Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(78)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Issue 2685783014: Replace NULL with nullptr in all C++ files. (Closed)
Patch Set: Fixing android. Created 3 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 190 matching lines...) Expand 10 before | Expand all | Expand 10 after
201 dtmf_duration_samples, 201 dtmf_duration_samples,
202 !dtmf_event_first_packet_sent_)) { 202 !dtmf_event_first_packet_sent_)) {
203 return false; 203 return false;
204 } 204 }
205 dtmf_event_first_packet_sent_ = true; 205 dtmf_event_first_packet_sent_ = true;
206 return true; 206 return true;
207 } 207 }
208 } 208 }
209 return true; 209 return true;
210 } 210 }
211 if (payload_size == 0 || payload_data == NULL) { 211 if (payload_size == 0 || payload_data == nullptr) {
212 if (frame_type == kEmptyFrame) { 212 if (frame_type == kEmptyFrame) {
213 // we don't send empty audio RTP packets 213 // we don't send empty audio RTP packets
214 // no error since we use it to drive DTMF when we use VAD 214 // no error since we use it to drive DTMF when we use VAD
215 return true; 215 return true;
216 } 216 }
217 return false; 217 return false;
218 } 218 }
219 219
220 std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket(); 220 std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket();
221 packet->SetMarker(MarkerBit(frame_type, payload_type)); 221 packet->SetMarker(MarkerBit(frame_type, payload_type));
(...skipping 118 matching lines...) Expand 10 before | Expand all | Expand 10 after
340 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent", 340 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
341 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber()); 341 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber());
342 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission, 342 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
343 RtpPacketSender::kHighPriority); 343 RtpPacketSender::kHighPriority);
344 send_count--; 344 send_count--;
345 } while (send_count > 0 && result); 345 } while (send_count > 0 && result);
346 346
347 return result; 347 return result;
348 } 348 }
349 } // namespace webrtc 349 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698