henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1 | /* |
kjellander | 1afca73 | 2016-02-08 04:46:45 | [diff] [blame] | 2 | * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 3 | * |
kjellander | 1afca73 | 2016-02-08 04:46:45 | [diff] [blame] | 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. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 9 | */ |
| 10 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 11 | #ifdef HAVE_WEBRTC_VOICE |
| 12 | |
Mirko Bonadei | 92ea95e | 2017-09-15 04:47:31 | [diff] [blame] | 13 | #include "media/engine/webrtcvoiceengine.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 14 | |
| 15 | #include <algorithm> |
| 16 | #include <cstdio> |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 17 | #include <functional> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 18 | #include <string> |
Steve Anton | e78bcb9 | 2017-10-31 16:53:08 | [diff] [blame] | 19 | #include <utility> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 20 | #include <vector> |
| 21 | |
Mirko Bonadei | 92ea95e | 2017-09-15 04:47:31 | [diff] [blame] | 22 | #include "api/call/audio_sink.h" |
| 23 | #include "media/base/audiosource.h" |
| 24 | #include "media/base/mediaconstants.h" |
| 25 | #include "media/base/streamparams.h" |
| 26 | #include "media/engine/adm_helpers.h" |
| 27 | #include "media/engine/apm_helpers.h" |
| 28 | #include "media/engine/payload_type_mapper.h" |
| 29 | #include "media/engine/webrtcmediaengine.h" |
| 30 | #include "media/engine/webrtcvoe.h" |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 31 | #include "modules/audio_device/audio_device_impl.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 04:47:31 | [diff] [blame] | 32 | #include "modules/audio_mixer/audio_mixer_impl.h" |
| 33 | #include "modules/audio_processing/aec_dump/aec_dump_factory.h" |
| 34 | #include "modules/audio_processing/include/audio_processing.h" |
| 35 | #include "rtc_base/arraysize.h" |
| 36 | #include "rtc_base/base64.h" |
| 37 | #include "rtc_base/byteorder.h" |
| 38 | #include "rtc_base/constructormagic.h" |
| 39 | #include "rtc_base/helpers.h" |
| 40 | #include "rtc_base/logging.h" |
| 41 | #include "rtc_base/race_checker.h" |
| 42 | #include "rtc_base/stringencode.h" |
| 43 | #include "rtc_base/stringutils.h" |
| 44 | #include "rtc_base/trace_event.h" |
| 45 | #include "system_wrappers/include/field_trial.h" |
| 46 | #include "system_wrappers/include/metrics.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 04:47:31 | [diff] [blame] | 47 | #include "voice_engine/transmit_mixer.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 48 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 49 | namespace cricket { |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 50 | namespace { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 51 | |
solenberg | 418b7d3 | 2017-06-13 07:38:27 | [diff] [blame] | 52 | constexpr size_t kMaxUnsignaledRecvStreams = 4; |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 53 | |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 54 | constexpr int kNackRtpHistoryMs = 5000; |
minyue@webrtc.org | 2dc6f31 | 2014-10-31 05:33:10 | [diff] [blame] | 55 | |
peah | 1bcfce5 | 2016-08-26 14:16:04 | [diff] [blame] | 56 | // Check to verify that the define for the intelligibility enhancer is properly |
| 57 | // set. |
| 58 | #if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \ |
| 59 | (WEBRTC_INTELLIGIBILITY_ENHANCER != 0 && \ |
| 60 | WEBRTC_INTELLIGIBILITY_ENHANCER != 1) |
| 61 | #error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1" |
| 62 | #endif |
| 63 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 64 | // For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000. |
minyue | 10cbb46 | 2016-11-07 17:29:22 | [diff] [blame] | 65 | const int kOpusMinBitrateBps = 6000; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 66 | const int kOpusBitrateFbBps = 32000; |
deadbeef | 8034614 | 2016-04-27 21:17:10 | [diff] [blame] | 67 | |
wu@webrtc.org | de30501 | 2013-10-31 15:40:38 | [diff] [blame] | 68 | // Default audio dscp value. |
| 69 | // See http://tools.ietf.org/html/rfc2474 for details. |
| 70 | // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00 |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 71 | const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF; |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 72 | |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 73 | const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1) |
| 74 | const int kMaxTelephoneEventCode = 255; |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 75 | |
solenberg | 31642aa | 2016-03-14 15:00:37 | [diff] [blame] | 76 | const int kMinPayloadType = 0; |
| 77 | const int kMaxPayloadType = 127; |
| 78 | |
deadbeef | 884f585 | 2016-01-15 17:20:04 | [diff] [blame] | 79 | class ProxySink : public webrtc::AudioSinkInterface { |
| 80 | public: |
Steve Anton | e78bcb9 | 2017-10-31 16:53:08 | [diff] [blame] | 81 | explicit ProxySink(AudioSinkInterface* sink) : sink_(sink) { |
| 82 | RTC_DCHECK(sink); |
| 83 | } |
deadbeef | 884f585 | 2016-01-15 17:20:04 | [diff] [blame] | 84 | |
| 85 | void OnData(const Data& audio) override { sink_->OnData(audio); } |
| 86 | |
| 87 | private: |
| 88 | webrtc::AudioSinkInterface* sink_; |
| 89 | }; |
| 90 | |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 91 | bool ValidateStreamParams(const StreamParams& sp) { |
| 92 | if (sp.ssrcs.empty()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 93 | RTC_LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 94 | return false; |
| 95 | } |
| 96 | if (sp.ssrcs.size() > 1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 97 | RTC_LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " |
| 98 | << sp.ToString(); |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 99 | return false; |
| 100 | } |
| 101 | return true; |
| 102 | } |
| 103 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 104 | // Dumps an AudioCodec in RFC 2327-ish format. |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 105 | std::string ToString(const AudioCodec& codec) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 106 | std::stringstream ss; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 107 | ss << codec.name << "/" << codec.clockrate << "/" << codec.channels; |
| 108 | if (!codec.params.empty()) { |
| 109 | ss << " {"; |
| 110 | for (const auto& param : codec.params) { |
| 111 | ss << " " << param.first << "=" << param.second; |
| 112 | } |
| 113 | ss << " }"; |
| 114 | } |
| 115 | ss << " (" << codec.id << ")"; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 116 | return ss.str(); |
| 117 | } |
Minyue Li | 7100dcd | 2015-03-27 04:05:59 | [diff] [blame] | 118 | |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 119 | bool IsCodec(const AudioCodec& codec, const char* ref_name) { |
Minyue Li | 7100dcd | 2015-03-27 04:05:59 | [diff] [blame] | 120 | return (_stricmp(codec.name.c_str(), ref_name) == 0); |
| 121 | } |
| 122 | |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 123 | bool FindCodec(const std::vector<AudioCodec>& codecs, |
solenberg | 26c8c91 | 2015-11-27 12:00:25 | [diff] [blame] | 124 | const AudioCodec& codec, |
| 125 | AudioCodec* found_codec) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 08:45:53 | [diff] [blame] | 126 | for (const AudioCodec& c : codecs) { |
| 127 | if (c.Matches(codec)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 128 | if (found_codec != NULL) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 08:45:53 | [diff] [blame] | 129 | *found_codec = c; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 130 | } |
| 131 | return true; |
| 132 | } |
| 133 | } |
| 134 | return false; |
| 135 | } |
wu@webrtc.org | 1d1ffc9 | 2013-10-16 18:12:02 | [diff] [blame] | 136 | |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 137 | bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) { |
| 138 | if (codecs.empty()) { |
| 139 | return true; |
| 140 | } |
| 141 | std::vector<int> payload_types; |
| 142 | for (const AudioCodec& codec : codecs) { |
| 143 | payload_types.push_back(codec.id); |
| 144 | } |
| 145 | std::sort(payload_types.begin(), payload_types.end()); |
| 146 | auto it = std::unique(payload_types.begin(), payload_types.end()); |
| 147 | return it == payload_types.end(); |
| 148 | } |
| 149 | |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 150 | rtc::Optional<std::string> GetAudioNetworkAdaptorConfig( |
| 151 | const AudioOptions& options) { |
| 152 | if (options.audio_network_adaptor && *options.audio_network_adaptor && |
| 153 | options.audio_network_adaptor_config) { |
| 154 | // Turn on audio network adaptor only when |options_.audio_network_adaptor| |
| 155 | // equals true and |options_.audio_network_adaptor_config| has a value. |
| 156 | return options.audio_network_adaptor_config; |
| 157 | } |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 158 | return rtc::nullopt; |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 159 | } |
| 160 | |
gyzhou | 95aa964 | 2016-12-13 22:06:26 | [diff] [blame] | 161 | webrtc::AudioState::Config MakeAudioStateConfig( |
| 162 | VoEWrapper* voe_wrapper, |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 163 | rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 164 | rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 165 | webrtc::AudioState::Config config; |
| 166 | config.voice_engine = voe_wrapper->engine(); |
gyzhou | 95aa964 | 2016-12-13 22:06:26 | [diff] [blame] | 167 | if (audio_mixer) { |
| 168 | config.audio_mixer = audio_mixer; |
| 169 | } else { |
| 170 | config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| 171 | } |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 172 | config.audio_processing = audio_processing; |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 173 | return config; |
| 174 | } |
| 175 | |
deadbeef | e702b30 | 2017-02-04 20:09:01 | [diff] [blame] | 176 | // |max_send_bitrate_bps| is the bitrate from "b=" in SDP. |
| 177 | // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters. |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 178 | rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps, |
deadbeef | e702b30 | 2017-02-04 20:09:01 | [diff] [blame] | 179 | rtc::Optional<int> rtp_max_bitrate_bps, |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 180 | const webrtc::AudioCodecSpec& spec) { |
deadbeef | e702b30 | 2017-02-04 20:09:01 | [diff] [blame] | 181 | // If application-configured bitrate is set, take minimum of that and SDP |
| 182 | // bitrate. |
zstein | a5e0df6 | 2017-06-14 18:41:48 | [diff] [blame] | 183 | const int bps = |
| 184 | rtp_max_bitrate_bps |
| 185 | ? webrtc::MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps) |
| 186 | : max_send_bitrate_bps; |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 187 | if (bps <= 0) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 188 | return spec.info.default_bitrate_bps; |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 189 | } |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 190 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 191 | if (bps < spec.info.min_bitrate_bps) { |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 192 | // If codec is not multi-rate and |bps| is less than the fixed bitrate then |
| 193 | // fail. If codec is not multi-rate and |bps| exceeds or equal the fixed |
| 194 | // bitrate then ignore. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 195 | RTC_LOG(LS_ERROR) << "Failed to set codec " << spec.format.name |
| 196 | << " to bitrate " << bps << " bps" |
| 197 | << ", requires at least " << spec.info.min_bitrate_bps |
| 198 | << " bps."; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 199 | return rtc::nullopt; |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 200 | } |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 201 | |
| 202 | if (spec.info.HasFixedBitrate()) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 203 | return spec.info.default_bitrate_bps; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 204 | } else { |
| 205 | // If codec is multi-rate then just set the bitrate. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 206 | return std::min(bps, spec.info.max_bitrate_bps); |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 207 | } |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 208 | } |
| 209 | |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 210 | } // namespace |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 211 | |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 212 | WebRtcVoiceEngine::WebRtcVoiceEngine( |
| 213 | webrtc::AudioDeviceModule* adm, |
ossu | eb1fde4 | 2017-05-02 13:46:30 | [diff] [blame] | 214 | const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
gyzhou | 95aa964 | 2016-12-13 22:06:26 | [diff] [blame] | 215 | const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 216 | rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 217 | rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) |
ossu | eb1fde4 | 2017-05-02 13:46:30 | [diff] [blame] | 218 | : WebRtcVoiceEngine(adm, |
| 219 | encoder_factory, |
| 220 | decoder_factory, |
| 221 | audio_mixer, |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 222 | audio_processing, |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 223 | nullptr) {} |
solenberg | 26c8c91 | 2015-11-27 12:00:25 | [diff] [blame] | 224 | |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 225 | WebRtcVoiceEngine::WebRtcVoiceEngine( |
| 226 | webrtc::AudioDeviceModule* adm, |
ossu | eb1fde4 | 2017-05-02 13:46:30 | [diff] [blame] | 227 | const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 228 | const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
gyzhou | 95aa964 | 2016-12-13 22:06:26 | [diff] [blame] | 229 | rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 230 | rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing, |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 231 | VoEWrapper* voe_wrapper) |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 232 | : adm_(adm), |
ossu | eb1fde4 | 2017-05-02 13:46:30 | [diff] [blame] | 233 | encoder_factory_(encoder_factory), |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 234 | decoder_factory_(decoder_factory), |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 235 | audio_mixer_(audio_mixer), |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 236 | apm_(audio_processing), |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 237 | voe_wrapper_(voe_wrapper) { |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 238 | // This may be called from any thread, so detach thread checkers. |
| 239 | worker_thread_checker_.DetachFromThread(); |
solenberg | 26c8c91 | 2015-11-27 12:00:25 | [diff] [blame] | 240 | signal_thread_checker_.DetachFromThread(); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 241 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 242 | RTC_DCHECK(decoder_factory); |
| 243 | RTC_DCHECK(encoder_factory); |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 244 | RTC_DCHECK(audio_processing); |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 245 | // The rest of our initialization will happen in Init. |
| 246 | } |
| 247 | |
| 248 | WebRtcVoiceEngine::~WebRtcVoiceEngine() { |
| 249 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 250 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 251 | if (initialized_) { |
| 252 | StopAecDump(); |
| 253 | voe_wrapper_->base()->Terminate(); |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 254 | |
| 255 | // Stop AudioDevice. |
| 256 | adm()->StopPlayout(); |
| 257 | adm()->StopRecording(); |
| 258 | adm()->RegisterAudioCallback(nullptr); |
| 259 | adm()->Terminate(); |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 260 | } |
| 261 | } |
| 262 | |
| 263 | void WebRtcVoiceEngine::Init() { |
| 264 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 265 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::Init"; |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 266 | |
| 267 | // TaskQueue expects to be created/destroyed on the same thread. |
| 268 | low_priority_worker_queue_.reset( |
| 269 | new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW)); |
| 270 | |
| 271 | // VoEWrapper needs to be created on the worker thread. It's expected to be |
| 272 | // null here unless it's being injected for testing. |
| 273 | if (!voe_wrapper_) { |
| 274 | voe_wrapper_.reset(new VoEWrapper()); |
| 275 | } |
solenberg | 26c8c91 | 2015-11-27 12:00:25 | [diff] [blame] | 276 | |
ossu | eb1fde4 | 2017-05-02 13:46:30 | [diff] [blame] | 277 | // Load our audio codec lists. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 278 | RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:"; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 279 | send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders()); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 280 | for (const AudioCodec& codec : send_codecs_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 281 | RTC_LOG(LS_INFO) << ToString(codec); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 282 | } |
| 283 | |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 284 | RTC_LOG(LS_INFO) << "Supported recv codecs in order of preference:"; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 285 | recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders()); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 286 | for (const AudioCodec& codec : recv_codecs_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 287 | RTC_LOG(LS_INFO) << ToString(codec); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 288 | } |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 289 | |
solenberg | 88499ec | 2016-09-07 14:34:41 | [diff] [blame] | 290 | channel_config_.enable_voice_pacing = true; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 291 | |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 292 | #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) |
| 293 | // No ADM supplied? Create a default one. |
solenberg | ff97631 | 2016-03-31 06:28:51 | [diff] [blame] | 294 | if (!adm_) { |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 295 | adm_ = webrtc::AudioDeviceModule::Create( |
| 296 | webrtc::AudioDeviceModule::kPlatformDefaultAudio); |
solenberg | ff97631 | 2016-03-31 06:28:51 | [diff] [blame] | 297 | } |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 298 | #endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE |
| 299 | RTC_CHECK(adm()); |
| 300 | webrtc::adm_helpers::Init(adm()); |
Fredrik Solenberg | 55900fd | 2017-11-23 19:22:55 | [diff] [blame] | 301 | webrtc::apm_helpers::Init(apm()); |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 302 | RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm(), apm(), decoder_factory_)); |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 303 | transmit_mixer_ = voe_wrapper_->base()->transmit_mixer(); |
| 304 | RTC_DCHECK(transmit_mixer_); |
| 305 | |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 306 | // Save the default AGC configuration settings. This must happen before |
solenberg | 246b817 | 2015-12-08 17:50:23 | [diff] [blame] | 307 | // calling ApplyOptions or the default will be overwritten. |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 308 | default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm()); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 309 | |
solenberg | 0f7d293 | 2016-01-15 09:40:39 | [diff] [blame] | 310 | // Set default engine options. |
| 311 | { |
| 312 | AudioOptions options; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 313 | options.echo_cancellation = true; |
| 314 | options.auto_gain_control = true; |
| 315 | options.noise_suppression = true; |
| 316 | options.highpass_filter = true; |
| 317 | options.stereo_swapping = false; |
| 318 | options.audio_jitter_buffer_max_packets = 50; |
| 319 | options.audio_jitter_buffer_fast_accelerate = false; |
| 320 | options.typing_detection = true; |
| 321 | options.adjust_agc_delta = 0; |
| 322 | options.experimental_agc = false; |
| 323 | options.extended_filter_aec = false; |
| 324 | options.delay_agnostic_aec = false; |
| 325 | options.experimental_ns = false; |
| 326 | options.intelligibility_enhancer = false; |
| 327 | options.level_control = false; |
| 328 | options.residual_echo_detector = true; |
solenberg | ff97631 | 2016-03-31 06:28:51 | [diff] [blame] | 329 | bool error = ApplyOptions(options); |
| 330 | RTC_DCHECK(error); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 331 | } |
| 332 | |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 333 | // May be null for VoE injected for testing. |
| 334 | if (voe()->engine()) { |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 335 | audio_state_ = webrtc::AudioState::Create( |
| 336 | MakeAudioStateConfig(voe(), audio_mixer_, apm_)); |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 337 | |
| 338 | // Connect the ADM to our audio path. |
| 339 | adm()->RegisterAudioCallback(audio_state_->audio_transport()); |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 340 | } |
| 341 | |
| 342 | initialized_ = true; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 343 | } |
| 344 | |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 345 | rtc::scoped_refptr<webrtc::AudioState> |
| 346 | WebRtcVoiceEngine::GetAudioState() const { |
| 347 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 348 | return audio_state_; |
| 349 | } |
| 350 | |
nisse | 51542be | 2016-02-12 10:27:06 | [diff] [blame] | 351 | VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel( |
| 352 | webrtc::Call* call, |
| 353 | const MediaConfig& config, |
Jelena Marusic | c28a896 | 2015-05-29 13:05:44 | [diff] [blame] | 354 | const AudioOptions& options) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 355 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
nisse | 51542be | 2016-02-12 10:27:06 | [diff] [blame] | 356 | return new WebRtcVoiceMediaChannel(this, config, options, call); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 357 | } |
| 358 | |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 359 | bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 360 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 361 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: " |
| 362 | << options_in.ToString(); |
solenberg | 0f7d293 | 2016-01-15 09:40:39 | [diff] [blame] | 363 | AudioOptions options = options_in; // The options are modified below. |
solenberg | 246b817 | 2015-12-08 17:50:23 | [diff] [blame] | 364 | |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 365 | // Set and adjust echo canceller options. |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 366 | // kEcConference is AEC with high suppression. |
| 367 | webrtc::EcModes ec_mode = webrtc::kEcConference; |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 368 | if (options.aecm_generate_comfort_noise) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 369 | RTC_LOG(LS_VERBOSE) << "Comfort noise explicitly set to " |
| 370 | << *options.aecm_generate_comfort_noise |
| 371 | << " (default is false)."; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 372 | } |
| 373 | |
kjellander | fcfc804 | 2016-01-14 19:01:09 | [diff] [blame] | 374 | #if defined(WEBRTC_IOS) |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 375 | // On iOS, VPIO provides built-in EC. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 376 | options.echo_cancellation = false; |
| 377 | options.extended_filter_aec = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 378 | RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead."; |
Mirko Bonadei | c8c71b9 | 2017-10-16 09:08:54 | [diff] [blame] | 379 | #elif defined(WEBRTC_ANDROID) |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 380 | ec_mode = webrtc::kEcAecm; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 381 | options.extended_filter_aec = false; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 382 | #endif |
| 383 | |
Bjorn Volcker | bf395c1 | 2015-03-25 21:45:56 | [diff] [blame] | 384 | // Delay Agnostic AEC automatically turns on EC if not set except on iOS |
| 385 | // where the feature is not supported. |
| 386 | bool use_delay_agnostic_aec = false; |
kjellander | fcfc804 | 2016-01-14 19:01:09 | [diff] [blame] | 387 | #if !defined(WEBRTC_IOS) |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 388 | if (options.delay_agnostic_aec) { |
| 389 | use_delay_agnostic_aec = *options.delay_agnostic_aec; |
Bjorn Volcker | bf395c1 | 2015-03-25 21:45:56 | [diff] [blame] | 390 | if (use_delay_agnostic_aec) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 391 | options.echo_cancellation = true; |
| 392 | options.extended_filter_aec = true; |
Bjorn Volcker | bf395c1 | 2015-03-25 21:45:56 | [diff] [blame] | 393 | ec_mode = webrtc::kEcConference; |
| 394 | } |
| 395 | } |
| 396 | #endif |
| 397 | |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 398 | // Set and adjust noise suppressor options. |
| 399 | #if defined(WEBRTC_IOS) |
| 400 | // On iOS, VPIO provides built-in NS. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 401 | options.noise_suppression = false; |
| 402 | options.typing_detection = false; |
| 403 | options.experimental_ns = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 404 | RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead."; |
Mirko Bonadei | c8c71b9 | 2017-10-16 09:08:54 | [diff] [blame] | 405 | #elif defined(WEBRTC_ANDROID) |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 406 | options.typing_detection = false; |
| 407 | options.experimental_ns = false; |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 408 | #endif |
| 409 | |
| 410 | // Set and adjust gain control options. |
| 411 | #if defined(WEBRTC_IOS) |
| 412 | // On iOS, VPIO provides built-in AGC. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 413 | options.auto_gain_control = false; |
| 414 | options.experimental_agc = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 415 | RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead."; |
Mirko Bonadei | c8c71b9 | 2017-10-16 09:08:54 | [diff] [blame] | 416 | #elif defined(WEBRTC_ANDROID) |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 417 | options.experimental_agc = false; |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 418 | #endif |
| 419 | |
| 420 | #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) |
Mirko Bonadei | c8c71b9 | 2017-10-16 09:08:54 | [diff] [blame] | 421 | // Turn off the gain control if specified by the field trial. |
| 422 | // The purpose of the field trial is to reduce the amount of resampling |
| 423 | // performed inside the audio processing module on mobile platforms by |
| 424 | // whenever possible turning off the fixed AGC mode and the high-pass filter. |
| 425 | // (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181). |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 426 | if (webrtc::field_trial::IsEnabled( |
| 427 | "WebRTC-Audio-MinimizeResamplingOnMobile")) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 428 | options.auto_gain_control = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 429 | RTC_LOG(LS_INFO) << "Disable AGC according to field trial."; |
Steve Anton | e78bcb9 | 2017-10-31 16:53:08 | [diff] [blame] | 430 | if (!(options.noise_suppression.value_or(false) || |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 431 | options.echo_cancellation.value_or(false))) { |
| 432 | // If possible, turn off the high-pass filter. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 433 | RTC_LOG(LS_INFO) |
| 434 | << "Disable high-pass filter in response to field trial."; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 435 | options.highpass_filter = false; |
peah | 8a8ebd9 | 2017-05-22 22:48:47 | [diff] [blame] | 436 | } |
| 437 | } |
| 438 | #endif |
| 439 | |
peah | 1bcfce5 | 2016-08-26 14:16:04 | [diff] [blame] | 440 | #if (WEBRTC_INTELLIGIBILITY_ENHANCER == 0) |
| 441 | // Hardcode the intelligibility enhancer to be off. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 442 | options.intelligibility_enhancer = false; |
peah | 1bcfce5 | 2016-08-26 14:16:04 | [diff] [blame] | 443 | #endif |
| 444 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 445 | if (options.echo_cancellation) { |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 | [diff] [blame] | 446 | // Check if platform supports built-in EC. Currently only supported on |
| 447 | // Android and in combination with Java based audio layer. |
| 448 | // TODO(henrika): investigate possibility to support built-in EC also |
| 449 | // in combination with Open SL ES audio. |
solenberg | 5b5129a | 2016-04-08 12:35:48 | [diff] [blame] | 450 | const bool built_in_aec = adm()->BuiltInAECIsAvailable(); |
Bjorn Volcker | 73f7210 | 2015-06-03 12:50:15 | [diff] [blame] | 451 | if (built_in_aec) { |
Bjorn Volcker | ccfc939 | 2015-05-07 05:43:17 | [diff] [blame] | 452 | // Built-in EC exists on this device and use_delay_agnostic_aec is not |
| 453 | // overriding it. Enable/Disable it according to the echo_cancellation |
| 454 | // audio option. |
Bjorn Volcker | 73f7210 | 2015-06-03 12:50:15 | [diff] [blame] | 455 | const bool enable_built_in_aec = |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 456 | *options.echo_cancellation && !use_delay_agnostic_aec; |
solenberg | 5b5129a | 2016-04-08 12:35:48 | [diff] [blame] | 457 | if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 && |
Bjorn Volcker | 73f7210 | 2015-06-03 12:50:15 | [diff] [blame] | 458 | enable_built_in_aec) { |
Bjorn Volcker | bf395c1 | 2015-03-25 21:45:56 | [diff] [blame] | 459 | // Disable internal software EC if built-in EC is enabled, |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 | [diff] [blame] | 460 | // i.e., replace the software EC with the built-in EC. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 461 | options.echo_cancellation = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 462 | RTC_LOG(LS_INFO) |
| 463 | << "Disabling EC since built-in EC will be used instead"; |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 | [diff] [blame] | 464 | } |
| 465 | } |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 466 | webrtc::apm_helpers::SetEcStatus( |
| 467 | apm(), *options.echo_cancellation, ec_mode); |
Mirko Bonadei | c8c71b9 | 2017-10-16 09:08:54 | [diff] [blame] | 468 | #if !defined(WEBRTC_ANDROID) |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 469 | webrtc::apm_helpers::SetEcMetricsStatus(apm(), *options.echo_cancellation); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 470 | #endif |
| 471 | if (ec_mode == webrtc::kEcAecm) { |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 472 | bool cn = options.aecm_generate_comfort_noise.value_or(false); |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 473 | webrtc::apm_helpers::SetAecmMode(apm(), cn); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 474 | } |
| 475 | } |
| 476 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 477 | if (options.auto_gain_control) { |
peah | 72a5645 | 2016-08-22 19:08:55 | [diff] [blame] | 478 | bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable(); |
| 479 | if (built_in_agc_avaliable) { |
solenberg | 5b5129a | 2016-04-08 12:35:48 | [diff] [blame] | 480 | if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 && |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 481 | *options.auto_gain_control) { |
henrika | c14f5ff | 2015-09-23 12:08:33 | [diff] [blame] | 482 | // Disable internal software AGC if built-in AGC is enabled, |
| 483 | // i.e., replace the software AGC with the built-in AGC. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 484 | options.auto_gain_control = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 485 | RTC_LOG(LS_INFO) |
| 486 | << "Disabling AGC since built-in AGC will be used instead"; |
henrika | c14f5ff | 2015-09-23 12:08:33 | [diff] [blame] | 487 | } |
| 488 | } |
solenberg | 22818a5 | 2017-03-16 08:20:23 | [diff] [blame] | 489 | webrtc::apm_helpers::SetAgcStatus(apm(), adm(), *options.auto_gain_control); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 490 | } |
| 491 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 492 | if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain || |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 493 | options.tx_agc_limiter || options.adjust_agc_delta) { |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 494 | // Override default_agc_config_. Generally, an unset option means "leave |
| 495 | // the VoE bits alone" in this function, so we want whatever is set to be |
| 496 | // stored as the new "default". If we didn't, then setting e.g. |
| 497 | // tx_agc_target_dbov would reset digital compression gain and limiter |
| 498 | // settings. |
| 499 | // Also, if we don't update default_agc_config_, then adjust_agc_delta |
| 500 | // would be an offset from the original values, and not whatever was set |
| 501 | // explicitly. |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 502 | default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or( |
| 503 | default_agc_config_.targetLeveldBOv); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 504 | default_agc_config_.digitalCompressionGaindB = |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 505 | options.tx_agc_digital_compression_gain.value_or( |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 506 | default_agc_config_.digitalCompressionGaindB); |
| 507 | default_agc_config_.limiterEnable = |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 508 | options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable); |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 509 | |
| 510 | webrtc::AgcConfig config = default_agc_config_; |
| 511 | if (options.adjust_agc_delta) { |
| 512 | config.targetLeveldBOv -= *options.adjust_agc_delta; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 513 | RTC_LOG(LS_INFO) << "Adjusting AGC level from default -" |
| 514 | << default_agc_config_.targetLeveldBOv << "dB to -" |
| 515 | << config.targetLeveldBOv << "dB"; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 516 | } |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 517 | webrtc::apm_helpers::SetAgcConfig(apm(), config); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 518 | } |
| 519 | |
Alejandro Luebs | c9b0c26 | 2016-05-16 22:32:38 | [diff] [blame] | 520 | if (options.intelligibility_enhancer) { |
| 521 | intelligibility_enhancer_ = options.intelligibility_enhancer; |
| 522 | } |
| 523 | if (intelligibility_enhancer_ && *intelligibility_enhancer_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 524 | RTC_LOG(LS_INFO) << "Enabling NS when Intelligibility Enhancer is active."; |
Alejandro Luebs | c9b0c26 | 2016-05-16 22:32:38 | [diff] [blame] | 525 | options.noise_suppression = intelligibility_enhancer_; |
| 526 | } |
| 527 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 528 | if (options.noise_suppression) { |
Alejandro Luebs | c9b0c26 | 2016-05-16 22:32:38 | [diff] [blame] | 529 | if (adm()->BuiltInNSIsAvailable()) { |
| 530 | bool builtin_ns = |
| 531 | *options.noise_suppression && |
| 532 | !(intelligibility_enhancer_ && *intelligibility_enhancer_); |
| 533 | if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) { |
henrika | c14f5ff | 2015-09-23 12:08:33 | [diff] [blame] | 534 | // Disable internal software NS if built-in NS is enabled, |
| 535 | // i.e., replace the software NS with the built-in NS. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 536 | options.noise_suppression = false; |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 537 | RTC_LOG(LS_INFO) |
| 538 | << "Disabling NS since built-in NS will be used instead"; |
henrika | c14f5ff | 2015-09-23 12:08:33 | [diff] [blame] | 539 | } |
| 540 | } |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 541 | webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 542 | } |
| 543 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 544 | if (options.stereo_swapping) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 545 | RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping; |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 546 | transmit_mixer()->EnableStereoChannelSwapping(*options.stereo_swapping); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 547 | } |
| 548 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 549 | if (options.audio_jitter_buffer_max_packets) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 550 | RTC_LOG(LS_INFO) << "NetEq capacity is " |
| 551 | << *options.audio_jitter_buffer_max_packets; |
solenberg | 88499ec | 2016-09-07 14:34:41 | [diff] [blame] | 552 | channel_config_.acm_config.neteq_config.max_packets_in_buffer = |
| 553 | std::max(20, *options.audio_jitter_buffer_max_packets); |
Henrik Lundin | 64dad83 | 2015-05-11 10:44:23 | [diff] [blame] | 554 | } |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 555 | if (options.audio_jitter_buffer_fast_accelerate) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 556 | RTC_LOG(LS_INFO) << "NetEq fast mode? " |
| 557 | << *options.audio_jitter_buffer_fast_accelerate; |
solenberg | 88499ec | 2016-09-07 14:34:41 | [diff] [blame] | 558 | channel_config_.acm_config.neteq_config.enable_fast_accelerate = |
| 559 | *options.audio_jitter_buffer_fast_accelerate; |
Henrik Lundin | 5263b3c | 2015-06-01 08:29:41 | [diff] [blame] | 560 | } |
| 561 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 562 | if (options.typing_detection) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 563 | RTC_LOG(LS_INFO) << "Typing detection is enabled? " |
| 564 | << *options.typing_detection; |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 565 | webrtc::apm_helpers::SetTypingDetectionStatus( |
| 566 | apm(), *options.typing_detection); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 567 | } |
| 568 | |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 | [diff] [blame] | 569 | webrtc::Config config; |
| 570 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 571 | if (options.delay_agnostic_aec) |
| 572 | delay_agnostic_aec_ = options.delay_agnostic_aec; |
| 573 | if (delay_agnostic_aec_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 574 | RTC_LOG(LS_INFO) << "Delay agnostic aec is enabled? " |
| 575 | << *delay_agnostic_aec_; |
henrik.lundin | 0f133b9 | 2015-07-02 07:17:55 | [diff] [blame] | 576 | config.Set<webrtc::DelayAgnostic>( |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 577 | new webrtc::DelayAgnostic(*delay_agnostic_aec_)); |
Bjorn Volcker | bf395c1 | 2015-03-25 21:45:56 | [diff] [blame] | 578 | } |
| 579 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 580 | if (options.extended_filter_aec) { |
| 581 | extended_filter_aec_ = options.extended_filter_aec; |
| 582 | } |
| 583 | if (extended_filter_aec_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 584 | RTC_LOG(LS_INFO) << "Extended filter aec is enabled? " |
| 585 | << *extended_filter_aec_; |
Henrik Lundin | 441f634 | 2015-06-09 14:03:13 | [diff] [blame] | 586 | config.Set<webrtc::ExtendedFilter>( |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 587 | new webrtc::ExtendedFilter(*extended_filter_aec_)); |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 | [diff] [blame] | 588 | } |
| 589 | |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 590 | if (options.experimental_ns) { |
| 591 | experimental_ns_ = options.experimental_ns; |
| 592 | } |
| 593 | if (experimental_ns_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 594 | RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_; |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 | [diff] [blame] | 595 | config.Set<webrtc::ExperimentalNs>( |
kwiberg | 102c6a6 | 2015-10-30 09:47:38 | [diff] [blame] | 596 | new webrtc::ExperimentalNs(*experimental_ns_)); |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 | [diff] [blame] | 597 | } |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 | [diff] [blame] | 598 | |
Alejandro Luebs | c9b0c26 | 2016-05-16 22:32:38 | [diff] [blame] | 599 | if (intelligibility_enhancer_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 600 | RTC_LOG(LS_INFO) << "Intelligibility Enhancer is enabled? " |
| 601 | << *intelligibility_enhancer_; |
Alejandro Luebs | c9b0c26 | 2016-05-16 22:32:38 | [diff] [blame] | 602 | config.Set<webrtc::Intelligibility>( |
| 603 | new webrtc::Intelligibility(*intelligibility_enhancer_)); |
| 604 | } |
| 605 | |
peah | a3333bf | 2016-06-30 07:02:34 | [diff] [blame] | 606 | if (options.level_control) { |
| 607 | level_control_ = options.level_control; |
| 608 | } |
| 609 | |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 610 | webrtc::AudioProcessing::Config apm_config = apm()->GetConfig(); |
| 611 | |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 612 | RTC_LOG(LS_INFO) << "Level control: " |
| 613 | << (!!level_control_ ? *level_control_ : -1); |
peah | a3333bf | 2016-06-30 07:02:34 | [diff] [blame] | 614 | if (level_control_) { |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 615 | apm_config.level_controller.enabled = *level_control_; |
aleloi | e33c5d9 | 2016-10-20 08:53:27 | [diff] [blame] | 616 | if (options.level_control_initial_peak_level_dbfs) { |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 617 | apm_config.level_controller.initial_peak_level_dbfs = |
aleloi | e33c5d9 | 2016-10-20 08:53:27 | [diff] [blame] | 618 | *options.level_control_initial_peak_level_dbfs; |
| 619 | } |
peah | a3333bf | 2016-06-30 07:02:34 | [diff] [blame] | 620 | } |
| 621 | |
peah | 8271d04 | 2016-11-22 15:24:52 | [diff] [blame] | 622 | if (options.highpass_filter) { |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 623 | apm_config.high_pass_filter.enabled = *options.highpass_filter; |
peah | 8271d04 | 2016-11-22 15:24:52 | [diff] [blame] | 624 | } |
| 625 | |
ivoc | 4ca1869 | 2017-02-10 13:11:09 | [diff] [blame] | 626 | if (options.residual_echo_detector) { |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 627 | apm_config.residual_echo_detector.enabled = *options.residual_echo_detector; |
ivoc | 4ca1869 | 2017-02-10 13:11:09 | [diff] [blame] | 628 | } |
| 629 | |
solenberg | 059fb44 | 2016-10-26 12:12:24 | [diff] [blame] | 630 | apm()->SetExtraOptions(config); |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 631 | apm()->ApplyConfig(apm_config); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 | [diff] [blame] | 632 | return true; |
| 633 | } |
| 634 | |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 635 | // TODO(solenberg): Remove, once AudioMonitor is gone. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 636 | int WebRtcVoiceEngine::GetInputLevel() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 637 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 638 | int8_t level = transmit_mixer()->AudioLevel(); |
| 639 | RTC_DCHECK_LE(0, level); |
| 640 | return level; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 641 | } |
| 642 | |
ossu | dedfd28 | 2016-06-14 14:12:39 | [diff] [blame] | 643 | const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const { |
| 644 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 645 | return send_codecs_; |
ossu | dedfd28 | 2016-06-14 14:12:39 | [diff] [blame] | 646 | } |
| 647 | |
| 648 | const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 649 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 650 | return recv_codecs_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 651 | } |
| 652 | |
Stefan Holmer | 9d69c3f | 2015-12-07 09:45:43 | [diff] [blame] | 653 | RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 654 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
Stefan Holmer | 9d69c3f | 2015-12-07 09:45:43 | [diff] [blame] | 655 | RtpCapabilities capabilities; |
Stefan Holmer | 9d69c3f | 2015-12-07 09:45:43 | [diff] [blame] | 656 | capabilities.header_extensions.push_back( |
isheriff | 6f8d686 | 2016-05-26 18:24:55 | [diff] [blame] | 657 | webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, |
| 658 | webrtc::RtpExtension::kAudioLevelDefaultId)); |
sprang | c1b57a1 | 2017-02-28 16:50:47 | [diff] [blame] | 659 | if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) { |
isheriff | 6f8d686 | 2016-05-26 18:24:55 | [diff] [blame] | 660 | capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| 661 | webrtc::RtpExtension::kTransportSequenceNumberUri, |
| 662 | webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
stefan | ba4c0e4 | 2016-02-04 12:12:24 | [diff] [blame] | 663 | } |
Stefan Holmer | 9d69c3f | 2015-12-07 09:45:43 | [diff] [blame] | 664 | return capabilities; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 665 | } |
| 666 | |
solenberg | 63b3454 | 2015-09-29 13:06:31 | [diff] [blame] | 667 | void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 668 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 669 | RTC_DCHECK(channel); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 670 | channels_.push_back(channel); |
| 671 | } |
| 672 | |
solenberg | 63b3454 | 2015-09-29 13:06:31 | [diff] [blame] | 673 | void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 674 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 63b3454 | 2015-09-29 13:06:31 | [diff] [blame] | 675 | auto it = std::find(channels_.begin(), channels_.end(), channel); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 676 | RTC_DCHECK(it != channels_.end()); |
| 677 | channels_.erase(it); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 678 | } |
| 679 | |
ivoc | d66b44d | 2016-01-15 11:06:36 | [diff] [blame] | 680 | bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
| 681 | int64_t max_size_bytes) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 682 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 683 | auto aec_dump = webrtc::AecDumpFactory::Create( |
| 684 | file, max_size_bytes, low_priority_worker_queue_.get()); |
aleloi | 048cbdd | 2017-05-29 09:56:27 | [diff] [blame] | 685 | if (!aec_dump) { |
wu@webrtc.org | a8910d2 | 2014-01-23 22:12:45 | [diff] [blame] | 686 | return false; |
| 687 | } |
aleloi | 048cbdd | 2017-05-29 09:56:27 | [diff] [blame] | 688 | apm()->AttachAecDump(std::move(aec_dump)); |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 | [diff] [blame] | 689 | return true; |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 | [diff] [blame] | 690 | } |
| 691 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 692 | void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 693 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
aleloi | 048cbdd | 2017-05-29 09:56:27 | [diff] [blame] | 694 | |
deadbeef | eb02c03 | 2017-06-15 15:29:25 | [diff] [blame] | 695 | auto aec_dump = webrtc::AecDumpFactory::Create( |
| 696 | filename, -1, low_priority_worker_queue_.get()); |
aleloi | 048cbdd | 2017-05-29 09:56:27 | [diff] [blame] | 697 | if (aec_dump) { |
| 698 | apm()->AttachAecDump(std::move(aec_dump)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 699 | } |
| 700 | } |
| 701 | |
| 702 | void WebRtcVoiceEngine::StopAecDump() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 703 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
aleloi | 048cbdd | 2017-05-29 09:56:27 | [diff] [blame] | 704 | apm()->DetachAecDump(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 705 | } |
| 706 | |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 707 | int WebRtcVoiceEngine::CreateVoEChannel() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 708 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 88499ec | 2016-09-07 14:34:41 | [diff] [blame] | 709 | return voe_wrapper_->base()->CreateChannel(channel_config_); |
sergeyu@chromium.org | 5bc25c4 | 2013-12-05 00:24:06 | [diff] [blame] | 710 | } |
| 711 | |
solenberg | 5b5129a | 2016-04-08 12:35:48 | [diff] [blame] | 712 | webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { |
| 713 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 714 | RTC_DCHECK(adm_); |
Fredrik Solenberg | d319534 | 2017-11-21 19:33:05 | [diff] [blame] | 715 | return adm_.get(); |
solenberg | 5b5129a | 2016-04-08 12:35:48 | [diff] [blame] | 716 | } |
| 717 | |
peah | b1c9d1d | 2017-07-25 22:45:24 | [diff] [blame] | 718 | webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const { |
solenberg | 059fb44 | 2016-10-26 12:12:24 | [diff] [blame] | 719 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 55900fd | 2017-11-23 19:22:55 | [diff] [blame] | 720 | RTC_DCHECK(apm_); |
peah | a9cc40b | 2017-06-29 15:32:09 | [diff] [blame] | 721 | return apm_.get(); |
solenberg | 059fb44 | 2016-10-26 12:12:24 | [diff] [blame] | 722 | } |
| 723 | |
solenberg | 76377c5 | 2017-02-21 08:54:31 | [diff] [blame] | 724 | webrtc::voe::TransmitMixer* WebRtcVoiceEngine::transmit_mixer() { |
| 725 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 726 | RTC_DCHECK(transmit_mixer_); |
| 727 | return transmit_mixer_; |
| 728 | } |
| 729 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 730 | AudioCodecs WebRtcVoiceEngine::CollectCodecs( |
| 731 | const std::vector<webrtc::AudioCodecSpec>& specs) const { |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 732 | PayloadTypeMapper mapper; |
| 733 | AudioCodecs out; |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 734 | |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 735 | // Only generate CN payload types for these clockrates: |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 736 | std::map<int, bool, std::greater<int>> generate_cn = {{ 8000, false }, |
| 737 | { 16000, false }, |
| 738 | { 32000, false }}; |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 739 | // Only generate telephone-event payload types for these clockrates: |
| 740 | std::map<int, bool, std::greater<int>> generate_dtmf = {{ 8000, false }, |
| 741 | { 16000, false }, |
| 742 | { 32000, false }, |
| 743 | { 48000, false }}; |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 744 | |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 745 | auto map_format = [&mapper](const webrtc::SdpAudioFormat& format, |
| 746 | AudioCodecs* out) { |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 747 | rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format); |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 748 | if (opt_codec) { |
| 749 | if (out) { |
| 750 | out->push_back(*opt_codec); |
| 751 | } |
| 752 | } else { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 753 | RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: " |
| 754 | << format; |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 755 | } |
| 756 | |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 757 | return opt_codec; |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 758 | }; |
| 759 | |
ossu | d4e9f62 | 2016-08-18 09:01:17 | [diff] [blame] | 760 | for (const auto& spec : specs) { |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 761 | // We need to do some extra stuff before adding the main codecs to out. |
| 762 | rtc::Optional<AudioCodec> opt_codec = map_format(spec.format, nullptr); |
| 763 | if (opt_codec) { |
| 764 | AudioCodec& codec = *opt_codec; |
ossu | a1a040a | 2017-04-06 17:03:21 | [diff] [blame] | 765 | if (spec.info.supports_network_adaption) { |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 766 | codec.AddFeedbackParam( |
| 767 | FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty)); |
| 768 | } |
| 769 | |
ossu | a1a040a | 2017-04-06 17:03:21 | [diff] [blame] | 770 | if (spec.info.allow_comfort_noise) { |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 771 | // Generate a CN entry if the decoder allows it and we support the |
| 772 | // clockrate. |
| 773 | auto cn = generate_cn.find(spec.format.clockrate_hz); |
| 774 | if (cn != generate_cn.end()) { |
| 775 | cn->second = true; |
| 776 | } |
| 777 | } |
| 778 | |
| 779 | // Generate a telephone-event entry if we support the clockrate. |
| 780 | auto dtmf = generate_dtmf.find(spec.format.clockrate_hz); |
| 781 | if (dtmf != generate_dtmf.end()) { |
| 782 | dtmf->second = true; |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 783 | } |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 784 | |
| 785 | out.push_back(codec); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 786 | } |
| 787 | } |
| 788 | |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 789 | // Add CN codecs after "proper" audio codecs. |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 790 | for (const auto& cn : generate_cn) { |
| 791 | if (cn.second) { |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 792 | map_format({kCnCodecName, cn.first, 1}, &out); |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 793 | } |
| 794 | } |
| 795 | |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 796 | // Add telephone-event codecs last. |
| 797 | for (const auto& dtmf : generate_dtmf) { |
| 798 | if (dtmf.second) { |
ossu | 9def800 | 2017-02-09 13:14:32 | [diff] [blame] | 799 | map_format({kDtmfCodecName, dtmf.first, 1}, &out); |
solenberg | 2779bab | 2016-11-17 12:45:19 | [diff] [blame] | 800 | } |
| 801 | } |
ossu | c54071d | 2016-08-17 09:45:41 | [diff] [blame] | 802 | |
| 803 | return out; |
| 804 | } |
| 805 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 806 | class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 807 | : public AudioSource::Sink { |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 808 | public: |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 809 | WebRtcAudioSendStream( |
| 810 | int ch, |
| 811 | webrtc::AudioTransport* voe_audio_transport, |
| 812 | uint32_t ssrc, |
| 813 | const std::string& c_name, |
Alex Narest | b3944f0 | 2017-10-13 12:56:18 | [diff] [blame] | 814 | const std::string track_id, |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 815 | const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>& |
| 816 | send_codec_spec, |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 817 | const std::vector<webrtc::RtpExtension>& extensions, |
| 818 | int max_send_bitrate_bps, |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 819 | const rtc::Optional<std::string>& audio_network_adaptor_config, |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 820 | webrtc::Call* call, |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 821 | webrtc::Transport* send_transport, |
| 822 | const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory) |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 823 | : voe_audio_transport_(voe_audio_transport), |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 824 | call_(call), |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 825 | config_(send_transport), |
sprang | c1b57a1 | 2017-02-28 16:50:47 | [diff] [blame] | 826 | send_side_bwe_with_overhead_( |
| 827 | webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 828 | max_send_bitrate_bps_(max_send_bitrate_bps), |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 829 | rtp_parameters_(CreateRtpParametersWithOneEncoding()) { |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 830 | RTC_DCHECK_GE(ch, 0); |
| 831 | // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore: |
| 832 | // RTC_DCHECK(voe_audio_transport); |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 833 | RTC_DCHECK(call); |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 834 | RTC_DCHECK(encoder_factory); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 835 | config_.rtp.ssrc = ssrc; |
| 836 | config_.rtp.c_name = c_name; |
| 837 | config_.voe_channel_id = ch; |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 838 | config_.rtp.extensions = extensions; |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 839 | config_.audio_network_adaptor_config = audio_network_adaptor_config; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 840 | config_.encoder_factory = encoder_factory; |
Alex Narest | b3944f0 | 2017-10-13 12:56:18 | [diff] [blame] | 841 | config_.track_id = track_id; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 842 | rtp_parameters_.encodings[0].ssrc = ssrc; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 843 | |
| 844 | if (send_codec_spec) { |
| 845 | UpdateSendCodecSpec(*send_codec_spec); |
| 846 | } |
| 847 | |
| 848 | stream_ = call_->CreateAudioSendStream(config_); |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 849 | } |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 850 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 851 | ~WebRtcAudioSendStream() override { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 852 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 853 | ClearSource(); |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 854 | call_->DestroyAudioSendStream(stream_); |
| 855 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 856 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 857 | void SetSendCodecSpec( |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 858 | const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 859 | UpdateSendCodecSpec(send_codec_spec); |
| 860 | ReconfigureAudioSendStream(); |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 861 | } |
| 862 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 863 | void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions) { |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 864 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 865 | config_.rtp.extensions = extensions; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 866 | ReconfigureAudioSendStream(); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 867 | } |
| 868 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 869 | void SetAudioNetworkAdaptorConfig( |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 870 | const rtc::Optional<std::string>& audio_network_adaptor_config) { |
| 871 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 872 | if (config_.audio_network_adaptor_config == audio_network_adaptor_config) { |
| 873 | return; |
| 874 | } |
| 875 | config_.audio_network_adaptor_config = audio_network_adaptor_config; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 876 | UpdateAllowedBitrateRange(); |
| 877 | ReconfigureAudioSendStream(); |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 878 | } |
| 879 | |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 880 | bool SetMaxSendBitrate(int bps) { |
| 881 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 882 | RTC_DCHECK(config_.send_codec_spec); |
| 883 | RTC_DCHECK(audio_codec_spec_); |
| 884 | auto send_rate = ComputeSendBitrate( |
| 885 | bps, rtp_parameters_.encodings[0].max_bitrate_bps, *audio_codec_spec_); |
| 886 | |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 887 | if (!send_rate) { |
| 888 | return false; |
| 889 | } |
| 890 | |
| 891 | max_send_bitrate_bps_ = bps; |
| 892 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 893 | if (send_rate != config_.send_codec_spec->target_bitrate_bps) { |
| 894 | config_.send_codec_spec->target_bitrate_bps = send_rate; |
| 895 | ReconfigureAudioSendStream(); |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 896 | } |
| 897 | return true; |
| 898 | } |
| 899 | |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 900 | bool SendTelephoneEvent(int payload_type, int payload_freq, int event, |
| 901 | int duration_ms) { |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 902 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 903 | RTC_DCHECK(stream_); |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 904 | return stream_->SendTelephoneEvent(payload_type, payload_freq, event, |
| 905 | duration_ms); |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 906 | } |
| 907 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 908 | void SetSend(bool send) { |
| 909 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 910 | send_ = send; |
| 911 | UpdateSendState(); |
| 912 | } |
| 913 | |
solenberg | 9421853 | 2016-06-16 17:53:22 | [diff] [blame] | 914 | void SetMuted(bool muted) { |
| 915 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 916 | RTC_DCHECK(stream_); |
| 917 | stream_->SetMuted(muted); |
| 918 | muted_ = muted; |
| 919 | } |
| 920 | |
| 921 | bool muted() const { |
| 922 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 923 | return muted_; |
| 924 | } |
| 925 | |
Ivo Creusen | 56d46090 | 2017-11-24 16:29:59 | [diff] [blame] | 926 | webrtc::AudioSendStream::Stats GetStats(bool has_remote_tracks) const { |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 927 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 928 | RTC_DCHECK(stream_); |
Ivo Creusen | 56d46090 | 2017-11-24 16:29:59 | [diff] [blame] | 929 | return stream_->GetStats(has_remote_tracks); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 930 | } |
| 931 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 932 | // Starts the sending by setting ourselves as a sink to the AudioSource to |
| 933 | // get data callbacks. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 934 | // This method is called on the libjingle worker thread. |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 935 | // TODO(xians): Make sure Start() is called only once. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 936 | void SetSource(AudioSource* source) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 937 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 938 | RTC_DCHECK(source); |
| 939 | if (source_) { |
| 940 | RTC_DCHECK(source_ == source); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 941 | return; |
| 942 | } |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 943 | source->SetSink(this); |
| 944 | source_ = source; |
| 945 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 946 | } |
| 947 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 948 | // Stops sending by setting the sink of the AudioSource to nullptr. No data |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 949 | // callback will be received after this method. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 950 | // This method is called on the libjingle worker thread. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 951 | void ClearSource() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 952 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 953 | if (source_) { |
| 954 | source_->SetSink(nullptr); |
| 955 | source_ = nullptr; |
solenberg | 98c6886 | 2015-10-09 10:27:14 | [diff] [blame] | 956 | } |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 957 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 958 | } |
| 959 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 960 | // AudioSource::Sink implementation. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 961 | // This method is called on the audio thread. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 962 | void OnData(const void* audio_data, |
| 963 | int bits_per_sample, |
| 964 | int sample_rate, |
Peter Kasting | 6955870 | 2016-01-13 00:26:35 | [diff] [blame] | 965 | size_t number_of_channels, |
Peter Kasting | dce40cf | 2015-08-24 21:52:23 | [diff] [blame] | 966 | size_t number_of_frames) override { |
solenberg | 347ec5c | 2016-09-23 11:21:47 | [diff] [blame] | 967 | RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_); |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 968 | RTC_DCHECK(voe_audio_transport_); |
maxmorin | 1aee0b5 | 2016-08-15 18:46:19 | [diff] [blame] | 969 | voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data, |
| 970 | bits_per_sample, sample_rate, |
| 971 | number_of_channels, number_of_frames); |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 972 | } |
| 973 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 974 | // Callback from the |source_| when it is going away. In case Start() has |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 975 | // never been called, this callback won't be triggered. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 976 | void OnClose() override { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 977 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 978 | // Set |source_| to nullptr to make sure no more callback will get into |
| 979 | // the source. |
| 980 | source_ = nullptr; |
| 981 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 982 | } |
| 983 | |
| 984 | // Accessor to the VoE channel ID. |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 985 | int channel() const { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 986 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 987 | return config_.voe_channel_id; |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 988 | } |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 989 | |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 990 | const webrtc::RtpParameters& rtp_parameters() const { |
| 991 | return rtp_parameters_; |
| 992 | } |
| 993 | |
deadbeef | fb2aced | 2017-01-07 07:05:37 | [diff] [blame] | 994 | bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) { |
| 995 | if (rtp_parameters.encodings.size() != 1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 996 | RTC_LOG(LS_ERROR) |
deadbeef | fb2aced | 2017-01-07 07:05:37 | [diff] [blame] | 997 | << "Attempted to set RtpParameters without exactly one encoding"; |
| 998 | return false; |
| 999 | } |
| 1000 | if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1001 | RTC_LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC"; |
deadbeef | fb2aced | 2017-01-07 07:05:37 | [diff] [blame] | 1002 | return false; |
| 1003 | } |
| 1004 | return true; |
| 1005 | } |
| 1006 | |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1007 | bool SetRtpParameters(const webrtc::RtpParameters& parameters) { |
deadbeef | fb2aced | 2017-01-07 07:05:37 | [diff] [blame] | 1008 | if (!ValidateRtpParameters(parameters)) { |
| 1009 | return false; |
| 1010 | } |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1011 | |
| 1012 | rtc::Optional<int> send_rate; |
| 1013 | if (audio_codec_spec_) { |
| 1014 | send_rate = ComputeSendBitrate(max_send_bitrate_bps_, |
| 1015 | parameters.encodings[0].max_bitrate_bps, |
| 1016 | *audio_codec_spec_); |
| 1017 | if (!send_rate) { |
| 1018 | return false; |
| 1019 | } |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1020 | } |
| 1021 | |
minyue | cecec10 | 2017-03-27 20:04:25 | [diff] [blame] | 1022 | const rtc::Optional<int> old_rtp_max_bitrate = |
| 1023 | rtp_parameters_.encodings[0].max_bitrate_bps; |
| 1024 | |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1025 | rtp_parameters_ = parameters; |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1026 | |
minyue | cecec10 | 2017-03-27 20:04:25 | [diff] [blame] | 1027 | if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1028 | // Reconfigure AudioSendStream with new bit rate. |
| 1029 | if (send_rate) { |
| 1030 | config_.send_codec_spec->target_bitrate_bps = send_rate; |
| 1031 | } |
| 1032 | UpdateAllowedBitrateRange(); |
| 1033 | ReconfigureAudioSendStream(); |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1034 | } else { |
| 1035 | // parameters.encodings[0].active could have changed. |
| 1036 | UpdateSendState(); |
| 1037 | } |
| 1038 | return true; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1039 | } |
| 1040 | |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 1041 | private: |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1042 | void UpdateSendState() { |
| 1043 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1044 | RTC_DCHECK(stream_); |
Taylor Brandstetter | 55dd708 | 2016-05-03 20:50:11 | [diff] [blame] | 1045 | RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size()); |
| 1046 | if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1047 | stream_->Start(); |
| 1048 | } else { // !send || source_ = nullptr |
| 1049 | stream_->Stop(); |
| 1050 | } |
| 1051 | } |
| 1052 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1053 | void UpdateAllowedBitrateRange() { |
michaelt | 53fe19d | 2016-10-18 16:39:22 | [diff] [blame] | 1054 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1055 | const bool is_opus = |
| 1056 | config_.send_codec_spec && |
| 1057 | !STR_CASE_CMP(config_.send_codec_spec->format.name.c_str(), |
| 1058 | kOpusCodecName); |
| 1059 | if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) { |
stefan | e9f36d5 | 2017-01-24 16:18:45 | [diff] [blame] | 1060 | config_.min_bitrate_bps = kOpusMinBitrateBps; |
minyue | cecec10 | 2017-03-27 20:04:25 | [diff] [blame] | 1061 | |
| 1062 | // This means that when RtpParameters is reset, we may change the |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1063 | // encoder's bit rate immediately (through ReconfigureAudioSendStream()), |
minyue | cecec10 | 2017-03-27 20:04:25 | [diff] [blame] | 1064 | // meanwhile change the cap to the output of BWE. |
| 1065 | config_.max_bitrate_bps = |
| 1066 | rtp_parameters_.encodings[0].max_bitrate_bps |
| 1067 | ? *rtp_parameters_.encodings[0].max_bitrate_bps |
| 1068 | : kOpusBitrateFbBps; |
| 1069 | |
michaelt | 53fe19d | 2016-10-18 16:39:22 | [diff] [blame] | 1070 | // TODO(mflodman): Keep testing this and set proper values. |
| 1071 | // Note: This is an early experiment currently only supported by Opus. |
elad.alon | 0fe1216 | 2017-01-31 13:48:37 | [diff] [blame] | 1072 | if (send_side_bwe_with_overhead_) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1073 | const int max_packet_size_ms = |
| 1074 | WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60; |
michaelt | 6672b26 | 2017-01-11 18:17:59 | [diff] [blame] | 1075 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1076 | // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12) |
| 1077 | constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12; |
michaelt | 6672b26 | 2017-01-11 18:17:59 | [diff] [blame] | 1078 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1079 | int min_overhead_bps = |
| 1080 | kOverheadPerPacket * 8 * 1000 / max_packet_size_ms; |
michaelt | 6672b26 | 2017-01-11 18:17:59 | [diff] [blame] | 1081 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1082 | // We assume that |config_.max_bitrate_bps| before the next line is |
| 1083 | // a hard limit on the payload bitrate, so we add min_overhead_bps to |
| 1084 | // it to ensure that, when overhead is deducted, the payload rate |
| 1085 | // never goes beyond the limit. |
| 1086 | // Note: this also means that if a higher overhead is forced, we |
| 1087 | // cannot reach the limit. |
| 1088 | // TODO(minyue): Reconsider this when the signaling to BWE is done |
| 1089 | // through a dedicated API. |
| 1090 | config_.max_bitrate_bps += min_overhead_bps; |
michaelt | 6672b26 | 2017-01-11 18:17:59 | [diff] [blame] | 1091 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1092 | // In contrast to max_bitrate_bps, we let min_bitrate_bps always be |
| 1093 | // reachable. |
| 1094 | config_.min_bitrate_bps += min_overhead_bps; |
michaelt | 6672b26 | 2017-01-11 18:17:59 | [diff] [blame] | 1095 | } |
michaelt | 53fe19d | 2016-10-18 16:39:22 | [diff] [blame] | 1096 | } |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1097 | } |
| 1098 | |
| 1099 | void UpdateSendCodecSpec( |
| 1100 | const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) { |
| 1101 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1102 | config_.rtp.nack.rtp_history_ms = |
| 1103 | send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0; |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1104 | config_.send_codec_spec = send_codec_spec; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1105 | auto info = |
| 1106 | config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format); |
| 1107 | RTC_DCHECK(info); |
| 1108 | // If a specific target bitrate has been set for the stream, use that as |
| 1109 | // the new default bitrate when computing send bitrate. |
| 1110 | if (send_codec_spec.target_bitrate_bps) { |
| 1111 | info->default_bitrate_bps = std::max( |
| 1112 | info->min_bitrate_bps, |
| 1113 | std::min(info->max_bitrate_bps, *send_codec_spec.target_bitrate_bps)); |
| 1114 | } |
| 1115 | |
| 1116 | audio_codec_spec_.emplace( |
| 1117 | webrtc::AudioCodecSpec{send_codec_spec.format, *info}); |
| 1118 | |
| 1119 | config_.send_codec_spec->target_bitrate_bps = ComputeSendBitrate( |
| 1120 | max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps, |
| 1121 | *audio_codec_spec_); |
| 1122 | |
| 1123 | UpdateAllowedBitrateRange(); |
| 1124 | } |
| 1125 | |
| 1126 | void ReconfigureAudioSendStream() { |
| 1127 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1128 | RTC_DCHECK(stream_); |
| 1129 | stream_->Reconfigure(config_); |
michaelt | 53fe19d | 2016-10-18 16:39:22 | [diff] [blame] | 1130 | } |
| 1131 | |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1132 | rtc::ThreadChecker worker_thread_checker_; |
solenberg | 347ec5c | 2016-09-23 11:21:47 | [diff] [blame] | 1133 | rtc::RaceChecker audio_capture_race_checker_; |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1134 | webrtc::AudioTransport* const voe_audio_transport_ = nullptr; |
| 1135 | webrtc::Call* call_ = nullptr; |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1136 | webrtc::AudioSendStream::Config config_; |
elad.alon | 0fe1216 | 2017-01-31 13:48:37 | [diff] [blame] | 1137 | const bool send_side_bwe_with_overhead_; |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1138 | // The stream is owned by WebRtcAudioSendStream and may be reallocated if |
| 1139 | // configuration changes. |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1140 | webrtc::AudioSendStream* stream_ = nullptr; |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 1141 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1142 | // Raw pointer to AudioSource owned by LocalAudioTrackHandler. |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 1143 | // PeerConnection will make sure invalidating the pointer before the object |
| 1144 | // goes away. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1145 | AudioSource* source_ = nullptr; |
| 1146 | bool send_ = false; |
solenberg | 9421853 | 2016-06-16 17:53:22 | [diff] [blame] | 1147 | bool muted_ = false; |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1148 | int max_send_bitrate_bps_; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1149 | webrtc::RtpParameters rtp_parameters_; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1150 | rtc::Optional<webrtc::AudioCodecSpec> audio_codec_spec_; |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 | [diff] [blame] | 1151 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1152 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream); |
| 1153 | }; |
| 1154 | |
| 1155 | class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { |
| 1156 | public: |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 1157 | WebRtcAudioReceiveStream( |
| 1158 | int ch, |
| 1159 | uint32_t remote_ssrc, |
| 1160 | uint32_t local_ssrc, |
| 1161 | bool use_transport_cc, |
solenberg | 8189b02 | 2016-06-14 19:13:00 | [diff] [blame] | 1162 | bool use_nack, |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 1163 | const std::string& sync_group, |
| 1164 | const std::vector<webrtc::RtpExtension>& extensions, |
| 1165 | webrtc::Call* call, |
| 1166 | webrtc::Transport* rtcp_send_transport, |
kwiberg | 1c07c70 | 2017-03-27 14:15:49 | [diff] [blame] | 1167 | const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
| 1168 | const std::map<int, webrtc::SdpAudioFormat>& decoder_map) |
stefan | ba4c0e4 | 2016-02-04 12:12:24 | [diff] [blame] | 1169 | : call_(call), config_() { |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1170 | RTC_DCHECK_GE(ch, 0); |
| 1171 | RTC_DCHECK(call); |
| 1172 | config_.rtp.remote_ssrc = remote_ssrc; |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1173 | config_.rtp.local_ssrc = local_ssrc; |
| 1174 | config_.rtp.transport_cc = use_transport_cc; |
| 1175 | config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0; |
| 1176 | config_.rtp.extensions = extensions; |
solenberg | 31fec40 | 2016-05-06 09:13:12 | [diff] [blame] | 1177 | config_.rtcp_send_transport = rtcp_send_transport; |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1178 | config_.voe_channel_id = ch; |
| 1179 | config_.sync_group = sync_group; |
ossu | 29b1a8d | 2016-06-13 14:34:51 | [diff] [blame] | 1180 | config_.decoder_factory = decoder_factory; |
kwiberg | 1c07c70 | 2017-03-27 14:15:49 | [diff] [blame] | 1181 | config_.decoder_map = decoder_map; |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1182 | RecreateAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1183 | } |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1184 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1185 | ~WebRtcAudioReceiveStream() { |
| 1186 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1187 | call_->DestroyAudioReceiveStream(stream_); |
| 1188 | } |
| 1189 | |
solenberg | 4a0f7b5 | 2016-06-16 20:07:33 | [diff] [blame] | 1190 | void RecreateAudioReceiveStream(uint32_t local_ssrc) { |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1191 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1192 | config_.rtp.local_ssrc = local_ssrc; |
| 1193 | RecreateAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1194 | } |
solenberg | 8189b02 | 2016-06-14 19:13:00 | [diff] [blame] | 1195 | |
| 1196 | void RecreateAudioReceiveStream(bool use_transport_cc, bool use_nack) { |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1197 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1198 | config_.rtp.transport_cc = use_transport_cc; |
| 1199 | config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0; |
| 1200 | RecreateAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1201 | } |
| 1202 | |
solenberg | 4a0f7b5 | 2016-06-16 20:07:33 | [diff] [blame] | 1203 | void RecreateAudioReceiveStream( |
| 1204 | const std::vector<webrtc::RtpExtension>& extensions) { |
| 1205 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1206 | config_.rtp.extensions = extensions; |
| 1207 | RecreateAudioReceiveStream(); |
| 1208 | } |
| 1209 | |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1210 | // Set a new payload type -> decoder map. |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1211 | void RecreateAudioReceiveStream( |
| 1212 | const std::map<int, webrtc::SdpAudioFormat>& decoder_map) { |
| 1213 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1214 | config_.decoder_map = decoder_map; |
| 1215 | RecreateAudioReceiveStream(); |
solenberg | 4a0f7b5 | 2016-06-16 20:07:33 | [diff] [blame] | 1216 | } |
| 1217 | |
solenberg | 4904fb6 | 2017-02-17 20:01:14 | [diff] [blame] | 1218 | void MaybeRecreateAudioReceiveStream(const std::string& sync_group) { |
| 1219 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1220 | if (config_.sync_group != sync_group) { |
| 1221 | config_.sync_group = sync_group; |
| 1222 | RecreateAudioReceiveStream(); |
| 1223 | } |
| 1224 | } |
| 1225 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1226 | webrtc::AudioReceiveStream::Stats GetStats() const { |
| 1227 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1228 | RTC_DCHECK(stream_); |
| 1229 | return stream_->GetStats(); |
| 1230 | } |
| 1231 | |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 1232 | int GetOutputLevel() const { |
| 1233 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1234 | RTC_DCHECK(stream_); |
| 1235 | return stream_->GetOutputLevel(); |
| 1236 | } |
| 1237 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1238 | int channel() const { |
| 1239 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1240 | return config_.voe_channel_id; |
| 1241 | } |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1242 | |
kwiberg | 686a8ef | 2016-02-26 11:00:35 | [diff] [blame] | 1243 | void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 1244 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | 686a8ef | 2016-02-26 11:00:35 | [diff] [blame] | 1245 | stream_->SetSink(std::move(sink)); |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 1246 | } |
| 1247 | |
solenberg | 217fb66 | 2016-06-17 15:30:54 | [diff] [blame] | 1248 | void SetOutputVolume(double volume) { |
| 1249 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1250 | stream_->SetGain(volume); |
| 1251 | } |
| 1252 | |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1253 | void SetPlayout(bool playout) { |
| 1254 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1255 | RTC_DCHECK(stream_); |
| 1256 | if (playout) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1257 | RTC_LOG(LS_INFO) << "Starting playout for channel #" << channel(); |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1258 | stream_->Start(); |
| 1259 | } else { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1260 | RTC_LOG(LS_INFO) << "Stopping playout for channel #" << channel(); |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1261 | stream_->Stop(); |
| 1262 | } |
aleloi | 18e0b67 | 2016-10-04 09:45:47 | [diff] [blame] | 1263 | playout_ = playout; |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1264 | } |
| 1265 | |
hbos | 8d609f6 | 2017-04-10 14:39:05 | [diff] [blame] | 1266 | std::vector<webrtc::RtpSource> GetSources() { |
| 1267 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1268 | RTC_DCHECK(stream_); |
| 1269 | return stream_->GetSources(); |
| 1270 | } |
| 1271 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1272 | private: |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1273 | void RecreateAudioReceiveStream() { |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1274 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1275 | if (stream_) { |
| 1276 | call_->DestroyAudioReceiveStream(stream_); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1277 | } |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1278 | stream_ = call_->CreateAudioReceiveStream(config_); |
| 1279 | RTC_CHECK(stream_); |
aleloi | 18e0b67 | 2016-10-04 09:45:47 | [diff] [blame] | 1280 | SetPlayout(playout_); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1281 | } |
| 1282 | |
| 1283 | rtc::ThreadChecker worker_thread_checker_; |
| 1284 | webrtc::Call* call_ = nullptr; |
| 1285 | webrtc::AudioReceiveStream::Config config_; |
| 1286 | // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if |
| 1287 | // configuration changes. |
| 1288 | webrtc::AudioReceiveStream* stream_ = nullptr; |
aleloi | 18e0b67 | 2016-10-04 09:45:47 | [diff] [blame] | 1289 | bool playout_ = false; |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1290 | |
| 1291 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1292 | }; |
| 1293 | |
Fredrik Solenberg | 709ed67 | 2015-09-15 10:26:33 | [diff] [blame] | 1294 | WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, |
nisse | 51542be | 2016-02-12 10:27:06 | [diff] [blame] | 1295 | const MediaConfig& config, |
Fredrik Solenberg | b071a19 | 2015-09-17 14:42:56 | [diff] [blame] | 1296 | const AudioOptions& options, |
Fredrik Solenberg | 709ed67 | 2015-09-15 10:26:33 | [diff] [blame] | 1297 | webrtc::Call* call) |
nisse | 51542be | 2016-02-12 10:27:06 | [diff] [blame] | 1298 | : VoiceMediaChannel(config), engine_(engine), call_(call) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1299 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel"; |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1300 | RTC_DCHECK(call); |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1301 | engine->RegisterChannel(this); |
Fredrik Solenberg | b071a19 | 2015-09-17 14:42:56 | [diff] [blame] | 1302 | SetOptions(options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1303 | } |
| 1304 | |
| 1305 | WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1306 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1307 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel"; |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1308 | // TODO(solenberg): Should be able to delete the streams directly, without |
| 1309 | // going through RemoveNnStream(), once stream objects handle |
| 1310 | // all (de)configuration. |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1311 | while (!send_streams_.empty()) { |
| 1312 | RemoveSendStream(send_streams_.begin()->first); |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 1313 | } |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1314 | while (!recv_streams_.empty()) { |
| 1315 | RemoveRecvStream(recv_streams_.begin()->first); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1316 | } |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1317 | engine()->UnregisterChannel(this); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1318 | } |
| 1319 | |
nisse | 51542be | 2016-02-12 10:27:06 | [diff] [blame] | 1320 | rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const { |
| 1321 | return kAudioDscpValue; |
| 1322 | } |
| 1323 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 23:05:34 | [diff] [blame] | 1324 | bool WebRtcVoiceMediaChannel::SetSendParameters( |
| 1325 | const AudioSendParameters& params) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1326 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1327 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1328 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: " |
| 1329 | << params.ToString(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 23:05:34 | [diff] [blame] | 1330 | // TODO(pthatcher): Refactor this to be more clean now that we have |
| 1331 | // all the information at once. |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1332 | |
| 1333 | if (!SetSendCodecs(params.codecs)) { |
| 1334 | return false; |
| 1335 | } |
| 1336 | |
solenberg | 7e4e01a | 2015-12-02 16:05:01 | [diff] [blame] | 1337 | if (!ValidateRtpExtensions(params.extensions)) { |
| 1338 | return false; |
| 1339 | } |
| 1340 | std::vector<webrtc::RtpExtension> filtered_extensions = |
| 1341 | FilterRtpExtensions(params.extensions, |
| 1342 | webrtc::RtpExtension::IsSupportedForAudio, true); |
| 1343 | if (send_rtp_extensions_ != filtered_extensions) { |
| 1344 | send_rtp_extensions_.swap(filtered_extensions); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1345 | for (auto& it : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1346 | it.second->SetRtpExtensions(send_rtp_extensions_); |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1347 | } |
| 1348 | } |
| 1349 | |
deadbeef | 8034614 | 2016-04-27 21:17:10 | [diff] [blame] | 1350 | if (!SetMaxSendBitrate(params.max_bandwidth_bps)) { |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1351 | return false; |
| 1352 | } |
| 1353 | return SetOptions(params.options); |
Peter Thatcher | c2ee2c8 | 2015-08-07 23:05:34 | [diff] [blame] | 1354 | } |
| 1355 | |
| 1356 | bool WebRtcVoiceMediaChannel::SetRecvParameters( |
| 1357 | const AudioRecvParameters& params) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1358 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1359 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1360 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: " |
| 1361 | << params.ToString(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 23:05:34 | [diff] [blame] | 1362 | // TODO(pthatcher): Refactor this to be more clean now that we have |
| 1363 | // all the information at once. |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1364 | |
| 1365 | if (!SetRecvCodecs(params.codecs)) { |
| 1366 | return false; |
| 1367 | } |
| 1368 | |
solenberg | 7e4e01a | 2015-12-02 16:05:01 | [diff] [blame] | 1369 | if (!ValidateRtpExtensions(params.extensions)) { |
| 1370 | return false; |
| 1371 | } |
| 1372 | std::vector<webrtc::RtpExtension> filtered_extensions = |
| 1373 | FilterRtpExtensions(params.extensions, |
| 1374 | webrtc::RtpExtension::IsSupportedForAudio, false); |
| 1375 | if (recv_rtp_extensions_ != filtered_extensions) { |
| 1376 | recv_rtp_extensions_.swap(filtered_extensions); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1377 | for (auto& it : recv_streams_) { |
| 1378 | it.second->RecreateAudioReceiveStream(recv_rtp_extensions_); |
| 1379 | } |
| 1380 | } |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1381 | return true; |
Peter Thatcher | c2ee2c8 | 2015-08-07 23:05:34 | [diff] [blame] | 1382 | } |
| 1383 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1384 | webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters( |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1385 | uint32_t ssrc) const { |
| 1386 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1387 | auto it = send_streams_.find(ssrc); |
| 1388 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1389 | RTC_LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream " |
| 1390 | << "with ssrc " << ssrc << " which doesn't exist."; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1391 | return webrtc::RtpParameters(); |
| 1392 | } |
| 1393 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 23:23:10 | [diff] [blame] | 1394 | webrtc::RtpParameters rtp_params = it->second->rtp_parameters(); |
| 1395 | // Need to add the common list of codecs to the send stream-specific |
| 1396 | // RTP parameters. |
| 1397 | for (const AudioCodec& codec : send_codecs_) { |
| 1398 | rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| 1399 | } |
| 1400 | return rtp_params; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1401 | } |
| 1402 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1403 | bool WebRtcVoiceMediaChannel::SetRtpSendParameters( |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1404 | uint32_t ssrc, |
| 1405 | const webrtc::RtpParameters& parameters) { |
| 1406 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1407 | auto it = send_streams_.find(ssrc); |
| 1408 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1409 | RTC_LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream " |
| 1410 | << "with ssrc " << ssrc << " which doesn't exist."; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1411 | return false; |
| 1412 | } |
| 1413 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1414 | // TODO(deadbeef): Handle setting parameters with a list of codecs in a |
| 1415 | // different order (which should change the send codec). |
| 1416 | webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); |
| 1417 | if (current_parameters.codecs != parameters.codecs) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1418 | RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs " |
| 1419 | << "is not currently supported."; |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1420 | return false; |
| 1421 | } |
| 1422 | |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1423 | // TODO(minyue): The following legacy actions go into |
| 1424 | // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end, |
| 1425 | // though there are two difference: |
| 1426 | // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls |
| 1427 | // |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls |
| 1428 | // |SetSendCodecs|. The outcome should be the same. |
| 1429 | // 2. AudioSendStream can be recreated. |
| 1430 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 23:23:10 | [diff] [blame] | 1431 | // Codecs are handled at the WebRtcVoiceMediaChannel level. |
| 1432 | webrtc::RtpParameters reduced_params = parameters; |
| 1433 | reduced_params.codecs.clear(); |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 1434 | return it->second->SetRtpParameters(reduced_params); |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1435 | } |
| 1436 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1437 | webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters( |
| 1438 | uint32_t ssrc) const { |
| 1439 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1440 | webrtc::RtpParameters rtp_params; |
| 1441 | // SSRC of 0 represents the default receive stream. |
| 1442 | if (ssrc == 0) { |
| 1443 | if (!default_sink_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1444 | RTC_LOG(LS_WARNING) |
| 1445 | << "Attempting to get RTP parameters for the default, " |
| 1446 | "unsignaled audio receive stream, but not yet " |
| 1447 | "configured to receive such a stream."; |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1448 | return rtp_params; |
| 1449 | } |
| 1450 | rtp_params.encodings.emplace_back(); |
| 1451 | } else { |
| 1452 | auto it = recv_streams_.find(ssrc); |
| 1453 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1454 | RTC_LOG(LS_WARNING) |
| 1455 | << "Attempting to get RTP receive parameters for stream " |
| 1456 | << "with ssrc " << ssrc << " which doesn't exist."; |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1457 | return webrtc::RtpParameters(); |
| 1458 | } |
| 1459 | rtp_params.encodings.emplace_back(); |
| 1460 | // TODO(deadbeef): Return stream-specific parameters. |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1461 | rtp_params.encodings[0].ssrc = ssrc; |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1462 | } |
| 1463 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1464 | for (const AudioCodec& codec : recv_codecs_) { |
| 1465 | rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| 1466 | } |
| 1467 | return rtp_params; |
| 1468 | } |
| 1469 | |
| 1470 | bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters( |
| 1471 | uint32_t ssrc, |
| 1472 | const webrtc::RtpParameters& parameters) { |
| 1473 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1474 | // SSRC of 0 represents the default receive stream. |
| 1475 | if (ssrc == 0) { |
| 1476 | if (!default_sink_) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1477 | RTC_LOG(LS_WARNING) |
| 1478 | << "Attempting to set RTP parameters for the default, " |
| 1479 | "unsignaled audio receive stream, but not yet " |
| 1480 | "configured to receive such a stream."; |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1481 | return false; |
| 1482 | } |
| 1483 | } else { |
| 1484 | auto it = recv_streams_.find(ssrc); |
| 1485 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1486 | RTC_LOG(LS_WARNING) |
| 1487 | << "Attempting to set RTP receive parameters for stream " |
| 1488 | << "with ssrc " << ssrc << " which doesn't exist."; |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 1489 | return false; |
| 1490 | } |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1491 | } |
| 1492 | |
| 1493 | webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); |
| 1494 | if (current_parameters != parameters) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1495 | RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently " |
| 1496 | << "unsupported."; |
Taylor Brandstetter | db0cd9e | 2016-05-16 18:40:30 | [diff] [blame] | 1497 | return false; |
| 1498 | } |
| 1499 | return true; |
| 1500 | } |
| 1501 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1502 | bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1503 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1504 | RTC_LOG(LS_INFO) << "Setting voice channel options: " << options.ToString(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1505 | |
| 1506 | // We retain all of the existing options, and apply the given ones |
| 1507 | // on top. This means there is no way to "clear" options such that |
| 1508 | // they go back to the engine default. |
| 1509 | options_.SetAll(options); |
solenberg | 246b817 | 2015-12-08 17:50:23 | [diff] [blame] | 1510 | if (!engine()->ApplyOptions(options_)) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1511 | RTC_LOG(LS_WARNING) |
| 1512 | << "Failed to apply engine options during channel SetOptions."; |
solenberg | 246b817 | 2015-12-08 17:50:23 | [diff] [blame] | 1513 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1514 | } |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 1515 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1516 | rtc::Optional<std::string> audio_network_adaptor_config = |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 1517 | GetAudioNetworkAdaptorConfig(options_); |
| 1518 | for (auto& it : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1519 | it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config); |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 1520 | } |
| 1521 | |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1522 | RTC_LOG(LS_INFO) << "Set voice channel options. Current options: " |
| 1523 | << options_.ToString(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1524 | return true; |
| 1525 | } |
| 1526 | |
| 1527 | bool WebRtcVoiceMediaChannel::SetRecvCodecs( |
| 1528 | const std::vector<AudioCodec>& codecs) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1529 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 8fb30c3 | 2015-10-13 10:06:58 | [diff] [blame] | 1530 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1531 | // Set the payload types to be used for incoming media. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1532 | RTC_LOG(LS_INFO) << "Setting receive voice codecs."; |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1533 | |
| 1534 | if (!VerifyUniquePayloadTypes(codecs)) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1535 | RTC_LOG(LS_ERROR) << "Codec payload types overlap."; |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1536 | return false; |
| 1537 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1538 | |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1539 | // Create a payload type -> SdpAudioFormat map with all the decoders. Fail |
| 1540 | // unless the factory claims to support all decoders. |
| 1541 | std::map<int, webrtc::SdpAudioFormat> decoder_map; |
| 1542 | for (const AudioCodec& codec : codecs) { |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1543 | // Log a warning if a codec's payload type is changing. This used to be |
| 1544 | // treated as an error. It's abnormal, but not really illegal. |
| 1545 | AudioCodec old_codec; |
| 1546 | if (FindCodec(recv_codecs_, codec, &old_codec) && |
| 1547 | old_codec.id != codec.id) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1548 | RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type (" |
| 1549 | << codec.id << ", was already mapped to " |
| 1550 | << old_codec.id << ")"; |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1551 | } |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1552 | auto format = AudioCodecToSdpAudioFormat(codec); |
| 1553 | if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") && |
| 1554 | !engine()->decoder_factory_->IsSupportedDecoder(format)) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1555 | RTC_LOG(LS_ERROR) << "Unsupported codec: " << format; |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1556 | return false; |
| 1557 | } |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1558 | // We allow adding new codecs but don't allow changing the payload type of |
| 1559 | // codecs that are already configured since we might already be receiving |
| 1560 | // packets with that payload type. See RFC3264, Section 8.3.2. |
| 1561 | // TODO(deadbeef): Also need to check for clashes with previously mapped |
| 1562 | // payload types, and not just currently mapped ones. For example, this |
| 1563 | // should be illegal: |
| 1564 | // 1. {100: opus/48000/2, 101: ISAC/16000} |
| 1565 | // 2. {100: opus/48000/2} |
| 1566 | // 3. {100: opus/48000/2, 101: ISAC/32000} |
| 1567 | // Though this check really should happen at a higher level, since this |
| 1568 | // conflict could happen between audio and video codecs. |
| 1569 | auto existing = decoder_map_.find(codec.id); |
| 1570 | if (existing != decoder_map_.end() && !existing->second.Matches(format)) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1571 | RTC_LOG(LS_ERROR) << "Attempting to use payload type " << codec.id |
| 1572 | << " for " << codec.name |
| 1573 | << ", but it is already used for " |
| 1574 | << existing->second.name; |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1575 | return false; |
| 1576 | } |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1577 | decoder_map.insert({codec.id, std::move(format)}); |
| 1578 | } |
| 1579 | |
deadbeef | cb38367 | 2017-04-26 23:28:42 | [diff] [blame] | 1580 | if (decoder_map == decoder_map_) { |
| 1581 | // There's nothing new to configure. |
| 1582 | return true; |
| 1583 | } |
| 1584 | |
kwiberg | 37b8b11 | 2016-11-03 09:46:53 | [diff] [blame] | 1585 | if (playout_) { |
| 1586 | // Receive codecs can not be changed while playing. So we temporarily |
| 1587 | // pause playout. |
| 1588 | ChangePlayout(false); |
| 1589 | } |
| 1590 | |
kwiberg | 1c07c70 | 2017-03-27 14:15:49 | [diff] [blame] | 1591 | decoder_map_ = std::move(decoder_map); |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1592 | for (auto& kv : recv_streams_) { |
kwiberg | 1c07c70 | 2017-03-27 14:15:49 | [diff] [blame] | 1593 | kv.second->RecreateAudioReceiveStream(decoder_map_); |
solenberg | 26c8c91 | 2015-11-27 12:00:25 | [diff] [blame] | 1594 | } |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1595 | recv_codecs_ = codecs; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1596 | |
kwiberg | 37b8b11 | 2016-11-03 09:46:53 | [diff] [blame] | 1597 | if (desired_playout_ && !playout_) { |
| 1598 | ChangePlayout(desired_playout_); |
| 1599 | } |
kwiberg | d32bf75 | 2017-01-19 15:03:59 | [diff] [blame] | 1600 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1601 | } |
| 1602 | |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1603 | // Utility function called from SetSendParameters() to extract current send |
| 1604 | // codec settings from the given list of codecs (originally from SDP). Both send |
| 1605 | // and receive streams may be reconfigured based on the new settings. |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 | [diff] [blame] | 1606 | bool WebRtcVoiceMediaChannel::SetSendCodecs( |
| 1607 | const std::vector<AudioCodec>& codecs) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1608 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1609 | dtmf_payload_type_ = rtc::nullopt; |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1610 | dtmf_payload_freq_ = -1; |
| 1611 | |
| 1612 | // Validate supplied codecs list. |
| 1613 | for (const AudioCodec& codec : codecs) { |
| 1614 | // TODO(solenberg): Validate more aspects of input - that payload types |
| 1615 | // don't overlap, remove redundant/unsupported codecs etc - |
| 1616 | // the same way it is done for RtpHeaderExtensions. |
| 1617 | if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1618 | RTC_LOG(LS_WARNING) << "Codec payload type out of range: " |
| 1619 | << ToString(codec); |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1620 | return false; |
| 1621 | } |
| 1622 | } |
| 1623 | |
| 1624 | // Find PT of telephone-event codec with lowest clockrate, as a fallback, in |
| 1625 | // case we don't have a DTMF codec with a rate matching the send codec's, or |
| 1626 | // if this function returns early. |
| 1627 | std::vector<AudioCodec> dtmf_codecs; |
Fredrik Solenberg | af9fb21 | 2015-08-26 08:45:53 | [diff] [blame] | 1628 | for (const AudioCodec& codec : codecs) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 08:45:53 | [diff] [blame] | 1629 | if (IsCodec(codec, kDtmfCodecName)) { |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1630 | dtmf_codecs.push_back(codec); |
| 1631 | if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1632 | dtmf_payload_type_ = codec.id; |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1633 | dtmf_payload_freq_ = codec.clockrate; |
solenberg | 31642aa | 2016-03-14 15:00:37 | [diff] [blame] | 1634 | } |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 | [diff] [blame] | 1635 | } |
| 1636 | } |
| 1637 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1638 | // Scan through the list to figure out the codec to use for sending. |
| 1639 | rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> send_codec_spec; |
stefan | 1ccf73f | 2017-03-27 10:51:18 | [diff] [blame] | 1640 | webrtc::Call::Config::BitrateConfig bitrate_config; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1641 | rtc::Optional<webrtc::AudioCodecInfo> voice_codec_info; |
| 1642 | for (const AudioCodec& voice_codec : codecs) { |
| 1643 | if (!(IsCodec(voice_codec, kCnCodecName) || |
| 1644 | IsCodec(voice_codec, kDtmfCodecName) || |
| 1645 | IsCodec(voice_codec, kRedCodecName))) { |
| 1646 | webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate, |
| 1647 | voice_codec.channels, voice_codec.params); |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1648 | |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1649 | voice_codec_info = engine()->encoder_factory_->QueryAudioEncoder(format); |
| 1650 | if (!voice_codec_info) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1651 | RTC_LOG(LS_WARNING) << "Unknown codec " << ToString(voice_codec); |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1652 | continue; |
| 1653 | } |
| 1654 | |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1655 | send_codec_spec = webrtc::AudioSendStream::Config::SendCodecSpec( |
| 1656 | voice_codec.id, format); |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1657 | if (voice_codec.bitrate > 0) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1658 | send_codec_spec->target_bitrate_bps = voice_codec.bitrate; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1659 | } |
| 1660 | send_codec_spec->transport_cc_enabled = HasTransportCc(voice_codec); |
| 1661 | send_codec_spec->nack_enabled = HasNack(voice_codec); |
| 1662 | bitrate_config = GetBitrateConfigForCodec(voice_codec); |
| 1663 | break; |
| 1664 | } |
| 1665 | } |
| 1666 | |
| 1667 | if (!send_codec_spec) { |
| 1668 | return false; |
| 1669 | } |
| 1670 | |
| 1671 | RTC_DCHECK(voice_codec_info); |
| 1672 | if (voice_codec_info->allow_comfort_noise) { |
| 1673 | // Loop through the codecs list again to find the CN codec. |
| 1674 | // TODO(solenberg): Break out into a separate function? |
| 1675 | for (const AudioCodec& cn_codec : codecs) { |
ossu | 0c4b849 | 2017-03-02 19:03:25 | [diff] [blame] | 1676 | if (IsCodec(cn_codec, kCnCodecName) && |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1677 | cn_codec.clockrate == send_codec_spec->format.clockrate_hz) { |
ossu | 0c4b849 | 2017-03-02 19:03:25 | [diff] [blame] | 1678 | switch (cn_codec.clockrate) { |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1679 | case 8000: |
| 1680 | case 16000: |
| 1681 | case 32000: |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1682 | send_codec_spec->cng_payload_type = cn_codec.id; |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1683 | break; |
| 1684 | default: |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1685 | RTC_LOG(LS_WARNING) |
| 1686 | << "CN frequency " << cn_codec.clockrate << " not supported."; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1687 | break; |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1688 | } |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1689 | break; |
| 1690 | } |
| 1691 | } |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1692 | |
| 1693 | // Find the telephone-event PT exactly matching the preferred send codec. |
| 1694 | for (const AudioCodec& dtmf_codec : dtmf_codecs) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1695 | if (dtmf_codec.clockrate == send_codec_spec->format.clockrate_hz) { |
Oskar Sundbom | 7880758 | 2017-11-16 10:09:55 | [diff] [blame] | 1696 | dtmf_payload_type_ = dtmf_codec.id; |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 1697 | dtmf_payload_freq_ = dtmf_codec.clockrate; |
| 1698 | break; |
| 1699 | } |
| 1700 | } |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1701 | } |
| 1702 | |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 1703 | if (send_codec_spec_ != send_codec_spec) { |
| 1704 | send_codec_spec_ = std::move(send_codec_spec); |
stefan | 13f1a0a | 2016-11-30 15:22:58 | [diff] [blame] | 1705 | // Apply new settings to all streams. |
solenberg | 971cab0 | 2016-06-14 17:02:41 | [diff] [blame] | 1706 | for (const auto& kv : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1707 | kv.second->SetSendCodecSpec(*send_codec_spec_); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 | [diff] [blame] | 1708 | } |
stefan | 13f1a0a | 2016-11-30 15:22:58 | [diff] [blame] | 1709 | } else { |
| 1710 | // If the codec isn't changing, set the start bitrate to -1 which means |
| 1711 | // "unchanged" so that BWE isn't affected. |
stefan | 1ccf73f | 2017-03-27 10:51:18 | [diff] [blame] | 1712 | bitrate_config.start_bitrate_bps = -1; |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 | [diff] [blame] | 1713 | } |
stefan | 1ccf73f | 2017-03-27 10:51:18 | [diff] [blame] | 1714 | call_->SetBitrateConfig(bitrate_config); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 | [diff] [blame] | 1715 | |
solenberg | 8189b02 | 2016-06-14 19:13:00 | [diff] [blame] | 1716 | // Check if the transport cc feedback or NACK status has changed on the |
| 1717 | // preferred send codec, and in that case reconfigure all receive streams. |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1718 | if (recv_transport_cc_enabled_ != send_codec_spec_->transport_cc_enabled || |
| 1719 | recv_nack_enabled_ != send_codec_spec_->nack_enabled) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1720 | RTC_LOG(LS_INFO) << "Recreate all the receive streams because the send " |
| 1721 | "codec has changed."; |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1722 | recv_transport_cc_enabled_ = send_codec_spec_->transport_cc_enabled; |
| 1723 | recv_nack_enabled_ = send_codec_spec_->nack_enabled; |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1724 | for (auto& kv : recv_streams_) { |
solenberg | 8189b02 | 2016-06-14 19:13:00 | [diff] [blame] | 1725 | kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_, |
| 1726 | recv_nack_enabled_); |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1727 | } |
| 1728 | } |
| 1729 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 23:23:10 | [diff] [blame] | 1730 | send_codecs_ = codecs; |
solenberg | 72e29d2 | 2016-03-08 14:35:16 | [diff] [blame] | 1731 | return true; |
| 1732 | } |
| 1733 | |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1734 | void WebRtcVoiceMediaChannel::SetPlayout(bool playout) { |
kwiberg | 37b8b11 | 2016-11-03 09:46:53 | [diff] [blame] | 1735 | desired_playout_ = playout; |
| 1736 | return ChangePlayout(desired_playout_); |
| 1737 | } |
| 1738 | |
| 1739 | void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) { |
| 1740 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1741 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1742 | if (playout_ == playout) { |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1743 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1744 | } |
| 1745 | |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1746 | for (const auto& kv : recv_streams_) { |
| 1747 | kv.second->SetPlayout(playout); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1748 | } |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 1749 | playout_ = playout; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1750 | } |
| 1751 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1752 | void WebRtcVoiceMediaChannel::SetSend(bool send) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1753 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1754 | if (send_ == send) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1755 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1756 | } |
| 1757 | |
solenberg | d53a3f9 | 2016-04-14 20:56:37 | [diff] [blame] | 1758 | // Apply channel specific options, and initialize the ADM for recording (this |
| 1759 | // may take time on some platforms, e.g. Android). |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1760 | if (send) { |
solenberg | 63b3454 | 2015-09-29 13:06:31 | [diff] [blame] | 1761 | engine()->ApplyOptions(options_); |
solenberg | d53a3f9 | 2016-04-14 20:56:37 | [diff] [blame] | 1762 | |
| 1763 | // InitRecording() may return an error if the ADM is already recording. |
| 1764 | if (!engine()->adm()->RecordingIsInitialized() && |
| 1765 | !engine()->adm()->Recording()) { |
| 1766 | if (engine()->adm()->InitRecording() != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1767 | RTC_LOG(LS_WARNING) << "Failed to initialize recording"; |
solenberg | d53a3f9 | 2016-04-14 20:56:37 | [diff] [blame] | 1768 | } |
| 1769 | } |
solenberg | 63b3454 | 2015-09-29 13:06:31 | [diff] [blame] | 1770 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1771 | |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1772 | // Change the settings on each send channel. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1773 | for (auto& kv : send_streams_) { |
| 1774 | kv.second->SetSend(send); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1775 | } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1776 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1777 | send_ = send; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1778 | } |
| 1779 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 1780 | bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, |
| 1781 | bool enable, |
solenberg | 1dd98f3 | 2015-09-10 08:57:14 | [diff] [blame] | 1782 | const AudioOptions* options, |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1783 | AudioSource* source) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1784 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 1dd98f3 | 2015-09-10 08:57:14 | [diff] [blame] | 1785 | // TODO(solenberg): The state change should be fully rolled back if any one of |
| 1786 | // these calls fail. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1787 | if (!SetLocalSource(ssrc, source)) { |
solenberg | 1dd98f3 | 2015-09-10 08:57:14 | [diff] [blame] | 1788 | return false; |
| 1789 | } |
solenberg | dfc8f4f | 2015-10-01 09:31:10 | [diff] [blame] | 1790 | if (!MuteStream(ssrc, !enable)) { |
solenberg | 1dd98f3 | 2015-09-10 08:57:14 | [diff] [blame] | 1791 | return false; |
| 1792 | } |
solenberg | dfc8f4f | 2015-10-01 09:31:10 | [diff] [blame] | 1793 | if (enable && options) { |
solenberg | 1dd98f3 | 2015-09-10 08:57:14 | [diff] [blame] | 1794 | return SetOptions(*options); |
| 1795 | } |
| 1796 | return true; |
| 1797 | } |
| 1798 | |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1799 | int WebRtcVoiceMediaChannel::CreateVoEChannel() { |
| 1800 | int id = engine()->CreateVoEChannel(); |
| 1801 | if (id == -1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1802 | RTC_LOG(LS_WARNING) << "CreateVoEChannel() failed."; |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1803 | return -1; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1804 | } |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 1805 | |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1806 | return id; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1807 | } |
| 1808 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1809 | bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) { |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1810 | if (engine()->voe()->base()->DeleteChannel(channel) == -1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1811 | RTC_LOG(LS_WARNING) << "DeleteChannel(" << channel << ") failed."; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1812 | return false; |
| 1813 | } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1814 | return true; |
| 1815 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1816 | |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1817 | bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1818 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1819 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1820 | RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1821 | |
| 1822 | uint32_t ssrc = sp.first_ssrc(); |
| 1823 | RTC_DCHECK(0 != ssrc); |
| 1824 | |
| 1825 | if (GetSendChannelId(ssrc) != -1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1826 | RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1827 | return false; |
| 1828 | } |
| 1829 | |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1830 | // Create a new channel for sending audio data. |
| 1831 | int channel = CreateVoEChannel(); |
| 1832 | if (channel == -1) { |
| 1833 | return false; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1834 | } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1835 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1836 | // Save the channel to send_streams_, so that RemoveSendStream() can still |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1837 | // delete the channel in case failure happens below. |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 1838 | webrtc::AudioTransport* audio_transport = |
| 1839 | engine()->voe()->base()->audio_transport(); |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 1840 | |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 1841 | rtc::Optional<std::string> audio_network_adaptor_config = |
| 1842 | GetAudioNetworkAdaptorConfig(options_); |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1843 | WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( |
Alex Narest | b3944f0 | 2017-10-13 12:56:18 | [diff] [blame] | 1844 | channel, audio_transport, ssrc, sp.cname, sp.id, send_codec_spec_, |
minyue | 6b825df | 2016-10-31 11:08:32 | [diff] [blame] | 1845 | send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config, |
ossu | 20a4b3f | 2017-04-27 09:08:52 | [diff] [blame] | 1846 | call_, this, engine()->encoder_factory_); |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 1847 | send_streams_.insert(std::make_pair(ssrc, stream)); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1848 | |
solenberg | 4a0f7b5 | 2016-06-16 20:07:33 | [diff] [blame] | 1849 | // At this point the stream's local SSRC has been updated. If it is the first |
| 1850 | // send stream, make sure that all the receive streams are updated with the |
| 1851 | // same SSRC in order to send receiver reports. |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1852 | if (send_streams_.size() == 1) { |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1853 | receiver_reports_ssrc_ = ssrc; |
solenberg | 4a0f7b5 | 2016-06-16 20:07:33 | [diff] [blame] | 1854 | for (const auto& kv : recv_streams_) { |
| 1855 | // TODO(solenberg): Allow applications to set the RTCP SSRC of receive |
| 1856 | // streams instead, so we can avoid recreating the streams here. |
| 1857 | kv.second->RecreateAudioReceiveStream(ssrc); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1858 | } |
| 1859 | } |
| 1860 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1861 | send_streams_[ssrc]->SetSend(send_); |
| 1862 | return true; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1863 | } |
| 1864 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 1865 | bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1866 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1867 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1868 | RTC_LOG(LS_INFO) << "RemoveSendStream: " << ssrc; |
solenberg | 3a94154 | 2015-11-16 15:34:50 | [diff] [blame] | 1869 | |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1870 | auto it = send_streams_.find(ssrc); |
| 1871 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1872 | RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc |
| 1873 | << " which doesn't exist."; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1874 | return false; |
| 1875 | } |
| 1876 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1877 | it->second->SetSend(false); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1878 | |
solenberg | 7602aab | 2016-11-14 19:30:07 | [diff] [blame] | 1879 | // TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find |
| 1880 | // the first active send stream and use that instead, reassociating receive |
| 1881 | // streams. |
| 1882 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1883 | // Clean up and delete the send stream+channel. |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1884 | int channel = it->second->channel(); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1885 | RTC_LOG(LS_INFO) << "Removing audio send stream " << ssrc |
| 1886 | << " with VoiceEngine channel #" << channel << "."; |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1887 | delete it->second; |
| 1888 | send_streams_.erase(it); |
| 1889 | if (!DeleteVoEChannel(channel)) { |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1890 | return false; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1891 | } |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1892 | if (send_streams_.empty()) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1893 | SetSend(false); |
solenberg | 0a617e2 | 2015-10-20 22:49:38 | [diff] [blame] | 1894 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1895 | return true; |
| 1896 | } |
| 1897 | |
| 1898 | bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1899 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1900 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1901 | RTC_LOG(LS_INFO) << "AddRecvStream: " << sp.ToString(); |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 1902 | |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1903 | if (!ValidateStreamParams(sp)) { |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 | [diff] [blame] | 1904 | return false; |
| 1905 | } |
| 1906 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1907 | const uint32_t ssrc = sp.first_ssrc(); |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1908 | if (ssrc == 0) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1909 | RTC_LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported."; |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1910 | return false; |
| 1911 | } |
| 1912 | |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 1913 | // If this stream was previously received unsignaled, we promote it, possibly |
| 1914 | // recreating the AudioReceiveStream, if sync_label has changed. |
| 1915 | if (MaybeDeregisterUnsignaledRecvStream(ssrc)) { |
solenberg | 4904fb6 | 2017-02-17 20:01:14 | [diff] [blame] | 1916 | recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.sync_label); |
solenberg | 4904fb6 | 2017-02-17 20:01:14 | [diff] [blame] | 1917 | return true; |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 1918 | } |
solenberg | 0b67546 | 2015-10-09 08:37:09 | [diff] [blame] | 1919 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1920 | if (GetReceiveChannelId(ssrc) != -1) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1921 | RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1922 | return false; |
| 1923 | } |
Fredrik Solenberg | 4b60c73 | 2015-05-07 12:07:48 | [diff] [blame] | 1924 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1925 | // Create a new channel for receiving audio data. |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1926 | const int channel = CreateVoEChannel(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1927 | if (channel == -1) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1928 | return false; |
| 1929 | } |
Minyue | 2013aec | 2015-05-13 12:14:42 | [diff] [blame] | 1930 | |
stefan | ba4c0e4 | 2016-02-04 12:12:24 | [diff] [blame] | 1931 | recv_streams_.insert(std::make_pair( |
kwiberg | 1c07c70 | 2017-03-27 14:15:49 | [diff] [blame] | 1932 | ssrc, |
| 1933 | new WebRtcAudioReceiveStream( |
| 1934 | channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_, |
| 1935 | recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this, |
| 1936 | engine()->decoder_factory_, decoder_map_))); |
aleloi | 84ef615 | 2016-08-04 12:28:21 | [diff] [blame] | 1937 | recv_streams_[ssrc]->SetPlayout(playout_); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1938 | |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 1939 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1940 | } |
| 1941 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 1942 | bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 1943 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1944 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1945 | RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 1946 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1947 | const auto it = recv_streams_.find(ssrc); |
| 1948 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1949 | RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc |
| 1950 | << " which doesn't exist."; |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1951 | return false; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1952 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1953 | |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 1954 | MaybeDeregisterUnsignaledRecvStream(ssrc); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1955 | |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1956 | const int channel = it->second->channel(); |
| 1957 | |
| 1958 | // Clean up and delete the receive stream+channel. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1959 | RTC_LOG(LS_INFO) << "Removing audio receive stream " << ssrc |
| 1960 | << " with VoiceEngine channel #" << channel << "."; |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 1961 | it->second->SetRawAudioSink(nullptr); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1962 | delete it->second; |
| 1963 | recv_streams_.erase(it); |
| 1964 | return DeleteVoEChannel(channel); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1965 | } |
| 1966 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1967 | bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc, |
| 1968 | AudioSource* source) { |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 1969 | auto it = send_streams_.find(ssrc); |
| 1970 | if (it == send_streams_.end()) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1971 | if (source) { |
| 1972 | // Return an error if trying to set a valid source with an invalid ssrc. |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 1973 | RTC_LOG(LS_ERROR) << "SetLocalSource failed with ssrc " << ssrc; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 1974 | return false; |
| 1975 | } |
| 1976 | |
| 1977 | // The channel likely has gone away, do nothing. |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1978 | return true; |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1979 | } |
| 1980 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1981 | if (source) { |
| 1982 | it->second->SetSource(source); |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 1983 | } else { |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 1984 | it->second->ClearSource(); |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 1985 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 1986 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1987 | return true; |
| 1988 | } |
| 1989 | |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 1990 | // TODO(solenberg): Remove, once AudioMonitor is gone. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1991 | bool WebRtcVoiceMediaChannel::GetActiveStreams( |
| 1992 | AudioInfo::StreamList* actives) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 1993 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1994 | actives->clear(); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 1995 | for (const auto& ch : recv_streams_) { |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 1996 | int level = ch.second->GetOutputLevel(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1997 | if (level > 0) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 08:45:53 | [diff] [blame] | 1998 | actives->push_back(std::make_pair(ch.first, level)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1999 | } |
| 2000 | } |
| 2001 | return true; |
| 2002 | } |
| 2003 | |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 2004 | // TODO(solenberg): Remove, once AudioMonitor is gone. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2005 | int WebRtcVoiceMediaChannel::GetOutputLevel() { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2006 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2007 | int highest = 0; |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 2008 | for (const auto& ch : recv_streams_) { |
solenberg | 796b8f9 | 2017-03-02 01:02:23 | [diff] [blame] | 2009 | highest = std::max(ch.second->GetOutputLevel(), highest); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2010 | } |
| 2011 | return highest; |
| 2012 | } |
| 2013 | |
solenberg | 4bac9c5 | 2015-10-09 09:32:53 | [diff] [blame] | 2014 | bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2015 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2016 | std::vector<uint32_t> ssrcs(1, ssrc); |
deadbeef | 3bc1510 | 2017-04-21 02:25:07 | [diff] [blame] | 2017 | // SSRC of 0 represents the default receive stream. |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2018 | if (ssrc == 0) { |
| 2019 | default_recv_volume_ = volume; |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2020 | ssrcs = unsignaled_recv_ssrcs_; |
| 2021 | } |
| 2022 | for (uint32_t ssrc : ssrcs) { |
| 2023 | const auto it = recv_streams_.find(ssrc); |
| 2024 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2025 | RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc; |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2026 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2027 | } |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2028 | it->second->SetOutputVolume(volume); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2029 | RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume |
| 2030 | << " for recv stream with ssrc " << ssrc; |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2031 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2032 | return true; |
| 2033 | } |
| 2034 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2035 | bool WebRtcVoiceMediaChannel::CanInsertDtmf() { |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 2036 | return dtmf_payload_type_ ? true : false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2037 | } |
| 2038 | |
solenberg | 1d63dd0 | 2015-12-02 20:35:09 | [diff] [blame] | 2039 | bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event, |
| 2040 | int duration) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2041 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2042 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf"; |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 2043 | if (!dtmf_payload_type_) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2044 | return false; |
| 2045 | } |
| 2046 | |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 2047 | // Figure out which WebRtcAudioSendStream to send the event on. |
| 2048 | auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin(); |
| 2049 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2050 | RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; |
solenberg | 1d63dd0 | 2015-12-02 20:35:09 | [diff] [blame] | 2051 | return false; |
| 2052 | } |
Fredrik Solenberg | b572768 | 2015-12-04 14:22:19 | [diff] [blame] | 2053 | if (event < kMinTelephoneEventCode || |
| 2054 | event > kMaxTelephoneEventCode) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2055 | RTC_LOG(LS_WARNING) << "DTMF event code " << event << " out of range."; |
solenberg | 1d63dd0 | 2015-12-02 20:35:09 | [diff] [blame] | 2056 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2057 | } |
solenberg | ffbbcac | 2016-11-17 13:25:37 | [diff] [blame] | 2058 | RTC_DCHECK_NE(-1, dtmf_payload_freq_); |
| 2059 | return it->second->SendTelephoneEvent(*dtmf_payload_type_, dtmf_payload_freq_, |
| 2060 | event, duration); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2061 | } |
| 2062 | |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 | [diff] [blame] | 2063 | void WebRtcVoiceMediaChannel::OnPacketReceived( |
jbauch | eec21bd | 2016-03-20 13:15:43 | [diff] [blame] | 2064 | rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2065 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 4b60c73 | 2015-05-07 12:07:48 | [diff] [blame] | 2066 | |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2067 | const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| 2068 | packet_time.not_before); |
| 2069 | webrtc::PacketReceiver::DeliveryStatus delivery_result = |
| 2070 | call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
| 2071 | packet->cdata(), packet->size(), |
| 2072 | webrtc_packet_time); |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2073 | if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) { |
| 2074 | return; |
| 2075 | } |
| 2076 | |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2077 | // Create an unsignaled receive stream for this previously not received ssrc. |
| 2078 | // If there already is N unsignaled receive streams, delete the oldest. |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2079 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208 |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2080 | uint32_t ssrc = 0; |
jbauch | eec21bd | 2016-03-20 13:15:43 | [diff] [blame] | 2081 | if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2082 | return; |
| 2083 | } |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2084 | RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(), |
| 2085 | unsignaled_recv_ssrcs_.end(), ssrc) == unsignaled_recv_ssrcs_.end()); |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2086 | |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2087 | // Add new stream. |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2088 | StreamParams sp; |
| 2089 | sp.ssrcs.push_back(ssrc); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2090 | RTC_LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc; |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2091 | if (!AddRecvStream(sp)) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2092 | RTC_LOG(LS_WARNING) << "Could not create unsignaled receive stream."; |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2093 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2094 | } |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2095 | unsignaled_recv_ssrcs_.push_back(ssrc); |
| 2096 | RTC_HISTOGRAM_COUNTS_LINEAR( |
| 2097 | "WebRTC.Audio.NumOfUnsignaledStreams", unsignaled_recv_ssrcs_.size(), 1, |
| 2098 | 100, 101); |
solenberg | f748ca4 | 2017-02-06 21:03:19 | [diff] [blame] | 2099 | |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2100 | // Remove oldest unsignaled stream, if we have too many. |
| 2101 | if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) { |
| 2102 | uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front(); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2103 | RTC_LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC=" |
| 2104 | << remove_ssrc; |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2105 | RemoveRecvStream(remove_ssrc); |
| 2106 | } |
| 2107 | RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size()); |
| 2108 | |
| 2109 | SetOutputVolume(ssrc, default_recv_volume_); |
| 2110 | |
| 2111 | // The default sink can only be attached to one stream at a time, so we hook |
| 2112 | // it up to the *latest* unsignaled stream we've seen, in order to support the |
| 2113 | // case where the SSRC of one unsignaled stream changes. |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2114 | if (default_sink_) { |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2115 | for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) { |
| 2116 | auto it = recv_streams_.find(drop_ssrc); |
| 2117 | it->second->SetRawAudioSink(nullptr); |
| 2118 | } |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2119 | std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( |
| 2120 | new ProxySink(default_sink_.get())); |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2121 | SetRawAudioSink(ssrc, std::move(proxy_sink)); |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2122 | } |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2123 | |
mflodman | 3d7db26 | 2016-04-29 07:57:13 | [diff] [blame] | 2124 | delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
| 2125 | packet->cdata(), |
| 2126 | packet->size(), |
| 2127 | webrtc_packet_time); |
| 2128 | RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2129 | } |
| 2130 | |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 | [diff] [blame] | 2131 | void WebRtcVoiceMediaChannel::OnRtcpReceived( |
jbauch | eec21bd | 2016-03-20 13:15:43 | [diff] [blame] | 2132 | rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2133 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 4b60c73 | 2015-05-07 12:07:48 | [diff] [blame] | 2134 | |
Fredrik Solenberg | 709ed67 | 2015-09-15 10:26:33 | [diff] [blame] | 2135 | // Forward packet to Call as well. |
| 2136 | const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| 2137 | packet_time.not_before); |
| 2138 | call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
jbauch | eec21bd | 2016-03-20 13:15:43 | [diff] [blame] | 2139 | packet->cdata(), packet->size(), webrtc_packet_time); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2140 | } |
| 2141 | |
Honghai Zhang | cc411c0 | 2016-03-30 00:27:21 | [diff] [blame] | 2142 | void WebRtcVoiceMediaChannel::OnNetworkRouteChanged( |
| 2143 | const std::string& transport_name, |
Honghai Zhang | 0e533ef | 2016-04-19 22:41:36 | [diff] [blame] | 2144 | const rtc::NetworkRoute& network_route) { |
Zhi Huang | 5f5918f | 2017-11-13 01:26:23 | [diff] [blame] | 2145 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2146 | // TODO(zhihaung): Merge these two callbacks. |
Honghai Zhang | 0e533ef | 2016-04-19 22:41:36 | [diff] [blame] | 2147 | call_->OnNetworkRouteChanged(transport_name, network_route); |
Zhi Huang | 5f5918f | 2017-11-13 01:26:23 | [diff] [blame] | 2148 | call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO, |
| 2149 | network_route.packet_overhead); |
Honghai Zhang | cc411c0 | 2016-03-30 00:27:21 | [diff] [blame] | 2150 | } |
| 2151 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 2152 | bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2153 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 9421853 | 2016-06-16 17:53:22 | [diff] [blame] | 2154 | const auto it = send_streams_.find(ssrc); |
| 2155 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2156 | RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2157 | return false; |
| 2158 | } |
solenberg | 9421853 | 2016-06-16 17:53:22 | [diff] [blame] | 2159 | it->second->SetMuted(muted); |
| 2160 | |
| 2161 | // TODO(solenberg): |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 | [diff] [blame] | 2162 | // We set the AGC to mute state only when all the channels are muted. |
| 2163 | // This implementation is not ideal, instead we should signal the AGC when |
| 2164 | // the mic channel is muted/unmuted. We can't do it today because there |
| 2165 | // is no good way to know which stream is mapping to the mic channel. |
| 2166 | bool all_muted = muted; |
solenberg | 9421853 | 2016-06-16 17:53:22 | [diff] [blame] | 2167 | for (const auto& kv : send_streams_) { |
| 2168 | all_muted = all_muted && kv.second->muted(); |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 | [diff] [blame] | 2169 | } |
solenberg | 059fb44 | 2016-10-26 12:12:24 | [diff] [blame] | 2170 | engine()->apm()->set_output_will_be_muted(all_muted); |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 | [diff] [blame] | 2171 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2172 | return true; |
| 2173 | } |
| 2174 | |
deadbeef | 8034614 | 2016-04-27 21:17:10 | [diff] [blame] | 2175 | bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2176 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate."; |
deadbeef | 8034614 | 2016-04-27 21:17:10 | [diff] [blame] | 2177 | max_send_bitrate_bps_ = bps; |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 2178 | bool success = true; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 2179 | for (const auto& kv : send_streams_) { |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 2180 | if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) { |
| 2181 | success = false; |
skvlad | e0d4637 | 2016-04-08 05:59:22 | [diff] [blame] | 2182 | } |
| 2183 | } |
minyue | 7a97344 | 2016-10-20 10:27:12 | [diff] [blame] | 2184 | return success; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2185 | } |
| 2186 | |
skvlad | 7a43d25 | 2016-03-22 22:32:27 | [diff] [blame] | 2187 | void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) { |
| 2188 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2189 | RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); |
skvlad | 7a43d25 | 2016-03-22 22:32:27 | [diff] [blame] | 2190 | call_->SignalChannelNetworkState( |
| 2191 | webrtc::MediaType::AUDIO, |
| 2192 | ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); |
| 2193 | } |
| 2194 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2195 | bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { |
Peter Boström | ca8b404 | 2016-03-08 22:24:13 | [diff] [blame] | 2196 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats"); |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2197 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2198 | RTC_DCHECK(info); |
solenberg | d97ec30 | 2015-10-07 08:40:33 | [diff] [blame] | 2199 | |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2200 | // Get SSRC and stats for each sender. |
hbos | 1acfbd2 | 2016-11-18 07:43:29 | [diff] [blame] | 2201 | RTC_DCHECK_EQ(info->senders.size(), 0U); |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2202 | for (const auto& stream : send_streams_) { |
Ivo Creusen | 56d46090 | 2017-11-24 16:29:59 | [diff] [blame] | 2203 | webrtc::AudioSendStream::Stats stats = |
| 2204 | stream.second->GetStats(recv_streams_.size() > 0); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 2205 | VoiceSenderInfo sinfo; |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2206 | sinfo.add_ssrc(stats.local_ssrc); |
| 2207 | sinfo.bytes_sent = stats.bytes_sent; |
| 2208 | sinfo.packets_sent = stats.packets_sent; |
| 2209 | sinfo.packets_lost = stats.packets_lost; |
| 2210 | sinfo.fraction_lost = stats.fraction_lost; |
| 2211 | sinfo.codec_name = stats.codec_name; |
hbos | 1acfbd2 | 2016-11-18 07:43:29 | [diff] [blame] | 2212 | sinfo.codec_payload_type = stats.codec_payload_type; |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2213 | sinfo.ext_seqnum = stats.ext_seqnum; |
| 2214 | sinfo.jitter_ms = stats.jitter_ms; |
| 2215 | sinfo.rtt_ms = stats.rtt_ms; |
| 2216 | sinfo.audio_level = stats.audio_level; |
zstein | e76bd3a | 2017-07-14 19:17:49 | [diff] [blame] | 2217 | sinfo.total_input_energy = stats.total_input_energy; |
| 2218 | sinfo.total_input_duration = stats.total_input_duration; |
Taylor Brandstetter | 1a018dc | 2016-03-08 20:37:39 | [diff] [blame] | 2219 | sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false); |
ivoc | e1198e0 | 2017-09-08 15:13:19 | [diff] [blame] | 2220 | sinfo.ana_statistics = stats.ana_statistics; |
Ivo Creusen | 56d46090 | 2017-11-24 16:29:59 | [diff] [blame] | 2221 | sinfo.apm_statistics = stats.apm_statistics; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 2222 | info->senders.push_back(sinfo); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2223 | } |
| 2224 | |
solenberg | 85a0496 | 2015-10-27 10:35:21 | [diff] [blame] | 2225 | // Get SSRC and stats for each receiver. |
hbos | 1acfbd2 | 2016-11-18 07:43:29 | [diff] [blame] | 2226 | RTC_DCHECK_EQ(info->receivers.size(), 0U); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 2227 | for (const auto& stream : recv_streams_) { |
deadbeef | 4e2deab | 2017-09-20 20:56:21 | [diff] [blame] | 2228 | uint32_t ssrc = stream.first; |
| 2229 | // When SSRCs are unsignaled, there's only one audio MediaStreamTrack, but |
| 2230 | // multiple RTP streams can be received over time (if the SSRC changes for |
| 2231 | // whatever reason). We only want the RTCMediaStreamTrackStats to represent |
| 2232 | // the stats for the most recent stream (the one whose audio is actually |
| 2233 | // routed to the MediaStreamTrack), so here we ignore any unsignaled SSRCs |
| 2234 | // except for the most recent one (last in the vector). This is somewhat of |
| 2235 | // a hack, and means you don't get *any* stats for these inactive streams, |
| 2236 | // but it's slightly better than the previous behavior, which was "highest |
| 2237 | // SSRC wins". |
| 2238 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8158 |
| 2239 | if (!unsignaled_recv_ssrcs_.empty()) { |
| 2240 | auto end_it = --unsignaled_recv_ssrcs_.end(); |
| 2241 | if (std::find(unsignaled_recv_ssrcs_.begin(), end_it, ssrc) != end_it) { |
| 2242 | continue; |
| 2243 | } |
| 2244 | } |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 08:49:27 | [diff] [blame] | 2245 | webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats(); |
| 2246 | VoiceReceiverInfo rinfo; |
| 2247 | rinfo.add_ssrc(stats.remote_ssrc); |
| 2248 | rinfo.bytes_rcvd = stats.bytes_rcvd; |
| 2249 | rinfo.packets_rcvd = stats.packets_rcvd; |
| 2250 | rinfo.packets_lost = stats.packets_lost; |
| 2251 | rinfo.fraction_lost = stats.fraction_lost; |
| 2252 | rinfo.codec_name = stats.codec_name; |
hbos | 1acfbd2 | 2016-11-18 07:43:29 | [diff] [blame] | 2253 | rinfo.codec_payload_type = stats.codec_payload_type; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 08:49:27 | [diff] [blame] | 2254 | rinfo.ext_seqnum = stats.ext_seqnum; |
| 2255 | rinfo.jitter_ms = stats.jitter_ms; |
| 2256 | rinfo.jitter_buffer_ms = stats.jitter_buffer_ms; |
| 2257 | rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms; |
| 2258 | rinfo.delay_estimate_ms = stats.delay_estimate_ms; |
| 2259 | rinfo.audio_level = stats.audio_level; |
zstein | e76bd3a | 2017-07-14 19:17:49 | [diff] [blame] | 2260 | rinfo.total_output_energy = stats.total_output_energy; |
Steve Anton | 2dbc69f | 2017-08-25 00:15:13 | [diff] [blame] | 2261 | rinfo.total_samples_received = stats.total_samples_received; |
zstein | e76bd3a | 2017-07-14 19:17:49 | [diff] [blame] | 2262 | rinfo.total_output_duration = stats.total_output_duration; |
Steve Anton | 2dbc69f | 2017-08-25 00:15:13 | [diff] [blame] | 2263 | rinfo.concealed_samples = stats.concealed_samples; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 07:28:20 | [diff] [blame] | 2264 | rinfo.concealment_events = stats.concealment_events; |
Gustaf Ullberg | b0a0207 | 2017-10-02 10:00:34 | [diff] [blame] | 2265 | rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 08:49:27 | [diff] [blame] | 2266 | rinfo.expand_rate = stats.expand_rate; |
| 2267 | rinfo.speech_expand_rate = stats.speech_expand_rate; |
| 2268 | rinfo.secondary_decoded_rate = stats.secondary_decoded_rate; |
minyue-webrtc | 0e320ec | 2017-08-28 11:51:27 | [diff] [blame] | 2269 | rinfo.secondary_discarded_rate = stats.secondary_discarded_rate; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 08:49:27 | [diff] [blame] | 2270 | rinfo.accelerate_rate = stats.accelerate_rate; |
| 2271 | rinfo.preemptive_expand_rate = stats.preemptive_expand_rate; |
| 2272 | rinfo.decoding_calls_to_silence_generator = |
| 2273 | stats.decoding_calls_to_silence_generator; |
| 2274 | rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq; |
| 2275 | rinfo.decoding_normal = stats.decoding_normal; |
| 2276 | rinfo.decoding_plc = stats.decoding_plc; |
| 2277 | rinfo.decoding_cng = stats.decoding_cng; |
| 2278 | rinfo.decoding_plc_cng = stats.decoding_plc_cng; |
henrik.lundin | 6348978 | 2016-09-20 08:47:12 | [diff] [blame] | 2279 | rinfo.decoding_muted_output = stats.decoding_muted_output; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 08:49:27 | [diff] [blame] | 2280 | rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms; |
| 2281 | info->receivers.push_back(rinfo); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2282 | } |
| 2283 | |
hbos | 1acfbd2 | 2016-11-18 07:43:29 | [diff] [blame] | 2284 | // Get codec info |
| 2285 | for (const AudioCodec& codec : send_codecs_) { |
| 2286 | webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| 2287 | info->send_codecs.insert( |
| 2288 | std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| 2289 | } |
| 2290 | for (const AudioCodec& codec : recv_codecs_) { |
| 2291 | webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| 2292 | info->receive_codecs.insert( |
| 2293 | std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| 2294 | } |
| 2295 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2296 | return true; |
| 2297 | } |
| 2298 | |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 2299 | void WebRtcVoiceMediaChannel::SetRawAudioSink( |
| 2300 | uint32_t ssrc, |
kwiberg | 686a8ef | 2016-02-26 11:00:35 | [diff] [blame] | 2301 | std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 2302 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2303 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" |
| 2304 | << ssrc << " " << (sink ? "(ptr)" : "NULL"); |
deadbeef | 884f585 | 2016-01-15 17:20:04 | [diff] [blame] | 2305 | if (ssrc == 0) { |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2306 | if (!unsignaled_recv_ssrcs_.empty()) { |
kwiberg | 686a8ef | 2016-02-26 11:00:35 | [diff] [blame] | 2307 | std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( |
deadbeef | 884f585 | 2016-01-15 17:20:04 | [diff] [blame] | 2308 | sink ? new ProxySink(sink.get()) : nullptr); |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2309 | SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink)); |
deadbeef | 884f585 | 2016-01-15 17:20:04 | [diff] [blame] | 2310 | } |
| 2311 | default_sink_ = std::move(sink); |
| 2312 | return; |
| 2313 | } |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 2314 | const auto it = recv_streams_.find(ssrc); |
| 2315 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 10:09:25 | [diff] [blame] | 2316 | RTC_LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc; |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 2317 | return; |
| 2318 | } |
deadbeef | 2d110be | 2016-01-13 20:00:26 | [diff] [blame] | 2319 | it->second->SetRawAudioSink(std::move(sink)); |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 2320 | } |
| 2321 | |
hbos | 8d609f6 | 2017-04-10 14:39:05 | [diff] [blame] | 2322 | std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources( |
| 2323 | uint32_t ssrc) const { |
| 2324 | auto it = recv_streams_.find(ssrc); |
| 2325 | RTC_DCHECK(it != recv_streams_.end()) |
| 2326 | << "Attempting to get contributing sources for SSRC:" << ssrc |
| 2327 | << " which doesn't exist."; |
| 2328 | return it->second->GetSources(); |
| 2329 | } |
| 2330 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 2331 | int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2332 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 7add058 | 2015-11-20 17:59:34 | [diff] [blame] | 2333 | const auto it = recv_streams_.find(ssrc); |
| 2334 | if (it != recv_streams_.end()) { |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 2335 | return it->second->channel(); |
solenberg | 8fb30c3 | 2015-10-13 10:06:58 | [diff] [blame] | 2336 | } |
solenberg | 1ac5614 | 2015-10-13 10:58:19 | [diff] [blame] | 2337 | return -1; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2338 | } |
| 2339 | |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 2340 | int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const { |
solenberg | 566ef24 | 2015-11-06 23:34:49 | [diff] [blame] | 2341 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | c96df77 | 2015-10-21 20:01:53 | [diff] [blame] | 2342 | const auto it = send_streams_.find(ssrc); |
| 2343 | if (it != send_streams_.end()) { |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 | [diff] [blame] | 2344 | return it->second->channel(); |
solenberg | 8fb30c3 | 2015-10-13 10:06:58 | [diff] [blame] | 2345 | } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 | [diff] [blame] | 2346 | return -1; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2347 | } |
solenberg | 2100c0b | 2017-03-01 19:29:29 | [diff] [blame] | 2348 | |
| 2349 | bool WebRtcVoiceMediaChannel:: |
| 2350 | MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc) { |
| 2351 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2352 | auto it = std::find(unsignaled_recv_ssrcs_.begin(), |
| 2353 | unsignaled_recv_ssrcs_.end(), |
| 2354 | ssrc); |
| 2355 | if (it != unsignaled_recv_ssrcs_.end()) { |
| 2356 | unsignaled_recv_ssrcs_.erase(it); |
| 2357 | return true; |
| 2358 | } |
| 2359 | return false; |
| 2360 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 2361 | } // namespace cricket |
| 2362 | |
| 2363 | #endif // HAVE_WEBRTC_VOICE |