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1 /* | |
2 * libjingle | |
3 * Copyright 2010 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 #ifndef TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | |
29 #define TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | |
30 | |
31 #include <list> | |
32 #include <map> | |
33 #include <vector> | |
34 | |
35 #include "talk/media/base/codec.h" | |
36 #include "talk/media/base/rtputils.h" | |
37 #include "talk/media/webrtc/fakewebrtccommon.h" | |
38 #include "talk/media/webrtc/webrtcvoe.h" | |
39 #include "webrtc/base/basictypes.h" | |
40 #include "webrtc/base/checks.h" | |
41 #include "webrtc/base/gunit.h" | |
42 #include "webrtc/base/stringutils.h" | |
43 #include "webrtc/config.h" | |
44 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | |
45 #include "webrtc/modules/audio_processing/include/audio_processing.h" | |
46 | |
47 namespace cricket { | |
48 | |
49 static const int kOpusBandwidthNb = 4000; | |
50 static const int kOpusBandwidthMb = 6000; | |
51 static const int kOpusBandwidthWb = 8000; | |
52 static const int kOpusBandwidthSwb = 12000; | |
53 static const int kOpusBandwidthFb = 20000; | |
54 | |
55 #define WEBRTC_CHECK_CHANNEL(channel) \ | |
56 if (channels_.find(channel) == channels_.end()) return -1; | |
57 | |
58 class FakeAudioProcessing : public webrtc::AudioProcessing { | |
59 public: | |
60 FakeAudioProcessing() : experimental_ns_enabled_(false) {} | |
61 | |
62 WEBRTC_STUB(Initialize, ()) | |
63 WEBRTC_STUB(Initialize, ( | |
64 int input_sample_rate_hz, | |
65 int output_sample_rate_hz, | |
66 int reverse_sample_rate_hz, | |
67 webrtc::AudioProcessing::ChannelLayout input_layout, | |
68 webrtc::AudioProcessing::ChannelLayout output_layout, | |
69 webrtc::AudioProcessing::ChannelLayout reverse_layout)); | |
70 WEBRTC_STUB(Initialize, ( | |
71 const webrtc::ProcessingConfig& processing_config)); | |
72 | |
73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { | |
74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; | |
75 } | |
76 | |
77 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); | |
78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); | |
79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); | |
80 size_t num_input_channels() const override { return 0; } | |
81 size_t num_proc_channels() const override { return 0; } | |
82 size_t num_output_channels() const override { return 0; } | |
83 size_t num_reverse_channels() const override { return 0; } | |
84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); | |
85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); | |
86 WEBRTC_STUB(ProcessStream, ( | |
87 const float* const* src, | |
88 size_t samples_per_channel, | |
89 int input_sample_rate_hz, | |
90 webrtc::AudioProcessing::ChannelLayout input_layout, | |
91 int output_sample_rate_hz, | |
92 webrtc::AudioProcessing::ChannelLayout output_layout, | |
93 float* const* dest)); | |
94 WEBRTC_STUB(ProcessStream, | |
95 (const float* const* src, | |
96 const webrtc::StreamConfig& input_config, | |
97 const webrtc::StreamConfig& output_config, | |
98 float* const* dest)); | |
99 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); | |
100 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); | |
101 WEBRTC_STUB(AnalyzeReverseStream, ( | |
102 const float* const* data, | |
103 size_t samples_per_channel, | |
104 int sample_rate_hz, | |
105 webrtc::AudioProcessing::ChannelLayout layout)); | |
106 WEBRTC_STUB(ProcessReverseStream, | |
107 (const float* const* src, | |
108 const webrtc::StreamConfig& reverse_input_config, | |
109 const webrtc::StreamConfig& reverse_output_config, | |
110 float* const* dest)); | |
111 WEBRTC_STUB(set_stream_delay_ms, (int delay)); | |
112 WEBRTC_STUB_CONST(stream_delay_ms, ()); | |
113 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | |
114 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); | |
115 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); | |
116 WEBRTC_STUB_CONST(delay_offset_ms, ()); | |
117 WEBRTC_STUB(StartDebugRecording, | |
118 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); | |
119 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); | |
120 WEBRTC_STUB(StopDebugRecording, ()); | |
121 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | |
122 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | |
123 webrtc::EchoControlMobile* echo_control_mobile() const override { | |
124 return NULL; | |
125 } | |
126 webrtc::GainControl* gain_control() const override { return NULL; } | |
127 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | |
128 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | |
129 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | |
130 webrtc::VoiceDetection* voice_detection() const override { return NULL; } | |
131 | |
132 bool experimental_ns_enabled() { | |
133 return experimental_ns_enabled_; | |
134 } | |
135 | |
136 private: | |
137 bool experimental_ns_enabled_; | |
138 }; | |
139 | |
140 class FakeWebRtcVoiceEngine | |
141 : public webrtc::VoEAudioProcessing, | |
142 public webrtc::VoEBase, public webrtc::VoECodec, | |
143 public webrtc::VoEHardware, | |
144 public webrtc::VoENetwork, public webrtc::VoERTP_RTCP, | |
145 public webrtc::VoEVolumeControl { | |
146 public: | |
147 struct Channel { | |
148 explicit Channel() | |
149 : external_transport(false), | |
150 send(false), | |
151 playout(false), | |
152 volume_scale(1.0), | |
153 vad(false), | |
154 codec_fec(false), | |
155 max_encoding_bandwidth(0), | |
156 opus_dtx(false), | |
157 red(false), | |
158 nack(false), | |
159 cn8_type(13), | |
160 cn16_type(105), | |
161 red_type(117), | |
162 nack_max_packets(0), | |
163 send_ssrc(0), | |
164 associate_send_channel(-1), | |
165 recv_codecs(), | |
166 neteq_capacity(-1), | |
167 neteq_fast_accelerate(false) { | |
168 memset(&send_codec, 0, sizeof(send_codec)); | |
169 } | |
170 bool external_transport; | |
171 bool send; | |
172 bool playout; | |
173 float volume_scale; | |
174 bool vad; | |
175 bool codec_fec; | |
176 int max_encoding_bandwidth; | |
177 bool opus_dtx; | |
178 bool red; | |
179 bool nack; | |
180 int cn8_type; | |
181 int cn16_type; | |
182 int red_type; | |
183 int nack_max_packets; | |
184 uint32_t send_ssrc; | |
185 int associate_send_channel; | |
186 std::vector<webrtc::CodecInst> recv_codecs; | |
187 webrtc::CodecInst send_codec; | |
188 webrtc::PacketTime last_rtp_packet_time; | |
189 std::list<std::string> packets; | |
190 int neteq_capacity; | |
191 bool neteq_fast_accelerate; | |
192 }; | |
193 | |
194 FakeWebRtcVoiceEngine() | |
195 : inited_(false), | |
196 last_channel_(-1), | |
197 fail_create_channel_(false), | |
198 num_set_send_codecs_(0), | |
199 ec_enabled_(false), | |
200 ec_metrics_enabled_(false), | |
201 cng_enabled_(false), | |
202 ns_enabled_(false), | |
203 agc_enabled_(false), | |
204 highpass_filter_enabled_(false), | |
205 stereo_swapping_enabled_(false), | |
206 typing_detection_enabled_(false), | |
207 ec_mode_(webrtc::kEcDefault), | |
208 aecm_mode_(webrtc::kAecmSpeakerphone), | |
209 ns_mode_(webrtc::kNsDefault), | |
210 agc_mode_(webrtc::kAgcDefault), | |
211 observer_(NULL), | |
212 playout_fail_channel_(-1), | |
213 send_fail_channel_(-1), | |
214 recording_sample_rate_(-1), | |
215 playout_sample_rate_(-1) { | |
216 memset(&agc_config_, 0, sizeof(agc_config_)); | |
217 } | |
218 ~FakeWebRtcVoiceEngine() { | |
219 RTC_CHECK(channels_.empty()); | |
220 } | |
221 | |
222 bool ec_metrics_enabled() const { return ec_metrics_enabled_; } | |
223 | |
224 bool IsInited() const { return inited_; } | |
225 int GetLastChannel() const { return last_channel_; } | |
226 int GetNumChannels() const { return static_cast<int>(channels_.size()); } | |
227 uint32_t GetLocalSSRC(int channel) { | |
228 return channels_[channel]->send_ssrc; | |
229 } | |
230 bool GetPlayout(int channel) { | |
231 return channels_[channel]->playout; | |
232 } | |
233 bool GetSend(int channel) { | |
234 return channels_[channel]->send; | |
235 } | |
236 bool GetVAD(int channel) { | |
237 return channels_[channel]->vad; | |
238 } | |
239 bool GetOpusDtx(int channel) { | |
240 return channels_[channel]->opus_dtx; | |
241 } | |
242 bool GetRED(int channel) { | |
243 return channels_[channel]->red; | |
244 } | |
245 bool GetCodecFEC(int channel) { | |
246 return channels_[channel]->codec_fec; | |
247 } | |
248 int GetMaxEncodingBandwidth(int channel) { | |
249 return channels_[channel]->max_encoding_bandwidth; | |
250 } | |
251 bool GetNACK(int channel) { | |
252 return channels_[channel]->nack; | |
253 } | |
254 int GetNACKMaxPackets(int channel) { | |
255 return channels_[channel]->nack_max_packets; | |
256 } | |
257 const webrtc::PacketTime& GetLastRtpPacketTime(int channel) { | |
258 RTC_DCHECK(channels_.find(channel) != channels_.end()); | |
259 return channels_[channel]->last_rtp_packet_time; | |
260 } | |
261 int GetSendCNPayloadType(int channel, bool wideband) { | |
262 return (wideband) ? | |
263 channels_[channel]->cn16_type : | |
264 channels_[channel]->cn8_type; | |
265 } | |
266 int GetSendREDPayloadType(int channel) { | |
267 return channels_[channel]->red_type; | |
268 } | |
269 bool CheckPacket(int channel, const void* data, size_t len) { | |
270 bool result = !CheckNoPacket(channel); | |
271 if (result) { | |
272 std::string packet = channels_[channel]->packets.front(); | |
273 result = (packet == std::string(static_cast<const char*>(data), len)); | |
274 channels_[channel]->packets.pop_front(); | |
275 } | |
276 return result; | |
277 } | |
278 bool CheckNoPacket(int channel) { | |
279 return channels_[channel]->packets.empty(); | |
280 } | |
281 void TriggerCallbackOnError(int channel_num, int err_code) { | |
282 RTC_DCHECK(observer_ != NULL); | |
283 observer_->CallbackOnError(channel_num, err_code); | |
284 } | |
285 void set_playout_fail_channel(int channel) { | |
286 playout_fail_channel_ = channel; | |
287 } | |
288 void set_send_fail_channel(int channel) { | |
289 send_fail_channel_ = channel; | |
290 } | |
291 void set_fail_create_channel(bool fail_create_channel) { | |
292 fail_create_channel_ = fail_create_channel; | |
293 } | |
294 int AddChannel(const webrtc::Config& config) { | |
295 if (fail_create_channel_) { | |
296 return -1; | |
297 } | |
298 Channel* ch = new Channel(); | |
299 auto db = webrtc::acm2::RentACodec::Database(); | |
300 ch->recv_codecs.assign(db.begin(), db.end()); | |
301 if (config.Get<webrtc::NetEqCapacityConfig>().enabled) { | |
302 ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity; | |
303 } | |
304 ch->neteq_fast_accelerate = | |
305 config.Get<webrtc::NetEqFastAccelerate>().enabled; | |
306 channels_[++last_channel_] = ch; | |
307 return last_channel_; | |
308 } | |
309 | |
310 int GetNumSetSendCodecs() const { return num_set_send_codecs_; } | |
311 | |
312 int GetAssociateSendChannel(int channel) { | |
313 return channels_[channel]->associate_send_channel; | |
314 } | |
315 | |
316 WEBRTC_STUB(Release, ()); | |
317 | |
318 // webrtc::VoEBase | |
319 WEBRTC_FUNC(RegisterVoiceEngineObserver, ( | |
320 webrtc::VoiceEngineObserver& observer)) { | |
321 observer_ = &observer; | |
322 return 0; | |
323 } | |
324 WEBRTC_STUB(DeRegisterVoiceEngineObserver, ()); | |
325 WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm, | |
326 webrtc::AudioProcessing* audioproc)) { | |
327 inited_ = true; | |
328 return 0; | |
329 } | |
330 WEBRTC_FUNC(Terminate, ()) { | |
331 inited_ = false; | |
332 return 0; | |
333 } | |
334 webrtc::AudioProcessing* audio_processing() override { | |
335 return &audio_processing_; | |
336 } | |
337 WEBRTC_FUNC(CreateChannel, ()) { | |
338 webrtc::Config empty_config; | |
339 return AddChannel(empty_config); | |
340 } | |
341 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) { | |
342 return AddChannel(config); | |
343 } | |
344 WEBRTC_FUNC(DeleteChannel, (int channel)) { | |
345 WEBRTC_CHECK_CHANNEL(channel); | |
346 for (const auto& ch : channels_) { | |
347 if (ch.second->associate_send_channel == channel) { | |
348 ch.second->associate_send_channel = -1; | |
349 } | |
350 } | |
351 delete channels_[channel]; | |
352 channels_.erase(channel); | |
353 return 0; | |
354 } | |
355 WEBRTC_STUB(StartReceive, (int channel)); | |
356 WEBRTC_FUNC(StartPlayout, (int channel)) { | |
357 if (playout_fail_channel_ != channel) { | |
358 WEBRTC_CHECK_CHANNEL(channel); | |
359 channels_[channel]->playout = true; | |
360 return 0; | |
361 } else { | |
362 // When playout_fail_channel_ == channel, fail the StartPlayout on this | |
363 // channel. | |
364 return -1; | |
365 } | |
366 } | |
367 WEBRTC_FUNC(StartSend, (int channel)) { | |
368 if (send_fail_channel_ != channel) { | |
369 WEBRTC_CHECK_CHANNEL(channel); | |
370 channels_[channel]->send = true; | |
371 return 0; | |
372 } else { | |
373 // When send_fail_channel_ == channel, fail the StartSend on this | |
374 // channel. | |
375 return -1; | |
376 } | |
377 } | |
378 WEBRTC_STUB(StopReceive, (int channel)); | |
379 WEBRTC_FUNC(StopPlayout, (int channel)) { | |
380 WEBRTC_CHECK_CHANNEL(channel); | |
381 channels_[channel]->playout = false; | |
382 return 0; | |
383 } | |
384 WEBRTC_FUNC(StopSend, (int channel)) { | |
385 WEBRTC_CHECK_CHANNEL(channel); | |
386 channels_[channel]->send = false; | |
387 return 0; | |
388 } | |
389 WEBRTC_STUB(GetVersion, (char version[1024])); | |
390 WEBRTC_STUB(LastError, ()); | |
391 WEBRTC_FUNC(AssociateSendChannel, (int channel, | |
392 int accociate_send_channel)) { | |
393 WEBRTC_CHECK_CHANNEL(channel); | |
394 channels_[channel]->associate_send_channel = accociate_send_channel; | |
395 return 0; | |
396 } | |
397 webrtc::RtcEventLog* GetEventLog() { return nullptr; } | |
398 | |
399 // webrtc::VoECodec | |
400 WEBRTC_STUB(NumOfCodecs, ()); | |
401 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); | |
402 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { | |
403 WEBRTC_CHECK_CHANNEL(channel); | |
404 // To match the behavior of the real implementation. | |
405 if (_stricmp(codec.plname, "telephone-event") == 0 || | |
406 _stricmp(codec.plname, "audio/telephone-event") == 0 || | |
407 _stricmp(codec.plname, "CN") == 0 || | |
408 _stricmp(codec.plname, "red") == 0 ) { | |
409 return -1; | |
410 } | |
411 channels_[channel]->send_codec = codec; | |
412 ++num_set_send_codecs_; | |
413 return 0; | |
414 } | |
415 WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) { | |
416 WEBRTC_CHECK_CHANNEL(channel); | |
417 codec = channels_[channel]->send_codec; | |
418 return 0; | |
419 } | |
420 WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps)); | |
421 WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec)); | |
422 WEBRTC_FUNC(SetRecPayloadType, (int channel, | |
423 const webrtc::CodecInst& codec)) { | |
424 WEBRTC_CHECK_CHANNEL(channel); | |
425 Channel* ch = channels_[channel]; | |
426 if (ch->playout) | |
427 return -1; // Channel is in use. | |
428 // Check if something else already has this slot. | |
429 if (codec.pltype != -1) { | |
430 for (std::vector<webrtc::CodecInst>::iterator it = | |
431 ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) { | |
432 if (it->pltype == codec.pltype && | |
433 _stricmp(it->plname, codec.plname) != 0) { | |
434 return -1; | |
435 } | |
436 } | |
437 } | |
438 // Otherwise try to find this codec and update its payload type. | |
439 int result = -1; // not found | |
440 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); | |
441 it != ch->recv_codecs.end(); ++it) { | |
442 if (strcmp(it->plname, codec.plname) == 0 && | |
443 it->plfreq == codec.plfreq && | |
444 it->channels == codec.channels) { | |
445 it->pltype = codec.pltype; | |
446 result = 0; | |
447 } | |
448 } | |
449 return result; | |
450 } | |
451 WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type, | |
452 webrtc::PayloadFrequencies frequency)) { | |
453 WEBRTC_CHECK_CHANNEL(channel); | |
454 if (frequency == webrtc::kFreq8000Hz) { | |
455 channels_[channel]->cn8_type = type; | |
456 } else if (frequency == webrtc::kFreq16000Hz) { | |
457 channels_[channel]->cn16_type = type; | |
458 } | |
459 return 0; | |
460 } | |
461 WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) { | |
462 WEBRTC_CHECK_CHANNEL(channel); | |
463 Channel* ch = channels_[channel]; | |
464 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); | |
465 it != ch->recv_codecs.end(); ++it) { | |
466 if (strcmp(it->plname, codec.plname) == 0 && | |
467 it->plfreq == codec.plfreq && | |
468 it->channels == codec.channels && | |
469 it->pltype != -1) { | |
470 codec.pltype = it->pltype; | |
471 return 0; | |
472 } | |
473 } | |
474 return -1; // not found | |
475 } | |
476 WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode, | |
477 bool disableDTX)) { | |
478 WEBRTC_CHECK_CHANNEL(channel); | |
479 if (channels_[channel]->send_codec.channels == 2) { | |
480 // Replicating VoE behavior; VAD cannot be enabled for stereo. | |
481 return -1; | |
482 } | |
483 channels_[channel]->vad = enable; | |
484 return 0; | |
485 } | |
486 WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled, | |
487 webrtc::VadModes& mode, bool& disabledDTX)); | |
488 | |
489 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) { | |
490 WEBRTC_CHECK_CHANNEL(channel); | |
491 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
492 // Return -1 if current send codec is not Opus. | |
493 // TODO(minyue): Excludes other codecs if they support inband FEC. | |
494 return -1; | |
495 } | |
496 channels_[channel]->codec_fec = enable; | |
497 return 0; | |
498 } | |
499 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) { | |
500 WEBRTC_CHECK_CHANNEL(channel); | |
501 enable = channels_[channel]->codec_fec; | |
502 return 0; | |
503 } | |
504 | |
505 WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) { | |
506 WEBRTC_CHECK_CHANNEL(channel); | |
507 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
508 // Return -1 if current send codec is not Opus. | |
509 return -1; | |
510 } | |
511 if (frequency_hz <= 8000) | |
512 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb; | |
513 else if (frequency_hz <= 12000) | |
514 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb; | |
515 else if (frequency_hz <= 16000) | |
516 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb; | |
517 else if (frequency_hz <= 24000) | |
518 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb; | |
519 else | |
520 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb; | |
521 return 0; | |
522 } | |
523 | |
524 WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) { | |
525 WEBRTC_CHECK_CHANNEL(channel); | |
526 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
527 // Return -1 if current send codec is not Opus. | |
528 return -1; | |
529 } | |
530 channels_[channel]->opus_dtx = enable_dtx; | |
531 return 0; | |
532 } | |
533 | |
534 // webrtc::VoEHardware | |
535 WEBRTC_STUB(GetNumOfRecordingDevices, (int& num)); | |
536 WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num)); | |
537 WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid)); | |
538 WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid)); | |
539 WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel)); | |
540 WEBRTC_STUB(SetPlayoutDevice, (int)); | |
541 WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers)); | |
542 WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&)); | |
543 WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) { | |
544 recording_sample_rate_ = samples_per_sec; | |
545 return 0; | |
546 } | |
547 WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) { | |
548 *samples_per_sec = recording_sample_rate_; | |
549 return 0; | |
550 } | |
551 WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) { | |
552 playout_sample_rate_ = samples_per_sec; | |
553 return 0; | |
554 } | |
555 WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) { | |
556 *samples_per_sec = playout_sample_rate_; | |
557 return 0; | |
558 } | |
559 WEBRTC_STUB(EnableBuiltInAEC, (bool enable)); | |
560 virtual bool BuiltInAECIsAvailable() const { return false; } | |
561 WEBRTC_STUB(EnableBuiltInAGC, (bool enable)); | |
562 virtual bool BuiltInAGCIsAvailable() const { return false; } | |
563 WEBRTC_STUB(EnableBuiltInNS, (bool enable)); | |
564 virtual bool BuiltInNSIsAvailable() const { return false; } | |
565 | |
566 // webrtc::VoENetwork | |
567 WEBRTC_FUNC(RegisterExternalTransport, (int channel, | |
568 webrtc::Transport& transport)) { | |
569 WEBRTC_CHECK_CHANNEL(channel); | |
570 channels_[channel]->external_transport = true; | |
571 return 0; | |
572 } | |
573 WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) { | |
574 WEBRTC_CHECK_CHANNEL(channel); | |
575 channels_[channel]->external_transport = false; | |
576 return 0; | |
577 } | |
578 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, | |
579 size_t length)) { | |
580 WEBRTC_CHECK_CHANNEL(channel); | |
581 if (!channels_[channel]->external_transport) return -1; | |
582 channels_[channel]->packets.push_back( | |
583 std::string(static_cast<const char*>(data), length)); | |
584 return 0; | |
585 } | |
586 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, | |
587 size_t length, | |
588 const webrtc::PacketTime& packet_time)) { | |
589 WEBRTC_CHECK_CHANNEL(channel); | |
590 if (ReceivedRTPPacket(channel, data, length) == -1) { | |
591 return -1; | |
592 } | |
593 channels_[channel]->last_rtp_packet_time = packet_time; | |
594 return 0; | |
595 } | |
596 | |
597 WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data, | |
598 size_t length)); | |
599 | |
600 // webrtc::VoERTP_RTCP | |
601 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) { | |
602 WEBRTC_CHECK_CHANNEL(channel); | |
603 channels_[channel]->send_ssrc = ssrc; | |
604 return 0; | |
605 } | |
606 WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc)); | |
607 WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc)); | |
608 WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable, | |
609 unsigned char id)); | |
610 WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable, | |
611 unsigned char id)); | |
612 WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable, | |
613 unsigned char id)); | |
614 WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable, | |
615 unsigned char id)); | |
616 WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable)); | |
617 WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled)); | |
618 WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256])); | |
619 WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256])); | |
620 WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname)); | |
621 WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh, | |
622 unsigned int& NTPLow, | |
623 unsigned int& timestamp, | |
624 unsigned int& playoutTimestamp, | |
625 unsigned int* jitter, | |
626 unsigned short* fractionLost)); | |
627 WEBRTC_STUB(GetRemoteRTCPReportBlocks, | |
628 (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)); | |
629 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, | |
630 unsigned int& maxJitterMs, | |
631 unsigned int& discardedPackets)); | |
632 WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)); | |
633 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) { | |
634 WEBRTC_CHECK_CHANNEL(channel); | |
635 channels_[channel]->red = enable; | |
636 channels_[channel]->red_type = redPayloadtype; | |
637 return 0; | |
638 } | |
639 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) { | |
640 WEBRTC_CHECK_CHANNEL(channel); | |
641 enable = channels_[channel]->red; | |
642 redPayloadtype = channels_[channel]->red_type; | |
643 return 0; | |
644 } | |
645 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) { | |
646 WEBRTC_CHECK_CHANNEL(channel); | |
647 channels_[channel]->nack = enable; | |
648 channels_[channel]->nack_max_packets = maxNoPackets; | |
649 return 0; | |
650 } | |
651 | |
652 // webrtc::VoEVolumeControl | |
653 WEBRTC_STUB(SetSpeakerVolume, (unsigned int)); | |
654 WEBRTC_STUB(GetSpeakerVolume, (unsigned int&)); | |
655 WEBRTC_STUB(SetMicVolume, (unsigned int)); | |
656 WEBRTC_STUB(GetMicVolume, (unsigned int&)); | |
657 WEBRTC_STUB(SetInputMute, (int, bool)); | |
658 WEBRTC_STUB(GetInputMute, (int, bool&)); | |
659 WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&)); | |
660 WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&)); | |
661 WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&)); | |
662 WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&)); | |
663 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) { | |
664 WEBRTC_CHECK_CHANNEL(channel); | |
665 channels_[channel]->volume_scale= scale; | |
666 return 0; | |
667 } | |
668 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) { | |
669 WEBRTC_CHECK_CHANNEL(channel); | |
670 scale = channels_[channel]->volume_scale; | |
671 return 0; | |
672 } | |
673 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right)); | |
674 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right)); | |
675 | |
676 // webrtc::VoEAudioProcessing | |
677 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { | |
678 ns_enabled_ = enable; | |
679 ns_mode_ = mode; | |
680 return 0; | |
681 } | |
682 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { | |
683 enabled = ns_enabled_; | |
684 mode = ns_mode_; | |
685 return 0; | |
686 } | |
687 | |
688 WEBRTC_FUNC(SetAgcStatus, (bool enable, webrtc::AgcModes mode)) { | |
689 agc_enabled_ = enable; | |
690 agc_mode_ = mode; | |
691 return 0; | |
692 } | |
693 WEBRTC_FUNC(GetAgcStatus, (bool& enabled, webrtc::AgcModes& mode)) { | |
694 enabled = agc_enabled_; | |
695 mode = agc_mode_; | |
696 return 0; | |
697 } | |
698 | |
699 WEBRTC_FUNC(SetAgcConfig, (webrtc::AgcConfig config)) { | |
700 agc_config_ = config; | |
701 return 0; | |
702 } | |
703 WEBRTC_FUNC(GetAgcConfig, (webrtc::AgcConfig& config)) { | |
704 config = agc_config_; | |
705 return 0; | |
706 } | |
707 WEBRTC_FUNC(SetEcStatus, (bool enable, webrtc::EcModes mode)) { | |
708 ec_enabled_ = enable; | |
709 ec_mode_ = mode; | |
710 return 0; | |
711 } | |
712 WEBRTC_FUNC(GetEcStatus, (bool& enabled, webrtc::EcModes& mode)) { | |
713 enabled = ec_enabled_; | |
714 mode = ec_mode_; | |
715 return 0; | |
716 } | |
717 WEBRTC_STUB(EnableDriftCompensation, (bool enable)) | |
718 WEBRTC_BOOL_STUB(DriftCompensationEnabled, ()) | |
719 WEBRTC_VOID_STUB(SetDelayOffsetMs, (int offset)) | |
720 WEBRTC_STUB(DelayOffsetMs, ()); | |
721 WEBRTC_FUNC(SetAecmMode, (webrtc::AecmModes mode, bool enableCNG)) { | |
722 aecm_mode_ = mode; | |
723 cng_enabled_ = enableCNG; | |
724 return 0; | |
725 } | |
726 WEBRTC_FUNC(GetAecmMode, (webrtc::AecmModes& mode, bool& enabledCNG)) { | |
727 mode = aecm_mode_; | |
728 enabledCNG = cng_enabled_; | |
729 return 0; | |
730 } | |
731 WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode)); | |
732 WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled, | |
733 webrtc::NsModes& mode)); | |
734 WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable, | |
735 webrtc::AgcModes mode)); | |
736 WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled, | |
737 webrtc::AgcModes& mode)); | |
738 WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)); | |
739 WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)); | |
740 | |
741 WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&)); | |
742 WEBRTC_STUB(DeRegisterRxVadObserver, (int channel)); | |
743 WEBRTC_STUB(VoiceActivityIndicator, (int channel)); | |
744 WEBRTC_FUNC(SetEcMetricsStatus, (bool enable)) { | |
745 ec_metrics_enabled_ = enable; | |
746 return 0; | |
747 } | |
748 WEBRTC_STUB(GetEcMetricsStatus, (bool& enabled)); | |
749 WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP)); | |
750 WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std, | |
751 float& fraction_poor_delays)); | |
752 | |
753 WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8)); | |
754 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); | |
755 WEBRTC_STUB(StopDebugRecording, ()); | |
756 | |
757 WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) { | |
758 typing_detection_enabled_ = enable; | |
759 return 0; | |
760 } | |
761 WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) { | |
762 enabled = typing_detection_enabled_; | |
763 return 0; | |
764 } | |
765 | |
766 WEBRTC_STUB(TimeSinceLastTyping, (int& seconds)); | |
767 WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow, | |
768 int costPerTyping, | |
769 int reportingThreshold, | |
770 int penaltyDecay, | |
771 int typeEventDelay)); | |
772 int EnableHighPassFilter(bool enable) { | |
773 highpass_filter_enabled_ = enable; | |
774 return 0; | |
775 } | |
776 bool IsHighPassFilterEnabled() { | |
777 return highpass_filter_enabled_; | |
778 } | |
779 bool IsStereoChannelSwappingEnabled() { | |
780 return stereo_swapping_enabled_; | |
781 } | |
782 void EnableStereoChannelSwapping(bool enable) { | |
783 stereo_swapping_enabled_ = enable; | |
784 } | |
785 int GetNetEqCapacity() const { | |
786 auto ch = channels_.find(last_channel_); | |
787 ASSERT(ch != channels_.end()); | |
788 return ch->second->neteq_capacity; | |
789 } | |
790 bool GetNetEqFastAccelerate() const { | |
791 auto ch = channels_.find(last_channel_); | |
792 ASSERT(ch != channels_.end()); | |
793 return ch->second->neteq_fast_accelerate; | |
794 } | |
795 | |
796 private: | |
797 bool inited_; | |
798 int last_channel_; | |
799 std::map<int, Channel*> channels_; | |
800 bool fail_create_channel_; | |
801 int num_set_send_codecs_; // how many times we call SetSendCodec(). | |
802 bool ec_enabled_; | |
803 bool ec_metrics_enabled_; | |
804 bool cng_enabled_; | |
805 bool ns_enabled_; | |
806 bool agc_enabled_; | |
807 bool highpass_filter_enabled_; | |
808 bool stereo_swapping_enabled_; | |
809 bool typing_detection_enabled_; | |
810 webrtc::EcModes ec_mode_; | |
811 webrtc::AecmModes aecm_mode_; | |
812 webrtc::NsModes ns_mode_; | |
813 webrtc::AgcModes agc_mode_; | |
814 webrtc::AgcConfig agc_config_; | |
815 webrtc::VoiceEngineObserver* observer_; | |
816 int playout_fail_channel_; | |
817 int send_fail_channel_; | |
818 int recording_sample_rate_; | |
819 int playout_sample_rate_; | |
820 FakeAudioProcessing audio_processing_; | |
821 }; | |
822 | |
823 } // namespace cricket | |
824 | |
825 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | |
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