rtp_receiver_impl.cc revision ff761fba8274d93bd73e76c8b8a1f2d0776dd840 
1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
12
13#include <assert.h>
14#include <math.h>
15#include <stdlib.h>
16#include <string.h>
17
18#include "webrtc/base/logging.h"
19#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
20#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
21#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
22
23namespace webrtc {
24
25using RtpUtility::GetCurrentRTP;
26using RtpUtility::Payload;
27using RtpUtility::StringCompare;
28
29RtpReceiver* RtpReceiver::CreateVideoReceiver(
30    Clock* clock,
31    RtpData* incoming_payload_callback,
32    RtpFeedback* incoming_messages_callback,
33    RTPPayloadRegistry* rtp_payload_registry) {
34  if (!incoming_payload_callback)
35    incoming_payload_callback = NullObjectRtpData();
36  if (!incoming_messages_callback)
37    incoming_messages_callback = NullObjectRtpFeedback();
38  return new RtpReceiverImpl(
39      clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
40      rtp_payload_registry,
41      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
42}
43
44RtpReceiver* RtpReceiver::CreateAudioReceiver(
45    Clock* clock,
46    RtpAudioFeedback* incoming_audio_feedback,
47    RtpData* incoming_payload_callback,
48    RtpFeedback* incoming_messages_callback,
49    RTPPayloadRegistry* rtp_payload_registry) {
50  if (!incoming_audio_feedback)
51    incoming_audio_feedback = NullObjectRtpAudioFeedback();
52  if (!incoming_payload_callback)
53    incoming_payload_callback = NullObjectRtpData();
54  if (!incoming_messages_callback)
55    incoming_messages_callback = NullObjectRtpFeedback();
56  return new RtpReceiverImpl(
57      clock, incoming_audio_feedback, incoming_messages_callback,
58      rtp_payload_registry,
59      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
60                                               incoming_audio_feedback));
61}
62
63RtpReceiverImpl::RtpReceiverImpl(
64    Clock* clock,
65    RtpAudioFeedback* incoming_audio_messages_callback,
66    RtpFeedback* incoming_messages_callback,
67    RTPPayloadRegistry* rtp_payload_registry,
68    RTPReceiverStrategy* rtp_media_receiver)
69    : clock_(clock),
70      rtp_payload_registry_(rtp_payload_registry),
71      rtp_media_receiver_(rtp_media_receiver),
72      cb_rtp_feedback_(incoming_messages_callback),
73      critical_section_rtp_receiver_(
74          CriticalSectionWrapper::CreateCriticalSection()),
75      last_receive_time_(0),
76      last_received_payload_length_(0),
77      ssrc_(0),
78      num_csrcs_(0),
79      current_remote_csrc_(),
80      last_received_timestamp_(0),
81      last_received_frame_time_ms_(-1),
82      last_received_sequence_number_(0),
83      nack_method_(kNackOff) {
84  assert(incoming_audio_messages_callback);
85  assert(incoming_messages_callback);
86
87  memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
88}
89
90RtpReceiverImpl::~RtpReceiverImpl() {
91  for (int i = 0; i < num_csrcs_; ++i) {
92    cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
93  }
94}
95
96int32_t RtpReceiverImpl::RegisterReceivePayload(
97    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
98    const int8_t payload_type,
99    const uint32_t frequency,
100    const uint8_t channels,
101    const uint32_t rate) {
102  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
103
104  // TODO(phoglund): Try to streamline handling of the RED codec and some other
105  // cases which makes it necessary to keep track of whether we created a
106  // payload or not.
107  bool created_new_payload = false;
108  int32_t result = rtp_payload_registry_->RegisterReceivePayload(
109      payload_name, payload_type, frequency, channels, rate,
110      &created_new_payload);
111  if (created_new_payload) {
112    if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
113                                                     frequency) != 0) {
114      LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
115                    << static_cast<int>(payload_type);
116      return -1;
117    }
118  }
119  return result;
120}
121
122int32_t RtpReceiverImpl::DeRegisterReceivePayload(
123    const int8_t payload_type) {
124  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
125  return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
126}
127
128NACKMethod RtpReceiverImpl::NACK() const {
129  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
130  return nack_method_;
131}
132
133// Turn negative acknowledgment requests on/off.
134void RtpReceiverImpl::SetNACKStatus(const NACKMethod method) {
135  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
136  nack_method_ = method;
137}
138
139uint32_t RtpReceiverImpl::SSRC() const {
140  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
141  return ssrc_;
142}
143
144// Get remote CSRC.
145int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
146  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
147
148  assert(num_csrcs_ <= kRtpCsrcSize);
149
150  if (num_csrcs_ > 0) {
151    memcpy(array_of_csrcs, current_remote_csrc_, sizeof(uint32_t)*num_csrcs_);
152  }
153  return num_csrcs_;
154}
155
156int32_t RtpReceiverImpl::Energy(
157    uint8_t array_of_energy[kRtpCsrcSize]) const {
158  return rtp_media_receiver_->Energy(array_of_energy);
159}
160
161bool RtpReceiverImpl::IncomingRtpPacket(
162  const RTPHeader& rtp_header,
163  const uint8_t* payload,
164  size_t payload_length,
165  PayloadUnion payload_specific,
166  bool in_order) {
167  // Trigger our callbacks.
168  CheckSSRCChanged(rtp_header);
169
170  int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
171  bool is_red = false;
172
173  if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
174                          &payload_specific) == -1) {
175    if (payload_length == 0) {
176      // OK, keep-alive packet.
177      return true;
178    }
179    LOG(LS_WARNING) << "Receiving invalid payload type.";
180    return false;
181  }
182
183  WebRtcRTPHeader webrtc_rtp_header;
184  memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
185  webrtc_rtp_header.header = rtp_header;
186  CheckCSRC(webrtc_rtp_header);
187
188  size_t payload_data_length = payload_length - rtp_header.paddingLength;
189
190  bool is_first_packet_in_frame = false;
191  {
192    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
193    if (HaveReceivedFrame()) {
194      is_first_packet_in_frame =
195          last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
196          last_received_timestamp_ != rtp_header.timestamp;
197    } else {
198      is_first_packet_in_frame = true;
199    }
200  }
201
202  int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
203      &webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
204      clock_->TimeInMilliseconds(), is_first_packet_in_frame);
205
206  if (ret_val < 0) {
207    return false;
208  }
209
210  {
211    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
212
213    last_receive_time_ = clock_->TimeInMilliseconds();
214    last_received_payload_length_ = payload_data_length;
215
216    if (in_order) {
217      if (last_received_timestamp_ != rtp_header.timestamp) {
218        last_received_timestamp_ = rtp_header.timestamp;
219        last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
220      }
221      last_received_sequence_number_ = rtp_header.sequenceNumber;
222    }
223  }
224  return true;
225}
226
227TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
228  return rtp_media_receiver_->GetTelephoneEventHandler();
229}
230
231bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
232  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
233  if (!HaveReceivedFrame())
234    return false;
235  *timestamp = last_received_timestamp_;
236  return true;
237}
238
239bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
240  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
241  if (!HaveReceivedFrame())
242    return false;
243  *receive_time_ms = last_received_frame_time_ms_;
244  return true;
245}
246
247bool RtpReceiverImpl::HaveReceivedFrame() const {
248  return last_received_frame_time_ms_ >= 0;
249}
250
251// Implementation note: must not hold critsect when called.
252void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
253  bool new_ssrc = false;
254  bool re_initialize_decoder = false;
255  char payload_name[RTP_PAYLOAD_NAME_SIZE];
256  uint8_t channels = 1;
257  uint32_t rate = 0;
258
259  {
260    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
261
262    int8_t last_received_payload_type =
263        rtp_payload_registry_->last_received_payload_type();
264    if (ssrc_ != rtp_header.ssrc ||
265        (last_received_payload_type == -1 && ssrc_ == 0)) {
266      // We need the payload_type_ to make the call if the remote SSRC is 0.
267      new_ssrc = true;
268
269      last_received_timestamp_ = 0;
270      last_received_sequence_number_ = 0;
271      last_received_frame_time_ms_ = -1;
272
273      // Do we have a SSRC? Then the stream is restarted.
274      if (ssrc_ != 0) {
275        // Do we have the same codec? Then re-initialize coder.
276        if (rtp_header.payloadType == last_received_payload_type) {
277          re_initialize_decoder = true;
278
279          Payload* payload;
280          if (!rtp_payload_registry_->PayloadTypeToPayload(
281              rtp_header.payloadType, payload)) {
282            return;
283          }
284          assert(payload);
285          payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
286          strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
287          if (payload->audio) {
288            channels = payload->typeSpecific.Audio.channels;
289            rate = payload->typeSpecific.Audio.rate;
290          }
291        }
292      }
293      ssrc_ = rtp_header.ssrc;
294    }
295  }
296
297  if (new_ssrc) {
298    // We need to get this to our RTCP sender and receiver.
299    // We need to do this outside critical section.
300    cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
301  }
302
303  if (re_initialize_decoder) {
304    if (-1 ==
305        cb_rtp_feedback_->OnInitializeDecoder(
306            rtp_header.payloadType, payload_name,
307            rtp_header.payload_type_frequency, channels, rate)) {
308      // New stream, same codec.
309      LOG(LS_ERROR) << "Failed to create decoder for payload type: "
310                    << static_cast<int>(rtp_header.payloadType);
311    }
312  }
313}
314
315// Implementation note: must not hold critsect when called.
316// TODO(phoglund): Move as much as possible of this code path into the media
317// specific receivers. Basically this method goes through a lot of trouble to
318// compute something which is only used by the media specific parts later. If
319// this code path moves we can get rid of some of the rtp_receiver ->
320// media_specific interface (such as CheckPayloadChange, possibly get/set
321// last known payload).
322int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
323                                             const int8_t first_payload_byte,
324                                             bool& is_red,
325                                             PayloadUnion* specific_payload) {
326  bool re_initialize_decoder = false;
327
328  char payload_name[RTP_PAYLOAD_NAME_SIZE];
329  int8_t payload_type = rtp_header.payloadType;
330
331  {
332    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
333
334    int8_t last_received_payload_type =
335        rtp_payload_registry_->last_received_payload_type();
336    // TODO(holmer): Remove this code when RED parsing has been broken out from
337    // RtpReceiverAudio.
338    if (payload_type != last_received_payload_type) {
339      if (rtp_payload_registry_->red_payload_type() == payload_type) {
340        // Get the real codec payload type.
341        payload_type = first_payload_byte & 0x7f;
342        is_red = true;
343
344        if (rtp_payload_registry_->red_payload_type() == payload_type) {
345          // Invalid payload type, traced by caller. If we proceeded here,
346          // this would be set as |_last_received_payload_type|, and we would no
347          // longer catch corrupt packets at this level.
348          return -1;
349        }
350
351        // When we receive RED we need to check the real payload type.
352        if (payload_type == last_received_payload_type) {
353          rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
354          return 0;
355        }
356      }
357      bool should_discard_changes = false;
358
359      rtp_media_receiver_->CheckPayloadChanged(
360        payload_type, specific_payload,
361        &should_discard_changes);
362
363      if (should_discard_changes) {
364        is_red = false;
365        return 0;
366      }
367
368      Payload* payload;
369      if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
370        // Not a registered payload type.
371        return -1;
372      }
373      assert(payload);
374      payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
375      strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
376
377      rtp_payload_registry_->set_last_received_payload_type(payload_type);
378
379      re_initialize_decoder = true;
380
381      rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
382      rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
383
384      if (!payload->audio) {
385        bool media_type_unchanged =
386            rtp_payload_registry_->ReportMediaPayloadType(payload_type);
387        if (media_type_unchanged) {
388          // Only reset the decoder if the media codec type has changed.
389          re_initialize_decoder = false;
390        }
391      }
392    } else {
393      rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
394      is_red = false;
395    }
396  }  // End critsect.
397
398  if (re_initialize_decoder) {
399    if (-1 ==
400        rtp_media_receiver_->InvokeOnInitializeDecoder(
401            cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
402      return -1;  // Wrong payload type.
403    }
404  }
405  return 0;
406}
407
408// Implementation note: must not hold critsect when called.
409void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
410  int32_t num_csrcs_diff = 0;
411  uint32_t old_remote_csrc[kRtpCsrcSize];
412  uint8_t old_num_csrcs = 0;
413
414  {
415    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
416
417    if (!rtp_media_receiver_->ShouldReportCsrcChanges(
418        rtp_header.header.payloadType)) {
419      return;
420    }
421    old_num_csrcs  = num_csrcs_;
422    if (old_num_csrcs > 0) {
423      // Make a copy of old.
424      memcpy(old_remote_csrc, current_remote_csrc_,
425             num_csrcs_ * sizeof(uint32_t));
426    }
427    const uint8_t num_csrcs = rtp_header.header.numCSRCs;
428    if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
429      // Copy new.
430      memcpy(current_remote_csrc_,
431             rtp_header.header.arrOfCSRCs,
432             num_csrcs * sizeof(uint32_t));
433    }
434    if (num_csrcs > 0 || old_num_csrcs > 0) {
435      num_csrcs_diff = num_csrcs - old_num_csrcs;
436      num_csrcs_ = num_csrcs;  // Update stored CSRCs.
437    } else {
438      // No change.
439      return;
440    }
441  }  // End critsect.
442
443  bool have_called_callback = false;
444  // Search for new CSRC in old array.
445  for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
446    const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];
447
448    bool found_match = false;
449    for (uint8_t j = 0; j < old_num_csrcs; ++j) {
450      if (csrc == old_remote_csrc[j]) {  // old list
451        found_match = true;
452        break;
453      }
454    }
455    if (!found_match && csrc) {
456      // Didn't find it, report it as new.
457      have_called_callback = true;
458      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
459    }
460  }
461  // Search for old CSRC in new array.
462  for (uint8_t i = 0; i < old_num_csrcs; ++i) {
463    const uint32_t csrc = old_remote_csrc[i];
464
465    bool found_match = false;
466    for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
467      if (csrc == rtp_header.header.arrOfCSRCs[j]) {
468        found_match = true;
469        break;
470      }
471    }
472    if (!found_match && csrc) {
473      // Did not find it, report as removed.
474      have_called_callback = true;
475      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
476    }
477  }
478  if (!have_called_callback) {
479    // If the CSRC list contain non-unique entries we will end up here.
480    // Using CSRC 0 to signal this event, not interop safe, other
481    // implementations might have CSRC 0 as a valid value.
482    if (num_csrcs_diff > 0) {
483      cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
484    } else if (num_csrcs_diff < 0) {
485      cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
486    }
487  }
488}
489
490}  // namespace webrtc
491