcVideoChannel2::OnRtcpReceived(
1127    rtc::Buffer* packet,
1128    const rtc::PacketTime& packet_time) {
1129  if (call_->Receiver()->DeliverPacket(
1130          reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) !=
1131      webrtc::PacketReceiver::DELIVERY_OK) {
1132    LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
1133  }
1134}
1135
1136void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
1137  LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
1138  call_->SignalNetworkState(ready ? webrtc::Call::kNetworkUp
1139                                  : webrtc::Call::kNetworkDown);
1140}
1141
1142bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
1143  LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
1144                  << (mute ? "mute" : "unmute");
1145  assert(ssrc != 0);
1146  rtc::CritScope stream_lock(&stream_crit_);
1147  if (send_streams_.find(ssrc) == send_streams_.end()) {
1148    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
1149    return false;
1150  }
1151
1152  send_streams_[ssrc]->MuteStream(mute);
1153  return true;
1154}
1155
1156bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
1157    const std::vector<RtpHeaderExtension>& extensions) {
1158  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
1159  LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
1160               << RtpExtensionsToString(extensions);
1161  if (!ValidateRtpHeaderExtensionIds(extensions))
1162    return false;
1163
1164  std::vector<webrtc::RtpExtension> filtered_extensions =
1165      FilterRtpExtensions(extensions);
1166  if (!RtpExtensionsHaveChanged(recv_rtp_extensions_, filtered_extensions))
1167    return true;
1168
1169  recv_rtp_extensions_ = filtered_extensions;
1170
1171  rtc::CritScope stream_lock(&stream_crit_);
1172  for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
1173           receive_streams_.begin();
1174       it != receive_streams_.end();
1175       ++it) {
1176    it->second->SetRtpExtensions(recv_rtp_extensions_);
1177  }
1178  return true;
1179}
1180
1181bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
1182    const std::vector<RtpHeaderExtension>& extensions) {
1183  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
1184  LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
1185               << RtpExtensionsToString(extensions);
1186  if (!ValidateRtpHeaderExtensionIds(extensions))
1187    return false;
1188
1189  std::vector<webrtc::RtpExtension> filtered_extensions =
1190      FilterRtpExtensions(extensions);
1191  if (!RtpExtensionsHaveChanged(send_rtp_extensions_, filtered_extensions))
1192    return true;
1193
1194  send_rtp_extensions_ = filtered_extensions;
1195
1196  rtc::CritScope stream_lock(&stream_crit_);
1197  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1198           send_streams_.begin();
1199       it != send_streams_.end();
1200       ++it) {
1201    it->second->SetRtpExtensions(send_rtp_extensions_);
1202  }
1203  return true;
1204}
1205
1206bool WebRtcVideoChannel2::SetMaxSendBandwidth(int max_bitrate_bps) {
1207  LOG(LS_INFO) << "SetMaxSendBandwidth: " << max_bitrate_bps << "bps.";
1208  if (max_bitrate_bps <= 0) {
1209    // Unsetting max bitrate.
1210    max_bitrate_bps = -1;
1211  }
1212  bitrate_config_.start_bitrate_bps = -1;
1213  bitrate_config_.max_bitrate_bps = max_bitrate_bps;
1214  if (max_bitrate_bps > 0 &&
1215      bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
1216    bitrate_config_.min_bitrate_bps = max_bitrate_bps;
1217  }
1218  call_->SetBitrateConfig(bitrate_config_);
1219  return true;
1220}
1221
1222bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
1223  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
1224  LOG(LS_INFO) << "SetOptions: " << options.ToString();
1225  VideoOptions old_options = options_;
1226  options_.SetAll(options);
1227  if (options_ == old_options) {
1228    // No new options to set.
1229    return true;
1230  }
1231  rtc::DiffServCodePoint dscp = options_.dscp.GetWithDefaultIfUnset(false)
1232                                    ? rtc::DSCP_AF41
1233                                    : rtc::DSCP_DEFAULT;
1234  MediaChannel::SetDscp(dscp);
1235  rtc::CritScope stream_lock(&stream_crit_);
1236  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1237           send_streams_.begin();
1238       it != send_streams_.end();
1239       ++it) {
1240    it->second->SetOptions(options_);
1241  }
1242  return true;
1243}
1244
1245void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
1246  MediaChannel::SetInterface(iface);
1247  // Set the RTP recv/send buffer to a bigger size
1248  MediaChannel::SetOption(NetworkInterface::ST_RTP,
1249                          rtc::Socket::OPT_RCVBUF,
1250                          kVideoRtpBufferSize);
1251
1252  // Speculative change to increase the outbound socket buffer size.
1253  // In b/15152257, we are seeing a significant number of packets discarded
1254  // due to lack of socket buffer space, although it's not yet clear what the
1255  // ideal value should be.
1256  MediaChannel::SetOption(NetworkInterface::ST_RTP,
1257                          rtc::Socket::OPT_SNDBUF,
1258                          kVideoRtpBufferSize);
1259}
1260
1261void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) {
1262  // TODO(pbos): Implement.
1263}
1264
1265void WebRtcVideoChannel2::OnMessage(rtc::Message* msg) {
1266  // Ignored.
1267}
1268
1269void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
1270  rtc::CritScope stream_lock(&stream_crit_);
1271  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1272           send_streams_.begin();
1273       it != send_streams_.end();
1274       ++it) {
1275    it->second->OnCpuResolutionRequest(load == kOveruse
1276                                           ? CoordinatedVideoAdapter::DOWNGRADE
1277                                           : CoordinatedVideoAdapter::UPGRADE);
1278  }
1279}
1280
1281bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, size_t len) {
1282  rtc::Buffer packet(data, len, kMaxRtpPacketLen);
1283  return MediaChannel::SendPacket(&packet);
1284}
1285
1286bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
1287  rtc::Buffer packet(data, len, kMaxRtpPacketLen);
1288  return MediaChannel::SendRtcp(&packet);
1289}
1290
1291void WebRtcVideoChannel2::StartAllSendStreams() {
1292  rtc::CritScope stream_lock(&stream_crit_);
1293  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1294           send_streams_.begin();
1295       it != send_streams_.end();
1296       ++it) {
1297    it->second->Start();
1298  }
1299}
1300
1301void WebRtcVideoChannel2::StopAllSendStreams() {
1302  rtc::CritScope stream_lock(&stream_crit_);
1303  for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
1304           send_streams_.begin();
1305       it != send_streams_.end();
1306       ++it) {
1307    it->second->Stop();
1308  }
1309}
1310
1311WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
1312    VideoSendStreamParameters(
1313        const webrtc::VideoSendStream::Config& config,
1314        const VideoOptions& options,
1315        const Settable<VideoCodecSettings>& codec_settings)
1316    : config(config), options(options), codec_settings(codec_settings) {
1317}
1318
1319WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
1320    webrtc::Call* call,
1321    WebRtcVideoEncoderFactory* external_encoder_factory,
1322    const VideoOptions& options,
1323    const Settable<VideoCodecSettings>& codec_settings,
1324    const StreamParams& sp,
1325    const std::vector<webrtc::RtpExtension>& rtp_extensions)
1326    : call_(call),
1327      external_encoder_factory_(external_encoder_factory),
1328      stream_(NULL),
1329      parameters_(webrtc::VideoSendStream::Config(), options, codec_settings),
1330      allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
1331      capturer_(NULL),
1332      sending_(false),
1333      muted_(false) {
1334  parameters_.config.rtp.max_packet_size = kVideoMtu;
1335
1336  sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
1337  sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
1338                 &parameters_.config.rtp.rtx.ssrcs);
1339  parameters_.config.rtp.c_name = sp.cname;
1340  parameters_.config.rtp.extensions = rtp_extensions;
1341
1342  VideoCodecSettings params;
1343  if (codec_settings.Get(&params)) {
1344    SetCodec(params);
1345  }
1346}
1347
1348WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
1349  DisconnectCapturer();
1350  if (stream_ != NULL) {
1351    call_->DestroyVideoSendStream(stream_);
1352  }
1353  DestroyVideoEncoder(&allocated_encoder_);
1354}
1355
1356static void CreateBlackFrame(webrtc::I420VideoFrame* video_frame,
1357                             int width,
1358                             int height) {
1359  video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
1360                                (width + 1) / 2);
1361  memset(video_frame->buffer(webrtc::kYPlane), 16,
1362         video_frame->allocated_size(webrtc::kYPlane));
1363  memset(video_frame->buffer(webrtc::kUPlane), 128,
1364         video_frame->allocated_size(webrtc::kUPlane));
1365  memset(video_frame->buffer(webrtc::kVPlane), 128,
1366         video_frame->allocated_size(webrtc::kVPlane));
1367}
1368
1369static void ConvertToI420VideoFrame(const VideoFrame& frame,
1370                                    webrtc::I420VideoFrame* i420_frame) {
1371  i420_frame->CreateFrame(
1372      static_cast<int>(frame.GetYPitch() * frame.GetHeight()),
1373      frame.GetYPlane(),
1374      static_cast<int>(frame.GetUPitch() * ((frame.GetHeight() + 1) / 2)),
1375      frame.GetUPlane(),
1376      static_cast<int>(frame.GetVPitch() * ((frame.GetHeight() + 1) / 2)),
1377      frame.GetVPlane(),
1378      static_cast<int>(frame.GetWidth()),
1379      static_cast<int>(frame.GetHeight()),
1380      static_cast<int>(frame.GetYPitch()),
1381      static_cast<int>(frame.GetUPitch()),
1382      static_cast<int>(frame.GetVPitch()));
1383}
1384
1385void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
1386    VideoCapturer* capturer,
1387    const VideoFrame* frame) {
1388  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame");
1389  LOG(LS_VERBOSE) << "InputFrame: " << frame->GetWidth() << "x"
1390                  << frame->GetHeight();
1391  // Lock before copying, can be called concurrently when swapping input source.
1392  rtc::CritScope frame_cs(&frame_lock_);
1393  ConvertToI420VideoFrame(*frame, &video_frame_);
1394
1395  rtc::CritScope cs(&lock_);
1396  if (stream_ == NULL) {
1397    LOG(LS_WARNING) << "Capturer inputting frames before send codecs are "
1398                       "configured, dropping.";
1399    return;
1400  }
1401
1402  // Not sending, abort early to prevent expensive reconfigurations while
1403  // setting up codecs etc.
1404  if (!sending_)
1405    return;
1406
1407  if (format_.width == 0) {  // Dropping frames.
1408    assert(format_.height == 0);
1409    LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
1410    return;
1411  }
1412  if (muted_) {
1413    // Create a black frame to transmit instead.
1414    CreateBlackFrame(&video_frame_,
1415                     static_cast<int>(frame->GetWidth()),
1416                     static_cast<int>(frame->GetHeight()));
1417  }
1418  // Reconfigure codec if necessary.
1419  SetDimensions(
1420      video_frame_.width(), video_frame_.height(), capturer->IsScreencast());
1421
1422  LOG(LS_VERBOSE) << "SwapFrame: " << video_frame_.width() << "x"
1423                  << video_frame_.height() << " -> (codec) "
1424                  << parameters_.encoder_config.streams.back().width << "x"
1425                  << parameters_.encoder_config.streams.back().height;
1426  stream_->Input()->SwapFrame(&video_frame_);
1427}
1428
1429bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
1430    VideoCapturer* capturer) {
1431  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
1432  if (!DisconnectCapturer() && capturer == NULL) {
1433    return false;
1434  }
1435
1436  {
1437    rtc::CritScope cs(&lock_);
1438
1439    if (capturer == NULL) {
1440      if (stream_ != NULL) {
1441        LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
1442        webrtc::I420VideoFrame black_frame;
1443
1444        CreateBlackFrame(&black_frame, last_dimensions_.width,
1445                         last_dimensions_.height);
1446        stream_->Input()->SwapFrame(&black_frame);
1447      }
1448
1449      capturer_ = NULL;
1450      return true;
1451    }
1452
1453    capturer_ = capturer;
1454  }
1455  // Lock cannot be held while connecting the capturer to prevent lock-order
1456  // violations.
1457  capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame);
1458  return true;
1459}
1460
1461bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat(
1462    const VideoFormat& format) {
1463  if ((format.width == 0 || format.height == 0) &&
1464      format.width != format.height) {
1465    LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not "
1466                     "both, 0x0 drops frames).";
1467    return false;
1468  }
1469
1470  rtc::CritScope cs(&lock_);
1471  if (format.width == 0 && format.height == 0) {
1472    LOG(LS_INFO)
1473        << "0x0 resolution selected. Captured frames will be dropped for ssrc: "
1474        << parameters_.config.rtp.ssrcs[0] << ".";
1475  } else {
1476    // TODO(pbos): Fix me, this only affects the last stream!
1477    parameters_.encoder_config.streams.back().max_framerate =
1478        VideoFormat::IntervalToFps(format.interval);
1479    SetDimensions(format.width, format.height, false);
1480  }
1481
1482  format_ = format;
1483  return true;
1484}
1485
1486void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
1487  rtc::CritScope cs(&lock_);
1488  muted_ = mute;
1489}
1490
1491bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
1492  cricket::VideoCapturer* capturer;
1493  {
1494    rtc::CritScope cs(&lock_);
1495    if (capturer_ == NULL) {
1496      return false;
1497    }
1498    capturer = capturer_;
1499    capturer_ = NULL;
1500  }
1501  capturer->SignalVideoFrame.disconnect(this);
1502  return true;
1503}
1504
1505void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
1506    const VideoOptions& options) {
1507  rtc::CritScope cs(&lock_);
1508  VideoCodecSettings codec_settings;
1509  if (parameters_.codec_settings.Get(&codec_settings)) {
1510    SetCodecAndOptions(codec_settings, options);
1511  } else {
1512    parameters_.options = options;
1513  }
1514}
1515
1516void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
1517    const VideoCodecSettings& codec_settings) {
1518  rtc::CritScope cs(&lock_);
1519  SetCodecAndOptions(codec_settings, parameters_.options);
1520}
1521
1522webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
1523  if (CodecNameMatches(name, kVp8CodecName)) {
1524    return webrtc::kVideoCodecVP8;
1525  } else if (CodecNameMatches(name, kVp9CodecName)) {
1526    return webrtc::kVideoCodecVP9;
1527  } else if (CodecNameMatches(name, kH264CodecName)) {
1528    return webrtc::kVideoCodecH264;
1529  }
1530  return webrtc::kVideoCodecUnknown;
1531}
1532
1533WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
1534WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
1535    const VideoCodec& codec) {
1536  webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
1537
1538  // Do not re-create encoders of the same type.
1539  if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) {
1540    return allocated_encoder_;
1541  }
1542
1543  if (external_encoder_factory_ != NULL) {
1544    webrtc::VideoEncoder* encoder =
1545        external_encoder_factory_->CreateVideoEncoder(type);
1546    if (encoder != NULL) {
1547      return AllocatedEncoder(encoder, type, true);
1548    }
1549  }
1550
1551  if (type == webrtc::kVideoCodecVP8) {
1552    return AllocatedEncoder(
1553        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false);
1554  } else if (type == webrtc::kVideoCodecVP9) {
1555    return AllocatedEncoder(
1556        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
1557  }
1558
1559  // This shouldn't happen, we should not be trying to create something we don't
1560  // support.
1561  assert(false);
1562  return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
1563}
1564
1565void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
1566    AllocatedEncoder* encoder) {
1567  if (encoder->external) {
1568    external_encoder_factory_->DestroyVideoEncoder(encoder->encoder);
1569  } else {
1570    delete encoder->encoder;
1571  }
1572}
1573
1574void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
1575    const VideoCodecSettings& codec_settings,
1576    const VideoOptions& options) {
1577  parameters_.encoder_config =
1578      CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
1579  if (parameters_.encoder_config.streams.empty())
1580    return;
1581
1582  format_ = VideoFormat(codec_settings.codec.width,
1583                        codec_settings.codec.height,
1584                        VideoFormat::FpsToInterval(30),
1585                        FOURCC_I420);
1586
1587  AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
1588  parameters_.config.encoder_settings.encoder = new_encoder.encoder;
1589  parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
1590  parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
1591  parameters_.config.rtp.fec = codec_settings.fec;
1592
1593  // Set RTX payload type if RTX is enabled.
1594  if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
1595    parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
1596  }
1597
1598  if (IsNackEnabled(codec_settings.codec)) {
1599    parameters_.config.rtp.nack.rtp_history_ms = kNackHistoryMs;
1600  }
1601
1602  options.suspend_below_min_bitrate.Get(
1603      &parameters_.config.suspend_below_min_bitrate);
1604
1605  parameters_.codec_settings.Set(codec_settings);
1606  parameters_.options = options;
1607
1608  RecreateWebRtcStream();
1609  if (allocated_encoder_.encoder != new_encoder.encoder) {
1610    DestroyVideoEncoder(&allocated_encoder_);
1611    allocated_encoder_ = new_encoder;
1612  }
1613}
1614
1615void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions(
1616    const std::vector<webrtc::RtpExtension>& rtp_extensions) {
1617  rtc::CritScope cs(&lock_);
1618  parameters_.config.rtp.extensions = rtp_extensions;
1619  RecreateWebRtcStream();
1620}
1621
1622webrtc::VideoEncoderConfig
1623WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
1624    const Dimensions& dimensions,
1625    const VideoCodec& codec) const {
1626  webrtc::VideoEncoderConfig encoder_config;
1627  if (dimensions.is_screencast) {
1628    int screencast_min_bitrate_kbps;
1629    parameters_.options.screencast_min_bitrate.Get(
1630        &screencast_min_bitrate_kbps);
1631    encoder_config.min_transmit_bitrate_bps =
1632        screencast_min_bitrate_kbps * 1000;
1633    encoder_config.content_type = webrtc::VideoEncoderConfig::kScreenshare;
1634  } else {
1635    encoder_config.min_transmit_bitrate_bps = 0;
1636    encoder_config.content_type = webrtc::VideoEncoderConfig::kRealtimeVideo;
1637  }
1638
1639  // Restrict dimensions according to codec max.
1640  int width = dimensions.width;
1641  int height = dimensions.height;
1642  if (!dimensions.is_screencast) {
1643    if (codec.width < width)
1644      width = codec.width;
1645    if (codec.height < height)
1646      height = codec.height;
1647  }
1648
1649  VideoCodec clamped_codec = codec;
1650  clamped_codec.width = width;
1651  clamped_codec.height = height;
1652
1653  encoder_config.streams = CreateVideoStreams(
1654      clamped_codec, parameters_.options, parameters_.config.rtp.ssrcs.size());
1655
1656  // Conference mode screencast uses 2 temporal layers split at 100kbit.
1657  if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) &&
1658      dimensions.is_screencast && encoder_config.streams.size() == 1) {
1659    ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
1660
1661    // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
1662    // on the VideoCodec struct as target and max bitrates, respectively.
1663    // See eg. webrtc::VP8EncoderImpl::SetRates().
1664    encoder_config.streams[0].target_bitrate_bps =
1665        config.tl0_bitrate_kbps * 1000;
1666    encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
1667    encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
1668    encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
1669        config.tl0_bitrate_kbps * 1000);
1670  }
1671  return encoder_config;
1672}
1673
1674void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
1675    int width,
1676    int height,
1677    bool is_screencast) {
1678  if (last_dimensions_.width == width && last_dimensions_.height == height &&
1679      last_dimensions_.is_screencast == is_screencast) {
1680    // Configured using the same parameters, do not reconfigure.
1681    return;
1682  }
1683  LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
1684               << (is_screencast ? " (screencast)" : " (not screencast)");
1685
1686  last_dimensions_.width = width;
1687  last_dimensions_.height = height;
1688  last_dimensions_.is_screencast = is_screencast;
1689
1690  assert(!parameters_.encoder_config.streams.empty());
1691
1692  VideoCodecSettings codec_settings;
1693  parameters_.codec_settings.Get(&codec_settings);
1694
1695  webrtc::VideoEncoderConfig encoder_config =
1696      CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
1697
1698  encoder_config.encoder_specific_settings =
1699      ConfigureVideoEncoderSettings(codec_settings.codec, parameters_.options);
1700
1701  bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
1702
1703  encoder_config.encoder_specific_settings = NULL;
1704
1705  if (!stream_reconfigured) {
1706    LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
1707                    << width << "x" << height;
1708    return;
1709  }
1710
1711  parameters_.encoder_config = encoder_config;
1712}
1713
1714void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
1715  rtc::CritScope cs(&lock_);
1716  assert(stream_ != NULL);
1717  stream_->Start();
1718  sending_ = true;
1719}
1720
1721void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
1722  rtc::CritScope cs(&lock_);
1723  if (stream_ != NULL) {
1724    stream_->Stop();
1725  }
1726  sending_ = false;
1727}
1728
1729VideoSenderInfo
1730WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
1731  VideoSenderInfo info;
1732  webrtc::VideoSendStream::Stats stats;
1733  {
1734    rtc::CritScope cs(&lock_);
1735    for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
1736      info.add_ssrc(ssrc);
1737
1738    for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
1739      if (i == parameters_.encoder_config.streams.size() - 1) {
1740        info.preferred_bitrate +=
1741            parameters_.encoder_config.streams[i].max_bitrate_bps;
1742      } else {
1743        info.preferred_bitrate +=
1744            parameters_.encoder_config.streams[i].target_bitrate_bps;
1745      }
1746    }
1747
1748    if (stream_ == NULL)
1749      return info;
1750
1751    stats = stream_->GetStats();
1752
1753    if (capturer_ != NULL && !capturer_->IsMuted()) {
1754      VideoFormat last_captured_frame_format;
1755      capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops,
1756                          &info.capturer_frame_time,
1757                          &last_captured_frame_format);
1758      info.input_frame_width = last_captured_frame_format.width;
1759      info.input_frame_height = last_captured_frame_format.height;
1760    }
1761  }
1762  info.framerate_input = stats.input_frame_rate;
1763  info.framerate_sent = stats.encode_frame_rate;
1764
1765  info.nominal_bitrate = stats.media_bitrate_bps;
1766
1767  info.send_frame_width = 0;
1768  info.send_frame_height = 0;
1769  for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
1770           stats.substreams.begin();
1771       it != stats.substreams.end(); ++it) {
1772    // TODO(pbos): Wire up additional stats, such as padding bytes.
1773    webrtc::VideoSendStream::StreamStats stream_stats = it->second;
1774    info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
1775                       stream_stats.rtp_stats.transmitted.header_bytes +
1776                       stream_stats.rtp_stats.transmitted.padding_bytes;
1777    info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
1778    info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
1779    if (stream_stats.width > info.send_frame_width)
1780      info.send_frame_width = stream_stats.width;
1781    if (stream_stats.height > info.send_frame_height)
1782      info.send_frame_height = stream_stats.height;
1783    info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
1784    info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
1785    info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
1786  }
1787
1788  if (!stats.substreams.empty()) {
1789    // TODO(pbos): Report fraction lost per SSRC.
1790    webrtc::VideoSendStream::StreamStats first_stream_stats =
1791        stats.substreams.begin()->second;
1792    info.fraction_lost =
1793        static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
1794        (1 << 8);
1795  }
1796
1797  return info;
1798}
1799
1800void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo(
1801    BandwidthEstimationInfo* bwe_info) {
1802  rtc::CritScope cs(&lock_);
1803  if (stream_ == NULL) {
1804    return;
1805  }
1806  webrtc::VideoSendStream::Stats stats = stream_->GetStats();
1807  for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
1808           stats.substreams.begin();
1809       it != stats.substreams.end(); ++it) {
1810    bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
1811    bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
1812  }
1813  bwe_info->actual_enc_bitrate += stats.media_bitrate_bps;
1814}
1815
1816void WebRtcVideoChannel2::WebRtcVideoSendStream::OnCpuResolutionRequest(
1817    CoordinatedVideoAdapter::AdaptRequest adapt_request) {
1818  rtc::CritScope cs(&lock_);
1819  bool adapt_cpu;
1820  parameters_.options.cpu_overuse_detection.Get(&adapt_cpu);
1821  if (!adapt_cpu) {
1822    return;
1823  }
1824  if (capturer_ == NULL || capturer_->video_adapter() == NULL) {
1825    return;
1826  }
1827
1828  capturer_->video_adapter()->OnCpuResolutionRequest(adapt_request);
1829}
1830
1831void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
1832  if (stream_ != NULL) {
1833    call_->DestroyVideoSendStream(stream_);
1834  }
1835
1836  VideoCodecSettings codec_settings;
1837  parameters_.codec_settings.Get(&codec_settings);
1838  parameters_.encoder_config.encoder_specific_settings =
1839      ConfigureVideoEncoderSettings(codec_settings.codec, parameters_.options);
1840
1841  stream_ = call_->CreateVideoSendStream(parameters_.config,
1842                                         parameters_.encoder_config);
1843
1844  parameters_.encoder_config.encoder_specific_settings = NULL;
1845
1846  if (sending_) {
1847    stream_->Start();
1848  }
1849}
1850
1851WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
1852    webrtc::Call* call,
1853    WebRtcVideoDecoderFactory* external_decoder_factory,
1854    const webrtc::VideoReceiveStream::Config& config,
1855    const std::vector<VideoCodecSettings>& recv_codecs)
1856    : call_(call),
1857      stream_(NULL),
1858      config_(config),
1859      external_decoder_factory_(external_decoder_factory),
1860      renderer_(NULL),
1861      last_width_(-1),
1862      last_height_(-1),
1863      first_frame_timestamp_(-1),
1864      estimated_remote_start_ntp_time_ms_(0) {
1865  config_.renderer = this;
1866  // SetRecvCodecs will also reset (start) the VideoReceiveStream.
1867  SetRecvCodecs(recv_codecs);
1868}
1869
1870WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
1871  call_->DestroyVideoReceiveStream(stream_);
1872  ClearDecoders(&allocated_decoders_);
1873}
1874
1875WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
1876WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
1877    std::vector<AllocatedDecoder>* old_decoders,
1878    const VideoCodec& codec) {
1879  webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
1880
1881  for (size_t i = 0; i < old_decoders->size(); ++i) {
1882    if ((*old_decoders)[i].type == type) {
1883      AllocatedDecoder decoder = (*old_decoders)[i];
1884      (*old_decoders)[i] = old_decoders->back();
1885      old_decoders->pop_back();
1886      return decoder;
1887    }
1888  }
1889
1890  if (external_decoder_factory_ != NULL) {
1891    webrtc::VideoDecoder* decoder =
1892        external_decoder_factory_->CreateVideoDecoder(type);
1893    if (decoder != NULL) {
1894      return AllocatedDecoder(decoder, type, true);
1895    }
1896  }
1897
1898  if (type == webrtc::kVideoCodecVP8) {
1899    return AllocatedDecoder(
1900        webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
1901  }
1902
1903  // This shouldn't happen, we should not be trying to create something we don't
1904  // support.
1905  assert(false);
1906  return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false);
1907}
1908
1909void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs(
1910    const std::vector<VideoCodecSettings>& recv_codecs) {
1911  std::vector<AllocatedDecoder> old_decoders = allocated_decoders_;
1912  allocated_decoders_.clear();
1913  config_.decoders.clear();
1914  for (size_t i = 0; i < recv_codecs.size(); ++i) {
1915    AllocatedDecoder allocated_decoder =
1916        CreateOrReuseVideoDecoder(&old_decoders, recv_codecs[i].codec);
1917    allocated_decoders_.push_back(allocated_decoder);
1918
1919    webrtc::VideoReceiveStream::Decoder decoder;
1920    decoder.decoder = allocated_decoder.decoder;
1921    decoder.payload_type = recv_codecs[i].codec.id;
1922    decoder.payload_name = recv_codecs[i].codec.name;
1923    config_.decoders.push_back(decoder);
1924  }
1925
1926  // TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
1927  config_.rtp.fec = recv_codecs.front().fec;
1928  config_.rtp.nack.rtp_history_ms =
1929      IsNackEnabled(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
1930  config_.rtp.remb = IsRembEnabled(recv_codecs.begin()->codec);
1931
1932  ClearDecoders(&old_decoders);
1933  RecreateWebRtcStream();
1934}
1935
1936void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRtpExtensions(
1937    const std::vector<webrtc::RtpExtension>& extensions) {
1938  config_.rtp.extensions = extensions;
1939  RecreateWebRtcStream();
1940}
1941
1942void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
1943  if (stream_ != NULL) {
1944    call_->DestroyVideoReceiveStream(stream_);
1945  }
1946  stream_ = call_->CreateVideoReceiveStream(config_);
1947  stream_->Start();
1948}
1949
1950void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
1951    std::vector<AllocatedDecoder>* allocated_decoders) {
1952  for (size_t i = 0; i < allocated_decoders->size(); ++i) {
1953    if ((*allocated_decoders)[i].external) {
1954      external_decoder_factory_->DestroyVideoDecoder(
1955          (*allocated_decoders)[i].decoder);
1956    } else {
1957      delete (*allocated_decoders)[i].decoder;
1958    }
1959  }
1960  allocated_decoders->clear();
1961}
1962
1963void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame(
1964    const webrtc::I420VideoFrame& frame,
1965    int time_to_render_ms) {
1966  rtc::CritScope crit(&renderer_lock_);
1967
1968  if (first_frame_timestamp_ < 0)
1969    first_frame_timestamp_ = frame.timestamp();
1970  int64_t rtp_time_elapsed_since_first_frame =
1971      (timestamp_wraparound_handler_.Unwrap(frame.timestamp()) -
1972       first_frame_timestamp_);
1973  int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame /
1974                            (cricket::kVideoCodecClockrate / 1000);
1975  if (frame.ntp_time_ms() > 0)
1976    estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
1977
1978  if (renderer_ == NULL) {
1979    LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoRenderer.";
1980    return;
1981  }
1982
1983  if (frame.width() != last_width_ || frame.height() != last_height_) {
1984    SetSize(frame.width(), frame.height());
1985  }
1986
1987  LOG(LS_VERBOSE) << "RenderFrame: (" << frame.width() << "x" << frame.height()
1988                  << ")";
1989
1990  const WebRtcVideoRenderFrame render_frame(&frame, elapsed_time_ms);
1991  renderer_->RenderFrame(&render_frame);
1992}
1993
1994bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsTextureSupported() const {
1995  return true;
1996}
1997
1998void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRenderer(
1999    cricket::VideoRenderer* renderer) {
2000  rtc::CritScope crit(&renderer_lock_);
2001  renderer_ = renderer;
2002  if (renderer_ != NULL && last_width_ != -1) {
2003    SetSize(last_width_, last_height_);
2004  }
2005}
2006
2007VideoRenderer* WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetRenderer() {
2008  // TODO(pbos): Remove GetRenderer and all uses of it, it's thread-unsafe by
2009  // design.
2010  rtc::CritScope crit(&renderer_lock_);
2011  return renderer_;
2012}
2013
2014void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSize(int width,
2015                                                            int height) {
2016  rtc::CritScope crit(&renderer_lock_);
2017  if (!renderer_->SetSize(width, height, 0)) {
2018    LOG(LS_ERROR) << "Could not set renderer size.";
2019  }
2020  last_width_ = width;
2021  last_height_ = height;
2022}
2023
2024VideoReceiverInfo
2025WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
2026  VideoReceiverInfo info;
2027  info.add_ssrc(config_.rtp.remote_ssrc);
2028  webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
2029  info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
2030                    stats.rtp_stats.transmitted.header_bytes +
2031                    stats.rtp_stats.transmitted.padding_bytes;
2032  info.packets_rcvd = stats.rtp_stats.transmitted.packets;
2033
2034  info.framerate_rcvd = stats.network_frame_rate;
2035  info.framerate_decoded = stats.decode_frame_rate;
2036  info.framerate_output = stats.render_frame_rate;
2037
2038  {
2039    rtc::CritScope frame_cs(&renderer_lock_);
2040    info.frame_width = last_width_;
2041    info.frame_height = last_height_;
2042    info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
2043  }
2044
2045  info.decode_ms = stats.decode_ms;
2046  info.max_decode_ms = stats.max_decode_ms;
2047  info.current_delay_ms = stats.current_delay_ms;
2048  info.target_delay_ms = stats.target_delay_ms;
2049  info.jitter_buffer_ms = stats.jitter_buffer_ms;
2050  info.min_playout_delay_ms = stats.min_playout_delay_ms;
2051  info.render_delay_ms = stats.render_delay_ms;
2052
2053  info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
2054  info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
2055  info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
2056
2057  return info;
2058}
2059
2060WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
2061    : rtx_payload_type(-1) {}
2062
2063bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
2064    const WebRtcVideoChannel2::VideoCodecSettings& other) const {
2065  return codec == other.codec &&
2066         fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
2067         fec.red_payload_type == other.fec.red_payload_type &&
2068         rtx_payload_type == other.rtx_payload_type;
2069}
2070
2071std::vector<WebRtcVideoChannel2::VideoCodecSettings>
2072WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
2073  assert(!codecs.empty());
2074
2075  std::vector<VideoCodecSettings> video_codecs;
2076  std::map<int, bool> payload_used;
2077  std::map<int, VideoCodec::CodecType> payload_codec_type;
2078  // |rtx_mapping| maps video payload type to rtx payload type.
2079  std::map<int, int> rtx_mapping;
2080
2081  webrtc::FecConfig fec_settings;
2082
2083  for (size_t i = 0; i < codecs.size(); ++i) {
2084    const VideoCodec& in_codec = codecs[i];
2085    int payload_type = in_codec.id;
2086
2087    if (payload_used[payload_type]) {
2088      LOG(LS_ERROR) << "Payload type already registered: "
2089                    << in_codec.ToString();
2090      return std::vector<VideoCodecSettings>();
2091    }
2092    payload_used[payload_type] = true;
2093    payload_codec_type[payload_type] = in_codec.GetCodecType();
2094
2095    switch (in_codec.GetCodecType()) {
2096      case VideoCodec::CODEC_RED: {
2097        // RED payload type, should not have duplicates.
2098        assert(fec_settings.red_payload_type == -1);
2099        fec_settings.red_payload_type = in_codec.id;
2100        continue;
2101      }
2102
2103      case VideoCodec::CODEC_ULPFEC: {
2104        // ULPFEC payload type, should not have duplicates.
2105        assert(fec_settings.ulpfec_payload_type == -1);
2106        fec_settings.ulpfec_payload_type = in_codec.id;
2107        continue;
2108      }
2109
2110      case VideoCodec::CODEC_RTX: {
2111        int associated_payload_type;
2112        if (!in_codec.GetParam(kCodecParamAssociatedPayloadType,
2113                               &associated_payload_type) ||
2114            !IsValidRtpPayloadType(associated_payload_type)) {
2115          LOG(LS_ERROR)
2116              << "RTX codec with invalid or no associated payload type: "
2117              << in_codec.ToString();
2118          return std::vector<VideoCodecSettings>();
2119        }
2120        rtx_mapping[associated_payload_type] = in_codec.id;
2121        continue;
2122      }
2123
2124      case VideoCodec::CODEC_VIDEO:
2125        break;
2126    }
2127
2128    video_codecs.push_back(VideoCodecSettings());
2129    video_codecs.back().codec = in_codec;
2130  }
2131
2132  // One of these codecs should have been a video codec. Only having FEC
2133  // parameters into this code is a logic error.
2134  assert(!video_codecs.empty());
2135
2136  for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
2137       it != rtx_mapping.end();
2138       ++it) {
2139    if (!payload_used[it->first]) {
2140      LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
2141      return std::vector<VideoCodecSettings>();
2142    }
2143    if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO) {
2144      LOG(LS_ERROR) << "RTX not mapped to regular video codec.";
2145      return std::vector<VideoCodecSettings>();
2146    }
2147  }
2148
2149  // TODO(pbos): Write tests that figure out that I have not verified that RTX
2150  // codecs aren't mapped to bogus payloads.
2151  for (size_t i = 0; i < video_codecs.size(); ++i) {
2152    video_codecs[i].fec = fec_settings;
2153    if (rtx_mapping[video_codecs[i].codec.id] != 0) {
2154      video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2155    }
2156  }
2157
2158  return video_codecs;
2159}
2160
2161}  // namespace cricket
2162
2163#endif  // HAVE_WEBRTC_VIDEO
2164