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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 |
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43 inband_vad_active_(false), | 43 inband_vad_active_(false), |
44 cngnb_payload_type_(-1), | 44 cngnb_payload_type_(-1), |
45 cngwb_payload_type_(-1), | 45 cngwb_payload_type_(-1), |
46 cngswb_payload_type_(-1), | 46 cngswb_payload_type_(-1), |
47 cngfb_payload_type_(-1), | 47 cngfb_payload_type_(-1), |
48 last_payload_type_(-1), | 48 last_payload_type_(-1), |
49 audio_level_dbov_(0) {} | 49 audio_level_dbov_(0) {} |
50 | 50 |
51 RTPSenderAudio::~RTPSenderAudio() {} | 51 RTPSenderAudio::~RTPSenderAudio() {} |
52 | 52 |
53 int RTPSenderAudio::AudioFrequency() const { | |
54 return kDtmfFrequencyHz; | |
55 } | |
56 | |
57 // set audio packet size, used to determine when it's time to send a DTMF packet | 53 // set audio packet size, used to determine when it's time to send a DTMF packet |
58 // in silence (CNG) | 54 // in silence (CNG) |
59 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) { | 55 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) { |
60 rtc::CritScope cs(&send_audio_critsect_); | 56 rtc::CritScope cs(&send_audio_critsect_); |
61 packet_size_samples_ = packet_size_samples; | 57 packet_size_samples_ = packet_size_samples; |
62 return 0; | 58 return 0; |
63 } | 59 } |
64 | 60 |
65 int32_t RTPSenderAudio::RegisterAudioPayload( | 61 int32_t RTPSenderAudio::RegisterAudioPayload( |
66 const char payloadName[RTP_PAYLOAD_NAME_SIZE], | 62 const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
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365 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent", | 361 TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent", |
366 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber()); | 362 "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber()); |
367 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission, | 363 result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission, |
368 RtpPacketSender::kHighPriority); | 364 RtpPacketSender::kHighPriority); |
369 send_count--; | 365 send_count--; |
370 } while (send_count > 0 && result); | 366 } while (send_count > 0 && result); |
371 | 367 |
372 return result; | 368 return result; |
373 } | 369 } |
374 } // namespace webrtc | 370 } // namespace webrtc |
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