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| 1 /* | |
| 2 * Copyright 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/api/dtmfsender.h" | |
| 12 | |
| 13 #include <ctype.h> | |
| 14 | |
| 15 #include <string> | |
| 16 | |
| 17 #include "webrtc/base/logging.h" | |
| 18 #include "webrtc/base/thread.h" | |
| 19 | |
| 20 namespace webrtc { | |
| 21 | |
| 22 enum { | |
| 23 MSG_DO_INSERT_DTMF = 0, | |
| 24 }; | |
| 25 | |
| 26 // RFC4733 | |
| 27 // +-------+--------+------+---------+ | |
| 28 // | Event | Code | Type | Volume? | | |
| 29 // +-------+--------+------+---------+ | |
| 30 // | 0--9 | 0--9 | tone | yes | | |
| 31 // | * | 10 | tone | yes | | |
| 32 // | # | 11 | tone | yes | | |
| 33 // | A--D | 12--15 | tone | yes | | |
| 34 // +-------+--------+------+---------+ | |
| 35 // The "," is a special event defined by the WebRTC spec. It means to delay for | |
| 36 // 2 seconds before processing the next tone. We use -1 as its code. | |
| 37 static const int kDtmfCodeTwoSecondDelay = -1; | |
| 38 static const int kDtmfTwoSecondInMs = 2000; | |
| 39 static const char kDtmfValidTones[] = ",0123456789*#ABCDabcd"; | |
| 40 static const char kDtmfTonesTable[] = ",0123456789*#ABCD"; | |
| 41 // The duration cannot be more than 6000ms or less than 70ms. The gap between | |
| 42 // tones must be at least 50 ms. | |
| 43 static const int kDtmfDefaultDurationMs = 100; | |
| 44 static const int kDtmfMinDurationMs = 70; | |
| 45 static const int kDtmfMaxDurationMs = 6000; | |
| 46 static const int kDtmfDefaultGapMs = 50; | |
| 47 static const int kDtmfMinGapMs = 50; | |
| 48 | |
| 49 // Get DTMF code from the DTMF event character. | |
| 50 bool GetDtmfCode(char tone, int* code) { | |
| 51 // Convert a-d to A-D. | |
| 52 char event = toupper(tone); | |
| 53 const char* p = strchr(kDtmfTonesTable, event); | |
| 54 if (!p) { | |
| 55 return false; | |
| 56 } | |
| 57 *code = p - kDtmfTonesTable - 1; | |
| 58 return true; | |
| 59 } | |
| 60 | |
| 61 rtc::scoped_refptr<DtmfSender> DtmfSender::Create( | |
| 62 AudioTrackInterface* track, | |
| 63 rtc::Thread* signaling_thread, | |
| 64 DtmfProviderInterface* provider) { | |
| 65 if (!track || !signaling_thread) { | |
| 66 return NULL; | |
| 67 } | |
| 68 rtc::scoped_refptr<DtmfSender> dtmf_sender( | |
| 69 new rtc::RefCountedObject<DtmfSender>(track, signaling_thread, | |
| 70 provider)); | |
| 71 return dtmf_sender; | |
| 72 } | |
| 73 | |
| 74 DtmfSender::DtmfSender(AudioTrackInterface* track, | |
| 75 rtc::Thread* signaling_thread, | |
| 76 DtmfProviderInterface* provider) | |
| 77 : track_(track), | |
| 78 observer_(NULL), | |
| 79 signaling_thread_(signaling_thread), | |
| 80 provider_(provider), | |
| 81 duration_(kDtmfDefaultDurationMs), | |
| 82 inter_tone_gap_(kDtmfDefaultGapMs) { | |
| 83 ASSERT(track_ != NULL); | |
| 84 ASSERT(signaling_thread_ != NULL); | |
| 85 // TODO(deadbeef): Once we can use shared_ptr and weak_ptr, | |
| 86 // do that instead of relying on a "destroyed" signal. | |
| 87 if (provider_) { | |
| 88 ASSERT(provider_->GetOnDestroyedSignal() != NULL); | |
| 89 provider_->GetOnDestroyedSignal()->connect( | |
| 90 this, &DtmfSender::OnProviderDestroyed); | |
| 91 } | |
| 92 } | |
| 93 | |
| 94 DtmfSender::~DtmfSender() { | |
| 95 StopSending(); | |
| 96 } | |
| 97 | |
| 98 void DtmfSender::RegisterObserver(DtmfSenderObserverInterface* observer) { | |
| 99 observer_ = observer; | |
| 100 } | |
| 101 | |
| 102 void DtmfSender::UnregisterObserver() { | |
| 103 observer_ = NULL; | |
| 104 } | |
| 105 | |
| 106 bool DtmfSender::CanInsertDtmf() { | |
| 107 ASSERT(signaling_thread_->IsCurrent()); | |
| 108 if (!provider_) { | |
| 109 return false; | |
| 110 } | |
| 111 return provider_->CanInsertDtmf(track_->id()); | |
| 112 } | |
| 113 | |
| 114 bool DtmfSender::InsertDtmf(const std::string& tones, int duration, | |
| 115 int inter_tone_gap) { | |
| 116 ASSERT(signaling_thread_->IsCurrent()); | |
| 117 | |
| 118 if (duration > kDtmfMaxDurationMs || | |
| 119 duration < kDtmfMinDurationMs || | |
| 120 inter_tone_gap < kDtmfMinGapMs) { | |
| 121 LOG(LS_ERROR) << "InsertDtmf is called with invalid duration or tones gap. " | |
| 122 << "The duration cannot be more than " << kDtmfMaxDurationMs | |
| 123 << "ms or less than " << kDtmfMinDurationMs << "ms. " | |
| 124 << "The gap between tones must be at least " << kDtmfMinGapMs << "ms."; | |
| 125 return false; | |
| 126 } | |
| 127 | |
| 128 if (!CanInsertDtmf()) { | |
| 129 LOG(LS_ERROR) | |
| 130 << "InsertDtmf is called on DtmfSender that can't send DTMF."; | |
| 131 return false; | |
| 132 } | |
| 133 | |
| 134 tones_ = tones; | |
| 135 duration_ = duration; | |
| 136 inter_tone_gap_ = inter_tone_gap; | |
| 137 // Clear the previous queue. | |
| 138 signaling_thread_->Clear(this, MSG_DO_INSERT_DTMF); | |
| 139 // Kick off a new DTMF task queue. | |
| 140 signaling_thread_->Post(RTC_FROM_HERE, this, MSG_DO_INSERT_DTMF); | |
| 141 return true; | |
| 142 } | |
| 143 | |
| 144 const AudioTrackInterface* DtmfSender::track() const { | |
| 145 return track_; | |
| 146 } | |
| 147 | |
| 148 std::string DtmfSender::tones() const { | |
| 149 return tones_; | |
| 150 } | |
| 151 | |
| 152 int DtmfSender::duration() const { | |
| 153 return duration_; | |
| 154 } | |
| 155 | |
| 156 int DtmfSender::inter_tone_gap() const { | |
| 157 return inter_tone_gap_; | |
| 158 } | |
| 159 | |
| 160 void DtmfSender::OnMessage(rtc::Message* msg) { | |
| 161 switch (msg->message_id) { | |
| 162 case MSG_DO_INSERT_DTMF: { | |
| 163 DoInsertDtmf(); | |
| 164 break; | |
| 165 } | |
| 166 default: { | |
| 167 ASSERT(false); | |
| 168 break; | |
| 169 } | |
| 170 } | |
| 171 } | |
| 172 | |
| 173 void DtmfSender::DoInsertDtmf() { | |
| 174 ASSERT(signaling_thread_->IsCurrent()); | |
| 175 | |
| 176 // Get the first DTMF tone from the tone buffer. Unrecognized characters will | |
| 177 // be ignored and skipped. | |
| 178 size_t first_tone_pos = tones_.find_first_of(kDtmfValidTones); | |
| 179 int code = 0; | |
| 180 if (first_tone_pos == std::string::npos) { | |
| 181 tones_.clear(); | |
| 182 // Fire a “OnToneChange” event with an empty string and stop. | |
| 183 if (observer_) { | |
| 184 observer_->OnToneChange(std::string()); | |
| 185 } | |
| 186 return; | |
| 187 } else { | |
| 188 char tone = tones_[first_tone_pos]; | |
| 189 if (!GetDtmfCode(tone, &code)) { | |
| 190 // The find_first_of(kDtmfValidTones) should have guarantee |tone| is | |
| 191 // a valid DTMF tone. | |
| 192 ASSERT(false); | |
| 193 } | |
| 194 } | |
| 195 | |
| 196 int tone_gap = inter_tone_gap_; | |
| 197 if (code == kDtmfCodeTwoSecondDelay) { | |
| 198 // Special case defined by WebRTC - The character',' indicates a delay of 2 | |
| 199 // seconds before processing the next character in the tones parameter. | |
| 200 tone_gap = kDtmfTwoSecondInMs; | |
| 201 } else { | |
| 202 if (!provider_) { | |
| 203 LOG(LS_ERROR) << "The DtmfProvider has been destroyed."; | |
| 204 return; | |
| 205 } | |
| 206 // The provider starts playout of the given tone on the | |
| 207 // associated RTP media stream, using the appropriate codec. | |
| 208 if (!provider_->InsertDtmf(track_->id(), code, duration_)) { | |
| 209 LOG(LS_ERROR) << "The DtmfProvider can no longer send DTMF."; | |
| 210 return; | |
| 211 } | |
| 212 // Wait for the number of milliseconds specified by |duration_|. | |
| 213 tone_gap += duration_; | |
| 214 } | |
| 215 | |
| 216 // Fire a “OnToneChange” event with the tone that's just processed. | |
| 217 if (observer_) { | |
| 218 observer_->OnToneChange(tones_.substr(first_tone_pos, 1)); | |
| 219 } | |
| 220 | |
| 221 // Erase the unrecognized characters plus the tone that's just processed. | |
| 222 tones_.erase(0, first_tone_pos + 1); | |
| 223 | |
| 224 // Continue with the next tone. | |
| 225 signaling_thread_->PostDelayed(RTC_FROM_HERE, tone_gap, this, | |
| 226 MSG_DO_INSERT_DTMF); | |
| 227 } | |
| 228 | |
| 229 void DtmfSender::OnProviderDestroyed() { | |
| 230 LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue."; | |
| 231 StopSending(); | |
| 232 provider_ = NULL; | |
| 233 } | |
| 234 | |
| 235 void DtmfSender::StopSending() { | |
| 236 signaling_thread_->Clear(this); | |
| 237 } | |
| 238 | |
| 239 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
