henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
kjellander | 1afca73 | 2016-02-07 20:46:45 -0800 | [diff] [blame] | 2 | * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 3 | * |
kjellander | 1afca73 | 2016-02-07 20:46:45 -0800 | [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 +0000 | [diff] [blame] | 9 | */ |
| 10 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 11 | #ifdef HAVE_WEBRTC_VOICE |
| 12 | |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 13 | #include "media/engine/webrtc_voice_engine.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 14 | |
| 15 | #include <algorithm> |
| 16 | #include <cstdio> |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 17 | #include <functional> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 18 | #include <string> |
Steve Anton | e78bcb9 | 2017-10-31 09:53:08 -0700 | [diff] [blame] | 19 | #include <utility> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 20 | #include <vector> |
| 21 | |
Niels Möller | 3c7d599 | 2018-10-19 15:29:54 +0200 | [diff] [blame] | 22 | #include "absl/strings/match.h" |
Karl Wiberg | 0812634 | 2018-03-20 19:18:55 +0100 | [diff] [blame] | 23 | #include "api/audio_codecs/audio_codec_pair_id.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 24 | #include "api/call/audio_sink.h" |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 25 | #include "api/media_transport_interface.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 26 | #include "media/base/audio_source.h" |
| 27 | #include "media/base/media_constants.h" |
| 28 | #include "media/base/stream_params.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 29 | #include "media/engine/adm_helpers.h" |
| 30 | #include "media/engine/apm_helpers.h" |
| 31 | #include "media/engine/payload_type_mapper.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 32 | #include "media/engine/webrtc_media_engine.h" |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 33 | #include "modules/audio_device/audio_device_impl.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 34 | #include "modules/audio_mixer/audio_mixer_impl.h" |
| 35 | #include "modules/audio_processing/aec_dump/aec_dump_factory.h" |
| 36 | #include "modules/audio_processing/include/audio_processing.h" |
| 37 | #include "rtc_base/arraysize.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 38 | #include "rtc_base/byte_order.h" |
| 39 | #include "rtc_base/constructor_magic.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 40 | #include "rtc_base/helpers.h" |
| 41 | #include "rtc_base/logging.h" |
| 42 | #include "rtc_base/race_checker.h" |
Jonas Olsson | abbe841 | 2018-04-03 13:40:05 +0200 | [diff] [blame] | 43 | #include "rtc_base/strings/audio_format_to_string.h" |
Jonas Olsson | 366a50c | 2018-09-06 13:41:30 +0200 | [diff] [blame] | 44 | #include "rtc_base/strings/string_builder.h" |
Artem Titov | a76af0c | 2018-07-23 17:38:12 +0200 | [diff] [blame] | 45 | #include "rtc_base/third_party/base64/base64.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 46 | #include "rtc_base/trace_event.h" |
| 47 | #include "system_wrappers/include/field_trial.h" |
| 48 | #include "system_wrappers/include/metrics.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 49 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 50 | namespace cricket { |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 51 | namespace { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 52 | |
solenberg | 418b7d3 | 2017-06-13 00:38:27 -0700 | [diff] [blame] | 53 | constexpr size_t kMaxUnsignaledRecvStreams = 4; |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 54 | |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 55 | constexpr int kNackRtpHistoryMs = 5000; |
minyue@webrtc.org | 2dc6f31 | 2014-10-31 05:33:10 +0000 | [diff] [blame] | 56 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 57 | // For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000. |
minyue | 10cbb46 | 2016-11-07 09:29:22 -0800 | [diff] [blame] | 58 | const int kOpusMinBitrateBps = 6000; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 59 | const int kOpusBitrateFbBps = 32000; |
deadbeef | 8034614 | 2016-04-27 14:17:10 -0700 | [diff] [blame] | 60 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 61 | const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1) |
Fredrik Solenberg | b572768 | 2015-12-04 15:22:19 +0100 | [diff] [blame] | 62 | const int kMaxTelephoneEventCode = 255; |
Fredrik Solenberg | b572768 | 2015-12-04 15:22:19 +0100 | [diff] [blame] | 63 | |
solenberg | 31642aa | 2016-03-14 08:00:37 -0700 | [diff] [blame] | 64 | const int kMinPayloadType = 0; |
| 65 | const int kMaxPayloadType = 127; |
| 66 | |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 67 | class ProxySink : public webrtc::AudioSinkInterface { |
| 68 | public: |
Steve Anton | e78bcb9 | 2017-10-31 09:53:08 -0700 | [diff] [blame] | 69 | explicit ProxySink(AudioSinkInterface* sink) : sink_(sink) { |
| 70 | RTC_DCHECK(sink); |
| 71 | } |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 72 | |
| 73 | void OnData(const Data& audio) override { sink_->OnData(audio); } |
| 74 | |
| 75 | private: |
| 76 | webrtc::AudioSinkInterface* sink_; |
| 77 | }; |
| 78 | |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 79 | bool ValidateStreamParams(const StreamParams& sp) { |
| 80 | if (sp.ssrcs.empty()) { |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 81 | RTC_DLOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 82 | return false; |
| 83 | } |
| 84 | if (sp.ssrcs.size() > 1) { |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 85 | RTC_DLOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " |
| 86 | << sp.ToString(); |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 87 | return false; |
| 88 | } |
| 89 | return true; |
| 90 | } |
| 91 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 92 | // Dumps an AudioCodec in RFC 2327-ish format. |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 93 | std::string ToString(const AudioCodec& codec) { |
Jonas Olsson | 366a50c | 2018-09-06 13:41:30 +0200 | [diff] [blame] | 94 | rtc::StringBuilder ss; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 95 | ss << codec.name << "/" << codec.clockrate << "/" << codec.channels; |
| 96 | if (!codec.params.empty()) { |
| 97 | ss << " {"; |
| 98 | for (const auto& param : codec.params) { |
| 99 | ss << " " << param.first << "=" << param.second; |
| 100 | } |
| 101 | ss << " }"; |
| 102 | } |
| 103 | ss << " (" << codec.id << ")"; |
Jonas Olsson | 84df1c7 | 2018-09-14 16:59:32 +0200 | [diff] [blame] | 104 | return ss.Release(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 105 | } |
Minyue Li | 7100dcd | 2015-03-27 05:05:59 +0100 | [diff] [blame] | 106 | |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 107 | bool IsCodec(const AudioCodec& codec, const char* ref_name) { |
Niels Möller | 3c7d599 | 2018-10-19 15:29:54 +0200 | [diff] [blame] | 108 | return absl::EqualsIgnoreCase(codec.name, ref_name); |
Minyue Li | 7100dcd | 2015-03-27 05:05:59 +0100 | [diff] [blame] | 109 | } |
| 110 | |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 111 | bool FindCodec(const std::vector<AudioCodec>& codecs, |
solenberg | 26c8c91 | 2015-11-27 04:00:25 -0800 | [diff] [blame] | 112 | const AudioCodec& codec, |
| 113 | AudioCodec* found_codec) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 10:45:53 +0200 | [diff] [blame] | 114 | for (const AudioCodec& c : codecs) { |
| 115 | if (c.Matches(codec)) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 116 | if (found_codec != NULL) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 10:45:53 +0200 | [diff] [blame] | 117 | *found_codec = c; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 118 | } |
| 119 | return true; |
| 120 | } |
| 121 | } |
| 122 | return false; |
| 123 | } |
wu@webrtc.org | 1d1ffc9 | 2013-10-16 18:12:02 +0000 | [diff] [blame] | 124 | |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 125 | bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) { |
| 126 | if (codecs.empty()) { |
| 127 | return true; |
| 128 | } |
| 129 | std::vector<int> payload_types; |
| 130 | for (const AudioCodec& codec : codecs) { |
| 131 | payload_types.push_back(codec.id); |
| 132 | } |
| 133 | std::sort(payload_types.begin(), payload_types.end()); |
| 134 | auto it = std::unique(payload_types.begin(), payload_types.end()); |
| 135 | return it == payload_types.end(); |
| 136 | } |
| 137 | |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 138 | absl::optional<std::string> GetAudioNetworkAdaptorConfig( |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 139 | const AudioOptions& options) { |
| 140 | if (options.audio_network_adaptor && *options.audio_network_adaptor && |
| 141 | options.audio_network_adaptor_config) { |
| 142 | // Turn on audio network adaptor only when |options_.audio_network_adaptor| |
| 143 | // equals true and |options_.audio_network_adaptor_config| has a value. |
| 144 | return options.audio_network_adaptor_config; |
| 145 | } |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 146 | return absl::nullopt; |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 147 | } |
| 148 | |
deadbeef | e702b30 | 2017-02-04 12:09:01 -0800 | [diff] [blame] | 149 | // |max_send_bitrate_bps| is the bitrate from "b=" in SDP. |
| 150 | // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters. |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 151 | absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps, |
| 152 | absl::optional<int> rtp_max_bitrate_bps, |
| 153 | const webrtc::AudioCodecSpec& spec) { |
deadbeef | e702b30 | 2017-02-04 12:09:01 -0800 | [diff] [blame] | 154 | // If application-configured bitrate is set, take minimum of that and SDP |
| 155 | // bitrate. |
zstein | a5e0df6 | 2017-06-14 11:41:48 -0700 | [diff] [blame] | 156 | const int bps = |
| 157 | rtp_max_bitrate_bps |
| 158 | ? webrtc::MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps) |
| 159 | : max_send_bitrate_bps; |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 160 | if (bps <= 0) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 161 | return spec.info.default_bitrate_bps; |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 162 | } |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 163 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 164 | if (bps < spec.info.min_bitrate_bps) { |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 165 | // If codec is not multi-rate and |bps| is less than the fixed bitrate then |
| 166 | // fail. If codec is not multi-rate and |bps| exceeds or equal the fixed |
| 167 | // bitrate then ignore. |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 168 | RTC_LOG(LS_ERROR) << "Failed to set codec " << spec.format.name |
| 169 | << " to bitrate " << bps << " bps" |
| 170 | << ", requires at least " << spec.info.min_bitrate_bps |
| 171 | << " bps."; |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 172 | return absl::nullopt; |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 173 | } |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 174 | |
| 175 | if (spec.info.HasFixedBitrate()) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 176 | return spec.info.default_bitrate_bps; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 177 | } else { |
| 178 | // If codec is multi-rate then just set the bitrate. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 179 | return std::min(bps, spec.info.max_bitrate_bps); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 180 | } |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 181 | } |
| 182 | |
solenberg | 76377c5 | 2017-02-21 00:54:31 -0800 | [diff] [blame] | 183 | } // namespace |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 184 | |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 185 | WebRtcVoiceEngine::WebRtcVoiceEngine( |
| 186 | webrtc::AudioDeviceModule* adm, |
ossu | eb1fde4 | 2017-05-02 06:46:30 -0700 | [diff] [blame] | 187 | const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
gyzhou | 95aa964 | 2016-12-13 14:06:26 -0800 | [diff] [blame] | 188 | const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
peah | a9cc40b | 2017-06-29 08:32:09 -0700 | [diff] [blame] | 189 | rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 190 | rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 191 | : adm_(adm), |
ossu | eb1fde4 | 2017-05-02 06:46:30 -0700 | [diff] [blame] | 192 | encoder_factory_(encoder_factory), |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 193 | decoder_factory_(decoder_factory), |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 194 | audio_mixer_(audio_mixer), |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 195 | apm_(audio_processing) { |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 196 | // This may be called from any thread, so detach thread checkers. |
| 197 | worker_thread_checker_.DetachFromThread(); |
solenberg | 26c8c91 | 2015-11-27 04:00:25 -0800 | [diff] [blame] | 198 | signal_thread_checker_.DetachFromThread(); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 199 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 200 | RTC_DCHECK(decoder_factory); |
| 201 | RTC_DCHECK(encoder_factory); |
peah | a9cc40b | 2017-06-29 08:32:09 -0700 | [diff] [blame] | 202 | RTC_DCHECK(audio_processing); |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 203 | // The rest of our initialization will happen in Init. |
| 204 | } |
| 205 | |
| 206 | WebRtcVoiceEngine::~WebRtcVoiceEngine() { |
| 207 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 208 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 209 | if (initialized_) { |
| 210 | StopAecDump(); |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 211 | |
| 212 | // Stop AudioDevice. |
| 213 | adm()->StopPlayout(); |
| 214 | adm()->StopRecording(); |
| 215 | adm()->RegisterAudioCallback(nullptr); |
| 216 | adm()->Terminate(); |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 217 | } |
| 218 | } |
| 219 | |
| 220 | void WebRtcVoiceEngine::Init() { |
| 221 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 222 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::Init"; |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 223 | |
| 224 | // TaskQueue expects to be created/destroyed on the same thread. |
| 225 | low_priority_worker_queue_.reset( |
| 226 | new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW)); |
| 227 | |
ossu | eb1fde4 | 2017-05-02 06:46:30 -0700 | [diff] [blame] | 228 | // Load our audio codec lists. |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 229 | RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:"; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 230 | send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders()); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 231 | for (const AudioCodec& codec : send_codecs_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 232 | RTC_LOG(LS_INFO) << ToString(codec); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 233 | } |
| 234 | |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 235 | RTC_LOG(LS_INFO) << "Supported recv codecs in order of preference:"; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 236 | recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders()); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 237 | for (const AudioCodec& codec : recv_codecs_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 238 | RTC_LOG(LS_INFO) << ToString(codec); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 239 | } |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 240 | |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 241 | #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) |
| 242 | // No ADM supplied? Create a default one. |
solenberg | ff97631 | 2016-03-30 23:28:51 -0700 | [diff] [blame] | 243 | if (!adm_) { |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 244 | adm_ = webrtc::AudioDeviceModule::Create( |
| 245 | webrtc::AudioDeviceModule::kPlatformDefaultAudio); |
solenberg | ff97631 | 2016-03-30 23:28:51 -0700 | [diff] [blame] | 246 | } |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 247 | #endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE |
| 248 | RTC_CHECK(adm()); |
| 249 | webrtc::adm_helpers::Init(adm()); |
Fredrik Solenberg | 55900fd | 2017-11-23 20:22:55 +0100 | [diff] [blame] | 250 | webrtc::apm_helpers::Init(apm()); |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 251 | |
| 252 | // Set up AudioState. |
| 253 | { |
| 254 | webrtc::AudioState::Config config; |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 255 | if (audio_mixer_) { |
| 256 | config.audio_mixer = audio_mixer_; |
| 257 | } else { |
| 258 | config.audio_mixer = webrtc::AudioMixerImpl::Create(); |
| 259 | } |
| 260 | config.audio_processing = apm_; |
| 261 | config.audio_device_module = adm_; |
| 262 | audio_state_ = webrtc::AudioState::Create(config); |
| 263 | } |
| 264 | |
| 265 | // Connect the ADM to our audio path. |
| 266 | adm()->RegisterAudioCallback(audio_state()->audio_transport()); |
solenberg | 76377c5 | 2017-02-21 00:54:31 -0800 | [diff] [blame] | 267 | |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 268 | // Save the default AGC configuration settings. This must happen before |
solenberg | 246b817 | 2015-12-08 09:50:23 -0800 | [diff] [blame] | 269 | // calling ApplyOptions or the default will be overwritten. |
peah | a9cc40b | 2017-06-29 08:32:09 -0700 | [diff] [blame] | 270 | default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm()); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 271 | |
solenberg | 0f7d293 | 2016-01-15 01:40:39 -0800 | [diff] [blame] | 272 | // Set default engine options. |
| 273 | { |
| 274 | AudioOptions options; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 275 | options.echo_cancellation = true; |
| 276 | options.auto_gain_control = true; |
| 277 | options.noise_suppression = true; |
| 278 | options.highpass_filter = true; |
| 279 | options.stereo_swapping = false; |
| 280 | options.audio_jitter_buffer_max_packets = 50; |
| 281 | options.audio_jitter_buffer_fast_accelerate = false; |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 282 | options.audio_jitter_buffer_min_delay_ms = 0; |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 283 | options.audio_jitter_buffer_enable_rtx_handling = false; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 284 | options.typing_detection = true; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 285 | options.experimental_agc = false; |
| 286 | options.extended_filter_aec = false; |
| 287 | options.delay_agnostic_aec = false; |
| 288 | options.experimental_ns = false; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 289 | options.residual_echo_detector = true; |
solenberg | ff97631 | 2016-03-30 23:28:51 -0700 | [diff] [blame] | 290 | bool error = ApplyOptions(options); |
| 291 | RTC_DCHECK(error); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 292 | } |
| 293 | |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 294 | initialized_ = true; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 295 | } |
| 296 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 297 | rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState() |
| 298 | const { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 299 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 300 | return audio_state_; |
| 301 | } |
| 302 | |
Sebastian Jansson | 84848f2 | 2018-11-16 10:40:36 +0100 | [diff] [blame] | 303 | VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel( |
nisse | 51542be | 2016-02-12 02:27:06 -0800 | [diff] [blame] | 304 | webrtc::Call* call, |
| 305 | const MediaConfig& config, |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 306 | const AudioOptions& options, |
| 307 | const webrtc::CryptoOptions& crypto_options) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 308 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 309 | return new WebRtcVoiceMediaChannel(this, config, options, crypto_options, |
| 310 | call); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 311 | } |
| 312 | |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 313 | bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 314 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 315 | RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: " |
| 316 | << options_in.ToString(); |
solenberg | 0f7d293 | 2016-01-15 01:40:39 -0800 | [diff] [blame] | 317 | AudioOptions options = options_in; // The options are modified below. |
solenberg | 246b817 | 2015-12-08 09:50:23 -0800 | [diff] [blame] | 318 | |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 319 | // Set and adjust echo canceller options. |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 320 | // kEcConference is AEC with high suppression. |
| 321 | webrtc::EcModes ec_mode = webrtc::kEcConference; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 322 | |
kjellander | fcfc804 | 2016-01-14 11:01:09 -0800 | [diff] [blame] | 323 | #if defined(WEBRTC_IOS) |
Jonathan Yu | 7622048 | 2017-12-21 04:18:07 -0800 | [diff] [blame] | 324 | if (options.ios_force_software_aec_HACK && |
| 325 | *options.ios_force_software_aec_HACK) { |
| 326 | // EC may be forced on for a device known to have non-functioning platform |
| 327 | // AEC. |
| 328 | options.echo_cancellation = true; |
| 329 | options.extended_filter_aec = true; |
| 330 | RTC_LOG(LS_WARNING) |
| 331 | << "Force software AEC on iOS. May conflict with platform AEC."; |
| 332 | } else { |
| 333 | // On iOS, VPIO provides built-in EC. |
| 334 | options.echo_cancellation = false; |
| 335 | options.extended_filter_aec = false; |
| 336 | RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead."; |
| 337 | } |
Mirko Bonadei | c8c71b9 | 2017-10-16 11:08:54 +0200 | [diff] [blame] | 338 | #elif defined(WEBRTC_ANDROID) |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 339 | ec_mode = webrtc::kEcAecm; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 340 | options.extended_filter_aec = false; |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 341 | #endif |
| 342 | |
Bjorn Volcker | bf395c1 | 2015-03-25 22:45:56 +0100 | [diff] [blame] | 343 | // Delay Agnostic AEC automatically turns on EC if not set except on iOS |
| 344 | // where the feature is not supported. |
| 345 | bool use_delay_agnostic_aec = false; |
kjellander | fcfc804 | 2016-01-14 11:01:09 -0800 | [diff] [blame] | 346 | #if !defined(WEBRTC_IOS) |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 347 | if (options.delay_agnostic_aec) { |
| 348 | use_delay_agnostic_aec = *options.delay_agnostic_aec; |
Bjorn Volcker | bf395c1 | 2015-03-25 22:45:56 +0100 | [diff] [blame] | 349 | if (use_delay_agnostic_aec) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 350 | options.echo_cancellation = true; |
| 351 | options.extended_filter_aec = true; |
Bjorn Volcker | bf395c1 | 2015-03-25 22:45:56 +0100 | [diff] [blame] | 352 | ec_mode = webrtc::kEcConference; |
| 353 | } |
| 354 | } |
| 355 | #endif |
| 356 | |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 357 | // Set and adjust noise suppressor options. |
| 358 | #if defined(WEBRTC_IOS) |
| 359 | // On iOS, VPIO provides built-in NS. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 360 | options.noise_suppression = false; |
| 361 | options.typing_detection = false; |
| 362 | options.experimental_ns = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 363 | RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead."; |
Mirko Bonadei | c8c71b9 | 2017-10-16 11:08:54 +0200 | [diff] [blame] | 364 | #elif defined(WEBRTC_ANDROID) |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 365 | options.typing_detection = false; |
| 366 | options.experimental_ns = false; |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 367 | #endif |
| 368 | |
| 369 | // Set and adjust gain control options. |
| 370 | #if defined(WEBRTC_IOS) |
| 371 | // On iOS, VPIO provides built-in AGC. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 372 | options.auto_gain_control = false; |
| 373 | options.experimental_agc = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 374 | RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead."; |
Mirko Bonadei | c8c71b9 | 2017-10-16 11:08:54 +0200 | [diff] [blame] | 375 | #elif defined(WEBRTC_ANDROID) |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 376 | options.experimental_agc = false; |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 377 | #endif |
| 378 | |
| 379 | #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) |
Mirko Bonadei | c8c71b9 | 2017-10-16 11:08:54 +0200 | [diff] [blame] | 380 | // Turn off the gain control if specified by the field trial. |
| 381 | // The purpose of the field trial is to reduce the amount of resampling |
| 382 | // performed inside the audio processing module on mobile platforms by |
| 383 | // whenever possible turning off the fixed AGC mode and the high-pass filter. |
| 384 | // (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181). |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 385 | if (webrtc::field_trial::IsEnabled( |
| 386 | "WebRTC-Audio-MinimizeResamplingOnMobile")) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 387 | options.auto_gain_control = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 388 | RTC_LOG(LS_INFO) << "Disable AGC according to field trial."; |
Steve Anton | e78bcb9 | 2017-10-31 09:53:08 -0700 | [diff] [blame] | 389 | if (!(options.noise_suppression.value_or(false) || |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 390 | options.echo_cancellation.value_or(false))) { |
| 391 | // If possible, turn off the high-pass filter. |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 392 | RTC_LOG(LS_INFO) |
| 393 | << "Disable high-pass filter in response to field trial."; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 394 | options.highpass_filter = false; |
peah | 8a8ebd9 | 2017-05-22 15:48:47 -0700 | [diff] [blame] | 395 | } |
| 396 | } |
| 397 | #endif |
| 398 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 399 | if (options.echo_cancellation) { |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 +0000 | [diff] [blame] | 400 | // Check if platform supports built-in EC. Currently only supported on |
| 401 | // Android and in combination with Java based audio layer. |
| 402 | // TODO(henrika): investigate possibility to support built-in EC also |
| 403 | // in combination with Open SL ES audio. |
solenberg | 5b5129a | 2016-04-08 05:35:48 -0700 | [diff] [blame] | 404 | const bool built_in_aec = adm()->BuiltInAECIsAvailable(); |
Bjorn Volcker | 73f7210 | 2015-06-03 14:50:15 +0200 | [diff] [blame] | 405 | if (built_in_aec) { |
Bjorn Volcker | ccfc939 | 2015-05-07 07:43:17 +0200 | [diff] [blame] | 406 | // Built-in EC exists on this device and use_delay_agnostic_aec is not |
| 407 | // overriding it. Enable/Disable it according to the echo_cancellation |
| 408 | // audio option. |
Bjorn Volcker | 73f7210 | 2015-06-03 14:50:15 +0200 | [diff] [blame] | 409 | const bool enable_built_in_aec = |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 410 | *options.echo_cancellation && !use_delay_agnostic_aec; |
solenberg | 5b5129a | 2016-04-08 05:35:48 -0700 | [diff] [blame] | 411 | if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 && |
Bjorn Volcker | 73f7210 | 2015-06-03 14:50:15 +0200 | [diff] [blame] | 412 | enable_built_in_aec) { |
Bjorn Volcker | bf395c1 | 2015-03-25 22:45:56 +0100 | [diff] [blame] | 413 | // Disable internal software EC if built-in EC is enabled, |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 +0000 | [diff] [blame] | 414 | // i.e., replace the software EC with the built-in EC. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 415 | options.echo_cancellation = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 416 | RTC_LOG(LS_INFO) |
| 417 | << "Disabling EC since built-in EC will be used instead"; |
henrika@webrtc.org | a954c07 | 2014-12-09 16:22:09 +0000 | [diff] [blame] | 418 | } |
| 419 | } |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 420 | webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation, |
| 421 | ec_mode); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 422 | } |
| 423 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 424 | if (options.auto_gain_control) { |
peah | 72a5645 | 2016-08-22 12:08:55 -0700 | [diff] [blame] | 425 | bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable(); |
| 426 | if (built_in_agc_avaliable) { |
solenberg | 5b5129a | 2016-04-08 05:35:48 -0700 | [diff] [blame] | 427 | if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 && |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 428 | *options.auto_gain_control) { |
henrika | c14f5ff | 2015-09-23 14:08:33 +0200 | [diff] [blame] | 429 | // Disable internal software AGC if built-in AGC is enabled, |
| 430 | // i.e., replace the software AGC with the built-in AGC. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 431 | options.auto_gain_control = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 432 | RTC_LOG(LS_INFO) |
| 433 | << "Disabling AGC since built-in AGC will be used instead"; |
henrika | c14f5ff | 2015-09-23 14:08:33 +0200 | [diff] [blame] | 434 | } |
| 435 | } |
henrika | e26456a | 2017-12-13 14:08:48 +0100 | [diff] [blame] | 436 | webrtc::apm_helpers::SetAgcStatus(apm(), *options.auto_gain_control); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 437 | } |
| 438 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 439 | if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain || |
Steve Anton | 606a597 | 2017-12-07 14:31:01 -0800 | [diff] [blame] | 440 | options.tx_agc_limiter) { |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 441 | // Override default_agc_config_. Generally, an unset option means "leave |
| 442 | // the VoE bits alone" in this function, so we want whatever is set to be |
| 443 | // stored as the new "default". If we didn't, then setting e.g. |
| 444 | // tx_agc_target_dbov would reset digital compression gain and limiter |
| 445 | // settings. |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 446 | default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or( |
| 447 | default_agc_config_.targetLeveldBOv); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 448 | default_agc_config_.digitalCompressionGaindB = |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 449 | options.tx_agc_digital_compression_gain.value_or( |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 450 | default_agc_config_.digitalCompressionGaindB); |
| 451 | default_agc_config_.limiterEnable = |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 452 | options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable); |
Steve Anton | 606a597 | 2017-12-07 14:31:01 -0800 | [diff] [blame] | 453 | webrtc::apm_helpers::SetAgcConfig(apm(), default_agc_config_); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 454 | } |
| 455 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 456 | if (options.noise_suppression) { |
Alejandro Luebs | c9b0c26 | 2016-05-16 15:32:38 -0700 | [diff] [blame] | 457 | if (adm()->BuiltInNSIsAvailable()) { |
Alessio Bazzica | cc22f51 | 2018-08-30 13:01:34 +0200 | [diff] [blame] | 458 | bool builtin_ns = *options.noise_suppression; |
Alejandro Luebs | c9b0c26 | 2016-05-16 15:32:38 -0700 | [diff] [blame] | 459 | if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) { |
henrika | c14f5ff | 2015-09-23 14:08:33 +0200 | [diff] [blame] | 460 | // Disable internal software NS if built-in NS is enabled, |
| 461 | // i.e., replace the software NS with the built-in NS. |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 462 | options.noise_suppression = false; |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 463 | RTC_LOG(LS_INFO) |
| 464 | << "Disabling NS since built-in NS will be used instead"; |
henrika | c14f5ff | 2015-09-23 14:08:33 +0200 | [diff] [blame] | 465 | } |
| 466 | } |
solenberg | 76377c5 | 2017-02-21 00:54:31 -0800 | [diff] [blame] | 467 | webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 468 | } |
| 469 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 470 | if (options.stereo_swapping) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 471 | RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping; |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 472 | audio_state()->SetStereoChannelSwapping(*options.stereo_swapping); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 473 | } |
| 474 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 475 | if (options.audio_jitter_buffer_max_packets) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 476 | RTC_LOG(LS_INFO) << "NetEq capacity is " |
| 477 | << *options.audio_jitter_buffer_max_packets; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 478 | audio_jitter_buffer_max_packets_ = |
solenberg | 88499ec | 2016-09-07 07:34:41 -0700 | [diff] [blame] | 479 | std::max(20, *options.audio_jitter_buffer_max_packets); |
Henrik Lundin | 64dad83 | 2015-05-11 12:44:23 +0200 | [diff] [blame] | 480 | } |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 481 | if (options.audio_jitter_buffer_fast_accelerate) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 482 | RTC_LOG(LS_INFO) << "NetEq fast mode? " |
| 483 | << *options.audio_jitter_buffer_fast_accelerate; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 484 | audio_jitter_buffer_fast_accelerate_ = |
solenberg | 88499ec | 2016-09-07 07:34:41 -0700 | [diff] [blame] | 485 | *options.audio_jitter_buffer_fast_accelerate; |
Henrik Lundin | 5263b3c | 2015-06-01 10:29:41 +0200 | [diff] [blame] | 486 | } |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 487 | if (options.audio_jitter_buffer_min_delay_ms) { |
| 488 | RTC_LOG(LS_INFO) << "NetEq minimum delay is " |
| 489 | << *options.audio_jitter_buffer_min_delay_ms; |
| 490 | audio_jitter_buffer_min_delay_ms_ = |
| 491 | *options.audio_jitter_buffer_min_delay_ms; |
| 492 | } |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 493 | if (options.audio_jitter_buffer_enable_rtx_handling) { |
| 494 | RTC_LOG(LS_INFO) << "NetEq handle reordered packets? " |
| 495 | << *options.audio_jitter_buffer_enable_rtx_handling; |
| 496 | audio_jitter_buffer_enable_rtx_handling_ = |
| 497 | *options.audio_jitter_buffer_enable_rtx_handling; |
| 498 | } |
Henrik Lundin | 5263b3c | 2015-06-01 10:29:41 +0200 | [diff] [blame] | 499 | |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 +0000 | [diff] [blame] | 500 | webrtc::Config config; |
| 501 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 502 | if (options.delay_agnostic_aec) |
| 503 | delay_agnostic_aec_ = options.delay_agnostic_aec; |
| 504 | if (delay_agnostic_aec_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 505 | RTC_LOG(LS_INFO) << "Delay agnostic aec is enabled? " |
| 506 | << *delay_agnostic_aec_; |
henrik.lundin | 0f133b9 | 2015-07-02 00:17:55 -0700 | [diff] [blame] | 507 | config.Set<webrtc::DelayAgnostic>( |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 508 | new webrtc::DelayAgnostic(*delay_agnostic_aec_)); |
Bjorn Volcker | bf395c1 | 2015-03-25 22:45:56 +0100 | [diff] [blame] | 509 | } |
| 510 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 511 | if (options.extended_filter_aec) { |
| 512 | extended_filter_aec_ = options.extended_filter_aec; |
| 513 | } |
| 514 | if (extended_filter_aec_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 515 | RTC_LOG(LS_INFO) << "Extended filter aec is enabled? " |
| 516 | << *extended_filter_aec_; |
Henrik Lundin | 441f634 | 2015-06-09 16:03:13 +0200 | [diff] [blame] | 517 | config.Set<webrtc::ExtendedFilter>( |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 518 | new webrtc::ExtendedFilter(*extended_filter_aec_)); |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 +0000 | [diff] [blame] | 519 | } |
| 520 | |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 521 | if (options.experimental_ns) { |
| 522 | experimental_ns_ = options.experimental_ns; |
| 523 | } |
| 524 | if (experimental_ns_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 525 | RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_; |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 +0000 | [diff] [blame] | 526 | config.Set<webrtc::ExperimentalNs>( |
kwiberg | 102c6a6 | 2015-10-30 02:47:38 -0700 | [diff] [blame] | 527 | new webrtc::ExperimentalNs(*experimental_ns_)); |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 +0000 | [diff] [blame] | 528 | } |
buildbot@webrtc.org | 1f8a237 | 2014-08-28 10:52:44 +0000 | [diff] [blame] | 529 | |
peah | b1c9d1d | 2017-07-25 15:45:24 -0700 | [diff] [blame] | 530 | webrtc::AudioProcessing::Config apm_config = apm()->GetConfig(); |
| 531 | |
peah | 8271d04 | 2016-11-22 07:24:52 -0800 | [diff] [blame] | 532 | if (options.highpass_filter) { |
peah | b1c9d1d | 2017-07-25 15:45:24 -0700 | [diff] [blame] | 533 | apm_config.high_pass_filter.enabled = *options.highpass_filter; |
peah | 8271d04 | 2016-11-22 07:24:52 -0800 | [diff] [blame] | 534 | } |
| 535 | |
ivoc | 4ca1869 | 2017-02-10 05:11:09 -0800 | [diff] [blame] | 536 | if (options.residual_echo_detector) { |
peah | b1c9d1d | 2017-07-25 15:45:24 -0700 | [diff] [blame] | 537 | apm_config.residual_echo_detector.enabled = *options.residual_echo_detector; |
ivoc | 4ca1869 | 2017-02-10 05:11:09 -0800 | [diff] [blame] | 538 | } |
| 539 | |
Sam Zackrisson | ba50223 | 2019-01-04 10:36:48 +0100 | [diff] [blame] | 540 | if (options.typing_detection) { |
| 541 | RTC_LOG(LS_INFO) << "Typing detection is enabled? " |
| 542 | << *options.typing_detection; |
| 543 | apm_config.voice_detection.enabled = *options.typing_detection; |
| 544 | } |
| 545 | |
solenberg | 059fb44 | 2016-10-26 05:12:24 -0700 | [diff] [blame] | 546 | apm()->SetExtraOptions(config); |
peah | b1c9d1d | 2017-07-25 15:45:24 -0700 | [diff] [blame] | 547 | apm()->ApplyConfig(apm_config); |
buildbot@webrtc.org | 13d6776 | 2014-05-02 17:33:29 +0000 | [diff] [blame] | 548 | return true; |
| 549 | } |
| 550 | |
ossu | dedfd28 | 2016-06-14 07:12:39 -0700 | [diff] [blame] | 551 | const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const { |
| 552 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 553 | return send_codecs_; |
ossu | dedfd28 | 2016-06-14 07:12:39 -0700 | [diff] [blame] | 554 | } |
| 555 | |
| 556 | const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 557 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 558 | return recv_codecs_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 559 | } |
| 560 | |
Stefan Holmer | 9d69c3f | 2015-12-07 10:45:43 +0100 | [diff] [blame] | 561 | RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 562 | RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
Stefan Holmer | 9d69c3f | 2015-12-07 10:45:43 +0100 | [diff] [blame] | 563 | RtpCapabilities capabilities; |
Stefan Holmer | 9d69c3f | 2015-12-07 10:45:43 +0100 | [diff] [blame] | 564 | capabilities.header_extensions.push_back( |
isheriff | 6f8d686 | 2016-05-26 11:24:55 -0700 | [diff] [blame] | 565 | webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, |
| 566 | webrtc::RtpExtension::kAudioLevelDefaultId)); |
Alex Narest | bcf9180 | 2018-06-25 16:08:36 +0200 | [diff] [blame] | 567 | if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") && |
| 568 | !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) { |
isheriff | 6f8d686 | 2016-05-26 11:24:55 -0700 | [diff] [blame] | 569 | capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| 570 | webrtc::RtpExtension::kTransportSequenceNumberUri, |
| 571 | webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
stefan | ba4c0e4 | 2016-02-04 04:12:24 -0800 | [diff] [blame] | 572 | } |
Amit Hilbuch | 77938e6 | 2018-12-21 09:23:38 -0800 | [diff] [blame] | 573 | |
Stefan Holmer | 9d69c3f | 2015-12-07 10:45:43 +0100 | [diff] [blame] | 574 | return capabilities; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 575 | } |
| 576 | |
solenberg | 63b3454 | 2015-09-29 06:06:31 -0700 | [diff] [blame] | 577 | void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 578 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 579 | RTC_DCHECK(channel); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 580 | channels_.push_back(channel); |
| 581 | } |
| 582 | |
solenberg | 63b3454 | 2015-09-29 06:06:31 -0700 | [diff] [blame] | 583 | void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 584 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 63b3454 | 2015-09-29 06:06:31 -0700 | [diff] [blame] | 585 | auto it = std::find(channels_.begin(), channels_.end(), channel); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 586 | RTC_DCHECK(it != channels_.end()); |
| 587 | channels_.erase(it); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 588 | } |
| 589 | |
ivoc | d66b44d | 2016-01-15 03:06:36 -0800 | [diff] [blame] | 590 | bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
| 591 | int64_t max_size_bytes) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 592 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 593 | auto aec_dump = webrtc::AecDumpFactory::Create( |
| 594 | file, max_size_bytes, low_priority_worker_queue_.get()); |
aleloi | 048cbdd | 2017-05-29 02:56:27 -0700 | [diff] [blame] | 595 | if (!aec_dump) { |
wu@webrtc.org | a8910d2 | 2014-01-23 22:12:45 +0000 | [diff] [blame] | 596 | return false; |
| 597 | } |
aleloi | 048cbdd | 2017-05-29 02:56:27 -0700 | [diff] [blame] | 598 | apm()->AttachAecDump(std::move(aec_dump)); |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 +0000 | [diff] [blame] | 599 | return true; |
wu@webrtc.org | a989080 | 2013-12-13 00:21:03 +0000 | [diff] [blame] | 600 | } |
| 601 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 602 | void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 603 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
aleloi | 048cbdd | 2017-05-29 02:56:27 -0700 | [diff] [blame] | 604 | |
deadbeef | eb02c03 | 2017-06-15 08:29:25 -0700 | [diff] [blame] | 605 | auto aec_dump = webrtc::AecDumpFactory::Create( |
| 606 | filename, -1, low_priority_worker_queue_.get()); |
aleloi | 048cbdd | 2017-05-29 02:56:27 -0700 | [diff] [blame] | 607 | if (aec_dump) { |
| 608 | apm()->AttachAecDump(std::move(aec_dump)); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 609 | } |
| 610 | } |
| 611 | |
| 612 | void WebRtcVoiceEngine::StopAecDump() { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 613 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
aleloi | 048cbdd | 2017-05-29 02:56:27 -0700 | [diff] [blame] | 614 | apm()->DetachAecDump(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 615 | } |
| 616 | |
solenberg | 5b5129a | 2016-04-08 05:35:48 -0700 | [diff] [blame] | 617 | webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { |
| 618 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 619 | RTC_DCHECK(adm_); |
Fredrik Solenberg | d319534 | 2017-11-21 20:33:05 +0100 | [diff] [blame] | 620 | return adm_.get(); |
solenberg | 5b5129a | 2016-04-08 05:35:48 -0700 | [diff] [blame] | 621 | } |
| 622 | |
peah | b1c9d1d | 2017-07-25 15:45:24 -0700 | [diff] [blame] | 623 | webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const { |
solenberg | 059fb44 | 2016-10-26 05:12:24 -0700 | [diff] [blame] | 624 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 55900fd | 2017-11-23 20:22:55 +0100 | [diff] [blame] | 625 | RTC_DCHECK(apm_); |
peah | a9cc40b | 2017-06-29 08:32:09 -0700 | [diff] [blame] | 626 | return apm_.get(); |
solenberg | 059fb44 | 2016-10-26 05:12:24 -0700 | [diff] [blame] | 627 | } |
| 628 | |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 629 | webrtc::AudioState* WebRtcVoiceEngine::audio_state() { |
solenberg | 76377c5 | 2017-02-21 00:54:31 -0800 | [diff] [blame] | 630 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 631 | RTC_DCHECK(audio_state_); |
| 632 | return audio_state_.get(); |
solenberg | 76377c5 | 2017-02-21 00:54:31 -0800 | [diff] [blame] | 633 | } |
| 634 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 635 | AudioCodecs WebRtcVoiceEngine::CollectCodecs( |
| 636 | const std::vector<webrtc::AudioCodecSpec>& specs) const { |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 637 | PayloadTypeMapper mapper; |
| 638 | AudioCodecs out; |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 639 | |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 640 | // Only generate CN payload types for these clockrates: |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 641 | std::map<int, bool, std::greater<int>> generate_cn = { |
| 642 | {8000, false}, {16000, false}, {32000, false}}; |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 643 | // Only generate telephone-event payload types for these clockrates: |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 644 | std::map<int, bool, std::greater<int>> generate_dtmf = { |
| 645 | {8000, false}, {16000, false}, {32000, false}, {48000, false}}; |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 646 | |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 647 | auto map_format = [&mapper](const webrtc::SdpAudioFormat& format, |
| 648 | AudioCodecs* out) { |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 649 | absl::optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format); |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 650 | if (opt_codec) { |
| 651 | if (out) { |
| 652 | out->push_back(*opt_codec); |
| 653 | } |
| 654 | } else { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 655 | RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: " |
Jonas Olsson | abbe841 | 2018-04-03 13:40:05 +0200 | [diff] [blame] | 656 | << rtc::ToString(format); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 657 | } |
| 658 | |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 659 | return opt_codec; |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 660 | }; |
| 661 | |
ossu | d4e9f62 | 2016-08-18 02:01:17 -0700 | [diff] [blame] | 662 | for (const auto& spec : specs) { |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 663 | // We need to do some extra stuff before adding the main codecs to out. |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 664 | absl::optional<AudioCodec> opt_codec = map_format(spec.format, nullptr); |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 665 | if (opt_codec) { |
| 666 | AudioCodec& codec = *opt_codec; |
ossu | a1a040a | 2017-04-06 10:03:21 -0700 | [diff] [blame] | 667 | if (spec.info.supports_network_adaption) { |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 668 | codec.AddFeedbackParam( |
| 669 | FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty)); |
| 670 | } |
| 671 | |
ossu | a1a040a | 2017-04-06 10:03:21 -0700 | [diff] [blame] | 672 | if (spec.info.allow_comfort_noise) { |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 673 | // Generate a CN entry if the decoder allows it and we support the |
| 674 | // clockrate. |
| 675 | auto cn = generate_cn.find(spec.format.clockrate_hz); |
| 676 | if (cn != generate_cn.end()) { |
| 677 | cn->second = true; |
| 678 | } |
| 679 | } |
| 680 | |
| 681 | // Generate a telephone-event entry if we support the clockrate. |
| 682 | auto dtmf = generate_dtmf.find(spec.format.clockrate_hz); |
| 683 | if (dtmf != generate_dtmf.end()) { |
| 684 | dtmf->second = true; |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 685 | } |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 686 | |
| 687 | out.push_back(codec); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 688 | } |
| 689 | } |
| 690 | |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 691 | // Add CN codecs after "proper" audio codecs. |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 692 | for (const auto& cn : generate_cn) { |
| 693 | if (cn.second) { |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 694 | map_format({kCnCodecName, cn.first, 1}, &out); |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 695 | } |
| 696 | } |
| 697 | |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 698 | // Add telephone-event codecs last. |
| 699 | for (const auto& dtmf : generate_dtmf) { |
| 700 | if (dtmf.second) { |
ossu | 9def800 | 2017-02-09 05:14:32 -0800 | [diff] [blame] | 701 | map_format({kDtmfCodecName, dtmf.first, 1}, &out); |
solenberg | 2779bab | 2016-11-17 04:45:19 -0800 | [diff] [blame] | 702 | } |
| 703 | } |
ossu | c54071d | 2016-08-17 02:45:41 -0700 | [diff] [blame] | 704 | |
| 705 | return out; |
| 706 | } |
| 707 | |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 708 | class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 709 | : public AudioSource::Sink { |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 710 | public: |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 711 | WebRtcAudioSendStream( |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 712 | uint32_t ssrc, |
Steve Anton | bb50ce5 | 2018-03-26 10:24:32 -0700 | [diff] [blame] | 713 | const std::string& mid, |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 714 | const std::string& c_name, |
Alex Narest | b3944f0 | 2017-10-13 14:56:18 +0200 | [diff] [blame] | 715 | const std::string track_id, |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 716 | const absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>& |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 717 | send_codec_spec, |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 718 | bool extmap_allow_mixed, |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 719 | const std::vector<webrtc::RtpExtension>& extensions, |
| 720 | int max_send_bitrate_bps, |
Jiawei Ou | 5571812 | 2018-11-09 13:17:39 -0800 | [diff] [blame] | 721 | int rtcp_report_interval_ms, |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 722 | const absl::optional<std::string>& audio_network_adaptor_config, |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 723 | webrtc::Call* call, |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 724 | webrtc::Transport* send_transport, |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 725 | webrtc::MediaTransportInterface* media_transport, |
Karl Wiberg | 77490b9 | 2018-03-21 15:18:42 +0100 | [diff] [blame] | 726 | const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory, |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 727 | const absl::optional<webrtc::AudioCodecPairId> codec_pair_id, |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 728 | rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor, |
| 729 | const webrtc::CryptoOptions& crypto_options) |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 730 | : call_(call), |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 731 | config_(send_transport, media_transport), |
sprang | c1b57a1 | 2017-02-28 08:50:47 -0800 | [diff] [blame] | 732 | send_side_bwe_with_overhead_( |
| 733 | webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 734 | max_send_bitrate_bps_(max_send_bitrate_bps), |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 735 | rtp_parameters_(CreateRtpParametersWithOneEncoding()) { |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 736 | RTC_DCHECK(call); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 737 | RTC_DCHECK(encoder_factory); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 738 | config_.rtp.ssrc = ssrc; |
Steve Anton | bb50ce5 | 2018-03-26 10:24:32 -0700 | [diff] [blame] | 739 | config_.rtp.mid = mid; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 740 | config_.rtp.c_name = c_name; |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 741 | config_.rtp.extmap_allow_mixed = extmap_allow_mixed; |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 742 | config_.rtp.extensions = extensions; |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 743 | config_.has_dscp = rtp_parameters_.encodings[0].network_priority != |
| 744 | webrtc::kDefaultBitratePriority; |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 745 | config_.audio_network_adaptor_config = audio_network_adaptor_config; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 746 | config_.encoder_factory = encoder_factory; |
Karl Wiberg | 77490b9 | 2018-03-21 15:18:42 +0100 | [diff] [blame] | 747 | config_.codec_pair_id = codec_pair_id; |
Alex Narest | b3944f0 | 2017-10-13 14:56:18 +0200 | [diff] [blame] | 748 | config_.track_id = track_id; |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 749 | config_.frame_encryptor = frame_encryptor; |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 750 | config_.crypto_options = crypto_options; |
Jiawei Ou | 5571812 | 2018-11-09 13:17:39 -0800 | [diff] [blame] | 751 | config_.rtcp_report_interval_ms = rtcp_report_interval_ms; |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 752 | rtp_parameters_.encodings[0].ssrc = ssrc; |
Florent Castelli | dacec71 | 2018-05-24 16:24:21 +0200 | [diff] [blame] | 753 | rtp_parameters_.rtcp.cname = c_name; |
Florent Castelli | abe301f | 2018-06-12 18:33:49 +0200 | [diff] [blame] | 754 | rtp_parameters_.header_extensions = extensions; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 755 | |
| 756 | if (send_codec_spec) { |
| 757 | UpdateSendCodecSpec(*send_codec_spec); |
| 758 | } |
| 759 | |
| 760 | stream_ = call_->CreateAudioSendStream(config_); |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 761 | } |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 762 | |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 763 | ~WebRtcAudioSendStream() override { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 764 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 765 | ClearSource(); |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 766 | call_->DestroyAudioSendStream(stream_); |
| 767 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 768 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 769 | void SetSendCodecSpec( |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 770 | const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 771 | UpdateSendCodecSpec(send_codec_spec); |
| 772 | ReconfigureAudioSendStream(); |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 773 | } |
| 774 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 775 | void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions) { |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 776 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 777 | config_.rtp.extensions = extensions; |
Florent Castelli | abe301f | 2018-06-12 18:33:49 +0200 | [diff] [blame] | 778 | rtp_parameters_.header_extensions = extensions; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 779 | ReconfigureAudioSendStream(); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 780 | } |
| 781 | |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 782 | void SetExtmapAllowMixed(bool extmap_allow_mixed) { |
| 783 | config_.rtp.extmap_allow_mixed = extmap_allow_mixed; |
| 784 | ReconfigureAudioSendStream(); |
| 785 | } |
| 786 | |
Steve Anton | bb50ce5 | 2018-03-26 10:24:32 -0700 | [diff] [blame] | 787 | void SetMid(const std::string& mid) { |
| 788 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 789 | if (config_.rtp.mid == mid) { |
| 790 | return; |
| 791 | } |
| 792 | config_.rtp.mid = mid; |
| 793 | ReconfigureAudioSendStream(); |
| 794 | } |
| 795 | |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 796 | void SetFrameEncryptor( |
| 797 | rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) { |
| 798 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 799 | config_.frame_encryptor = frame_encryptor; |
| 800 | ReconfigureAudioSendStream(); |
| 801 | } |
| 802 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 803 | void SetAudioNetworkAdaptorConfig( |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 804 | const absl::optional<std::string>& audio_network_adaptor_config) { |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 805 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 806 | if (config_.audio_network_adaptor_config == audio_network_adaptor_config) { |
| 807 | return; |
| 808 | } |
| 809 | config_.audio_network_adaptor_config = audio_network_adaptor_config; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 810 | UpdateAllowedBitrateRange(); |
| 811 | ReconfigureAudioSendStream(); |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 812 | } |
| 813 | |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 814 | bool SetMaxSendBitrate(int bps) { |
| 815 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 816 | RTC_DCHECK(config_.send_codec_spec); |
| 817 | RTC_DCHECK(audio_codec_spec_); |
| 818 | auto send_rate = ComputeSendBitrate( |
| 819 | bps, rtp_parameters_.encodings[0].max_bitrate_bps, *audio_codec_spec_); |
| 820 | |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 821 | if (!send_rate) { |
| 822 | return false; |
| 823 | } |
| 824 | |
| 825 | max_send_bitrate_bps_ = bps; |
| 826 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 827 | if (send_rate != config_.send_codec_spec->target_bitrate_bps) { |
| 828 | config_.send_codec_spec->target_bitrate_bps = send_rate; |
| 829 | ReconfigureAudioSendStream(); |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 830 | } |
| 831 | return true; |
| 832 | } |
| 833 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 834 | bool SendTelephoneEvent(int payload_type, |
| 835 | int payload_freq, |
| 836 | int event, |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 837 | int duration_ms) { |
Fredrik Solenberg | b572768 | 2015-12-04 15:22:19 +0100 | [diff] [blame] | 838 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 839 | RTC_DCHECK(stream_); |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 840 | return stream_->SendTelephoneEvent(payload_type, payload_freq, event, |
| 841 | duration_ms); |
Fredrik Solenberg | b572768 | 2015-12-04 15:22:19 +0100 | [diff] [blame] | 842 | } |
| 843 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 844 | void SetSend(bool send) { |
| 845 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 846 | send_ = send; |
| 847 | UpdateSendState(); |
| 848 | } |
| 849 | |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 850 | void SetMuted(bool muted) { |
| 851 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 852 | RTC_DCHECK(stream_); |
| 853 | stream_->SetMuted(muted); |
| 854 | muted_ = muted; |
| 855 | } |
| 856 | |
| 857 | bool muted() const { |
| 858 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 859 | return muted_; |
| 860 | } |
| 861 | |
Ivo Creusen | 56d4609 | 2017-11-24 17:29:59 +0100 | [diff] [blame] | 862 | webrtc::AudioSendStream::Stats GetStats(bool has_remote_tracks) const { |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 863 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 864 | RTC_DCHECK(stream_); |
Ivo Creusen | 56d4609 | 2017-11-24 17:29:59 +0100 | [diff] [blame] | 865 | return stream_->GetStats(has_remote_tracks); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 866 | } |
| 867 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 868 | // Starts the sending by setting ourselves as a sink to the AudioSource to |
| 869 | // get data callbacks. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 870 | // This method is called on the libjingle worker thread. |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 871 | // TODO(xians): Make sure Start() is called only once. |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 872 | void SetSource(AudioSource* source) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 873 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 874 | RTC_DCHECK(source); |
| 875 | if (source_) { |
| 876 | RTC_DCHECK(source_ == source); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 877 | return; |
| 878 | } |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 879 | source->SetSink(this); |
| 880 | source_ = source; |
| 881 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 882 | } |
| 883 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 884 | // Stops sending by setting the sink of the AudioSource to nullptr. No data |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 885 | // callback will be received after this method. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 886 | // This method is called on the libjingle worker thread. |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 887 | void ClearSource() { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 888 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 889 | if (source_) { |
| 890 | source_->SetSink(nullptr); |
| 891 | source_ = nullptr; |
solenberg | 98c6886 | 2015-10-09 03:27:14 -0700 | [diff] [blame] | 892 | } |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 893 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 894 | } |
| 895 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 896 | // AudioSource::Sink implementation. |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 897 | // This method is called on the audio thread. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 898 | void OnData(const void* audio_data, |
| 899 | int bits_per_sample, |
| 900 | int sample_rate, |
Peter Kasting | 6955870 | 2016-01-12 16:26:35 -0800 | [diff] [blame] | 901 | size_t number_of_channels, |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 902 | size_t number_of_frames) override { |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 903 | RTC_DCHECK_EQ(16, bits_per_sample); |
solenberg | 347ec5c | 2016-09-23 04:21:47 -0700 | [diff] [blame] | 904 | RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_); |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 905 | RTC_DCHECK(stream_); |
| 906 | std::unique_ptr<webrtc::AudioFrame> audio_frame(new webrtc::AudioFrame()); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 907 | audio_frame->UpdateFrame( |
| 908 | audio_frame->timestamp_, static_cast<const int16_t*>(audio_data), |
| 909 | number_of_frames, sample_rate, audio_frame->speech_type_, |
| 910 | audio_frame->vad_activity_, number_of_channels); |
Fredrik Solenberg | 2a87797 | 2017-12-15 16:42:15 +0100 | [diff] [blame] | 911 | stream_->SendAudioData(std::move(audio_frame)); |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 912 | } |
| 913 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 914 | // Callback from the |source_| when it is going away. In case Start() has |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 915 | // never been called, this callback won't be triggered. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 916 | void OnClose() override { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 917 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 918 | // Set |source_| to nullptr to make sure no more callback will get into |
| 919 | // the source. |
| 920 | source_ = nullptr; |
| 921 | UpdateSendState(); |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 922 | } |
| 923 | |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 924 | const webrtc::RtpParameters& rtp_parameters() const { |
| 925 | return rtp_parameters_; |
| 926 | } |
| 927 | |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 928 | webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters) { |
Florent Castelli | 892acf0 | 2018-10-01 22:47:20 +0200 | [diff] [blame] | 929 | webrtc::RTCError error = ValidateRtpParameters(rtp_parameters_, parameters); |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 930 | if (!error.ok()) { |
| 931 | return error; |
deadbeef | fb2aced | 2017-01-06 23:05:37 -0800 | [diff] [blame] | 932 | } |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 933 | |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 934 | absl::optional<int> send_rate; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 935 | if (audio_codec_spec_) { |
| 936 | send_rate = ComputeSendBitrate(max_send_bitrate_bps_, |
| 937 | parameters.encodings[0].max_bitrate_bps, |
| 938 | *audio_codec_spec_); |
| 939 | if (!send_rate) { |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 940 | return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 941 | } |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 942 | } |
| 943 | |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 944 | const absl::optional<int> old_rtp_max_bitrate = |
minyue | cecec10 | 2017-03-27 13:04:25 -0700 | [diff] [blame] | 945 | rtp_parameters_.encodings[0].max_bitrate_bps; |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 946 | double old_priority = rtp_parameters_.encodings[0].bitrate_priority; |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 947 | double old_dscp = rtp_parameters_.encodings[0].network_priority; |
Lu Liu | 8b77aea | 2017-12-20 23:48:03 +0000 | [diff] [blame] | 948 | rtp_parameters_ = parameters; |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 949 | config_.bitrate_priority = rtp_parameters_.encodings[0].bitrate_priority; |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 950 | config_.has_dscp = (rtp_parameters_.encodings[0].network_priority != |
| 951 | webrtc::kDefaultBitratePriority); |
Lu Liu | 8b77aea | 2017-12-20 23:48:03 +0000 | [diff] [blame] | 952 | |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 953 | bool reconfigure_send_stream = |
| 954 | (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) || |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 955 | (rtp_parameters_.encodings[0].bitrate_priority != old_priority) || |
| 956 | (rtp_parameters_.encodings[0].network_priority != old_dscp); |
minyue | cecec10 | 2017-03-27 13:04:25 -0700 | [diff] [blame] | 957 | if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) { |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 958 | // Update the bitrate range. |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 959 | if (send_rate) { |
| 960 | config_.send_codec_spec->target_bitrate_bps = send_rate; |
| 961 | } |
| 962 | UpdateAllowedBitrateRange(); |
Seth Hampson | d2b912a | 2017-12-20 11:56:37 -0800 | [diff] [blame] | 963 | } |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 964 | if (reconfigure_send_stream) { |
| 965 | ReconfigureAudioSendStream(); |
| 966 | } |
Florent Castelli | dacec71 | 2018-05-24 16:24:21 +0200 | [diff] [blame] | 967 | |
| 968 | rtp_parameters_.rtcp.cname = config_.rtp.c_name; |
| 969 | rtp_parameters_.rtcp.reduced_size = false; |
| 970 | |
Seth Hampson | 24722b3 | 2017-12-22 09:36:42 -0800 | [diff] [blame] | 971 | // parameters.encodings[0].active could have changed. |
| 972 | UpdateSendState(); |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 973 | return webrtc::RTCError::OK(); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 974 | } |
| 975 | |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 976 | private: |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 977 | void UpdateSendState() { |
| 978 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 979 | RTC_DCHECK(stream_); |
Taylor Brandstetter | 55dd708 | 2016-05-03 13:50:11 -0700 | [diff] [blame] | 980 | RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size()); |
| 981 | if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 982 | stream_->Start(); |
| 983 | } else { // !send || source_ = nullptr |
| 984 | stream_->Stop(); |
| 985 | } |
| 986 | } |
| 987 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 988 | void UpdateAllowedBitrateRange() { |
michaelt | 53fe19d | 2016-10-18 09:39:22 -0700 | [diff] [blame] | 989 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 990 | const bool is_opus = |
| 991 | config_.send_codec_spec && |
Niels Möller | 2edab4c | 2018-10-22 09:48:08 +0200 | [diff] [blame] | 992 | absl::EqualsIgnoreCase(config_.send_codec_spec->format.name, |
| 993 | kOpusCodecName); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 994 | if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) { |
stefan | e9f36d5 | 2017-01-24 08:18:45 -0800 | [diff] [blame] | 995 | config_.min_bitrate_bps = kOpusMinBitrateBps; |
minyue | cecec10 | 2017-03-27 13:04:25 -0700 | [diff] [blame] | 996 | |
| 997 | // This means that when RtpParameters is reset, we may change the |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 998 | // encoder's bit rate immediately (through ReconfigureAudioSendStream()), |
minyue | cecec10 | 2017-03-27 13:04:25 -0700 | [diff] [blame] | 999 | // meanwhile change the cap to the output of BWE. |
| 1000 | config_.max_bitrate_bps = |
| 1001 | rtp_parameters_.encodings[0].max_bitrate_bps |
| 1002 | ? *rtp_parameters_.encodings[0].max_bitrate_bps |
| 1003 | : kOpusBitrateFbBps; |
| 1004 | |
michaelt | 53fe19d | 2016-10-18 09:39:22 -0700 | [diff] [blame] | 1005 | // TODO(mflodman): Keep testing this and set proper values. |
| 1006 | // Note: This is an early experiment currently only supported by Opus. |
elad.alon | 0fe1216 | 2017-01-31 05:48:37 -0800 | [diff] [blame] | 1007 | if (send_side_bwe_with_overhead_) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1008 | const int max_packet_size_ms = |
| 1009 | WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60; |
michaelt | 6672b26 | 2017-01-11 10:17:59 -0800 | [diff] [blame] | 1010 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1011 | // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12) |
| 1012 | constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12; |
michaelt | 6672b26 | 2017-01-11 10:17:59 -0800 | [diff] [blame] | 1013 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1014 | int min_overhead_bps = |
| 1015 | kOverheadPerPacket * 8 * 1000 / max_packet_size_ms; |
michaelt | 6672b26 | 2017-01-11 10:17:59 -0800 | [diff] [blame] | 1016 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1017 | // We assume that |config_.max_bitrate_bps| before the next line is |
| 1018 | // a hard limit on the payload bitrate, so we add min_overhead_bps to |
| 1019 | // it to ensure that, when overhead is deducted, the payload rate |
| 1020 | // never goes beyond the limit. |
| 1021 | // Note: this also means that if a higher overhead is forced, we |
| 1022 | // cannot reach the limit. |
| 1023 | // TODO(minyue): Reconsider this when the signaling to BWE is done |
| 1024 | // through a dedicated API. |
| 1025 | config_.max_bitrate_bps += min_overhead_bps; |
michaelt | 6672b26 | 2017-01-11 10:17:59 -0800 | [diff] [blame] | 1026 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1027 | // In contrast to max_bitrate_bps, we let min_bitrate_bps always be |
| 1028 | // reachable. |
| 1029 | config_.min_bitrate_bps += min_overhead_bps; |
michaelt | 6672b26 | 2017-01-11 10:17:59 -0800 | [diff] [blame] | 1030 | } |
michaelt | 53fe19d | 2016-10-18 09:39:22 -0700 | [diff] [blame] | 1031 | } |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1032 | } |
| 1033 | |
| 1034 | void UpdateSendCodecSpec( |
| 1035 | const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) { |
| 1036 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 1037 | config_.send_codec_spec = send_codec_spec; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1038 | auto info = |
| 1039 | config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format); |
| 1040 | RTC_DCHECK(info); |
| 1041 | // If a specific target bitrate has been set for the stream, use that as |
| 1042 | // the new default bitrate when computing send bitrate. |
| 1043 | if (send_codec_spec.target_bitrate_bps) { |
| 1044 | info->default_bitrate_bps = std::max( |
| 1045 | info->min_bitrate_bps, |
| 1046 | std::min(info->max_bitrate_bps, *send_codec_spec.target_bitrate_bps)); |
| 1047 | } |
| 1048 | |
| 1049 | audio_codec_spec_.emplace( |
| 1050 | webrtc::AudioCodecSpec{send_codec_spec.format, *info}); |
| 1051 | |
| 1052 | config_.send_codec_spec->target_bitrate_bps = ComputeSendBitrate( |
| 1053 | max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps, |
| 1054 | *audio_codec_spec_); |
| 1055 | |
| 1056 | UpdateAllowedBitrateRange(); |
| 1057 | } |
| 1058 | |
| 1059 | void ReconfigureAudioSendStream() { |
| 1060 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1061 | RTC_DCHECK(stream_); |
| 1062 | stream_->Reconfigure(config_); |
michaelt | 53fe19d | 2016-10-18 09:39:22 -0700 | [diff] [blame] | 1063 | } |
| 1064 | |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1065 | rtc::ThreadChecker worker_thread_checker_; |
solenberg | 347ec5c | 2016-09-23 04:21:47 -0700 | [diff] [blame] | 1066 | rtc::RaceChecker audio_capture_race_checker_; |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1067 | webrtc::Call* call_ = nullptr; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1068 | webrtc::AudioSendStream::Config config_; |
elad.alon | 0fe1216 | 2017-01-31 05:48:37 -0800 | [diff] [blame] | 1069 | const bool send_side_bwe_with_overhead_; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1070 | // The stream is owned by WebRtcAudioSendStream and may be reallocated if |
| 1071 | // configuration changes. |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1072 | webrtc::AudioSendStream* stream_ = nullptr; |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 1073 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1074 | // Raw pointer to AudioSource owned by LocalAudioTrackHandler. |
mallinath@webrtc.org | 67ee6b9 | 2014-02-03 16:57:16 +0000 | [diff] [blame] | 1075 | // PeerConnection will make sure invalidating the pointer before the object |
| 1076 | // goes away. |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1077 | AudioSource* source_ = nullptr; |
| 1078 | bool send_ = false; |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 1079 | bool muted_ = false; |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 1080 | int max_send_bitrate_bps_; |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1081 | webrtc::RtpParameters rtp_parameters_; |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1082 | absl::optional<webrtc::AudioCodecSpec> audio_codec_spec_; |
henrike@webrtc.org | a7b9818 | 2014-02-21 15:51:43 +0000 | [diff] [blame] | 1083 | |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1084 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream); |
| 1085 | }; |
| 1086 | |
| 1087 | class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { |
| 1088 | public: |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 1089 | WebRtcAudioReceiveStream( |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 1090 | uint32_t remote_ssrc, |
| 1091 | uint32_t local_ssrc, |
| 1092 | bool use_transport_cc, |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 1093 | bool use_nack, |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1094 | const std::vector<std::string>& stream_ids, |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 1095 | const std::vector<webrtc::RtpExtension>& extensions, |
| 1096 | webrtc::Call* call, |
| 1097 | webrtc::Transport* rtcp_send_transport, |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 1098 | webrtc::MediaTransportInterface* media_transport, |
kwiberg | 1c07c70 | 2017-03-27 07:15:49 -0700 | [diff] [blame] | 1099 | const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1100 | const std::map<int, webrtc::SdpAudioFormat>& decoder_map, |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1101 | absl::optional<webrtc::AudioCodecPairId> codec_pair_id, |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1102 | size_t jitter_buffer_max_packets, |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 1103 | bool jitter_buffer_fast_accelerate, |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 1104 | int jitter_buffer_min_delay_ms, |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 1105 | bool jitter_buffer_enable_rtx_handling, |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 1106 | rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor, |
| 1107 | const webrtc::CryptoOptions& crypto_options) |
stefan | ba4c0e4 | 2016-02-04 04:12:24 -0800 | [diff] [blame] | 1108 | : call_(call), config_() { |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1109 | RTC_DCHECK(call); |
| 1110 | config_.rtp.remote_ssrc = remote_ssrc; |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1111 | config_.rtp.local_ssrc = local_ssrc; |
| 1112 | config_.rtp.transport_cc = use_transport_cc; |
| 1113 | config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0; |
| 1114 | config_.rtp.extensions = extensions; |
solenberg | 31fec40 | 2016-05-06 02:13:12 -0700 | [diff] [blame] | 1115 | config_.rtcp_send_transport = rtcp_send_transport; |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 1116 | config_.media_transport = media_transport; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1117 | config_.jitter_buffer_max_packets = jitter_buffer_max_packets; |
| 1118 | config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate; |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 1119 | config_.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms; |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 1120 | config_.jitter_buffer_enable_rtx_handling = |
| 1121 | jitter_buffer_enable_rtx_handling; |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1122 | if (!stream_ids.empty()) { |
| 1123 | config_.sync_group = stream_ids[0]; |
Steve Anton | 5a26a3a | 2018-02-28 11:38:47 -0800 | [diff] [blame] | 1124 | } |
ossu | 29b1a8d | 2016-06-13 07:34:51 -0700 | [diff] [blame] | 1125 | config_.decoder_factory = decoder_factory; |
kwiberg | 1c07c70 | 2017-03-27 07:15:49 -0700 | [diff] [blame] | 1126 | config_.decoder_map = decoder_map; |
Karl Wiberg | 0812634 | 2018-03-20 19:18:55 +0100 | [diff] [blame] | 1127 | config_.codec_pair_id = codec_pair_id; |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 1128 | config_.frame_decryptor = frame_decryptor; |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 1129 | config_.crypto_options = crypto_options; |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1130 | RecreateAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1131 | } |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1132 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1133 | ~WebRtcAudioReceiveStream() { |
| 1134 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1135 | call_->DestroyAudioReceiveStream(stream_); |
| 1136 | } |
| 1137 | |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 1138 | void SetFrameDecryptor( |
| 1139 | rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) { |
| 1140 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1141 | config_.frame_decryptor = frame_decryptor; |
| 1142 | RecreateAudioReceiveStream(); |
| 1143 | } |
| 1144 | |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1145 | void SetLocalSsrc(uint32_t local_ssrc) { |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1146 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1147 | config_.rtp.local_ssrc = local_ssrc; |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1148 | ReconfigureAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1149 | } |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 1150 | |
Fredrik Solenberg | 4613bdf | 2018-01-16 13:32:31 +0100 | [diff] [blame] | 1151 | void SetUseTransportCcAndRecreateStream(bool use_transport_cc, |
| 1152 | bool use_nack) { |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1153 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1154 | config_.rtp.transport_cc = use_transport_cc; |
| 1155 | config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0; |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1156 | ReconfigureAudioReceiveStream(); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1157 | } |
| 1158 | |
Fredrik Solenberg | 4613bdf | 2018-01-16 13:32:31 +0100 | [diff] [blame] | 1159 | void SetRtpExtensionsAndRecreateStream( |
| 1160 | const std::vector<webrtc::RtpExtension>& extensions) { |
solenberg | 4a0f7b5 | 2016-06-16 13:07:33 -0700 | [diff] [blame] | 1161 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1162 | config_.rtp.extensions = extensions; |
Fredrik Solenberg | 4613bdf | 2018-01-16 13:32:31 +0100 | [diff] [blame] | 1163 | RecreateAudioReceiveStream(); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1164 | } |
| 1165 | |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1166 | // Set a new payload type -> decoder map. |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1167 | void SetDecoderMap(const std::map<int, webrtc::SdpAudioFormat>& decoder_map) { |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1168 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1169 | config_.decoder_map = decoder_map; |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1170 | ReconfigureAudioReceiveStream(); |
solenberg | 4a0f7b5 | 2016-06-16 13:07:33 -0700 | [diff] [blame] | 1171 | } |
| 1172 | |
Steve Anton | 5a26a3a | 2018-02-28 11:38:47 -0800 | [diff] [blame] | 1173 | void MaybeRecreateAudioReceiveStream( |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1174 | const std::vector<std::string>& stream_ids) { |
solenberg | 4904fb6 | 2017-02-17 12:01:14 -0800 | [diff] [blame] | 1175 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Steve Anton | 5a26a3a | 2018-02-28 11:38:47 -0800 | [diff] [blame] | 1176 | std::string sync_group; |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1177 | if (!stream_ids.empty()) { |
| 1178 | sync_group = stream_ids[0]; |
Steve Anton | 5a26a3a | 2018-02-28 11:38:47 -0800 | [diff] [blame] | 1179 | } |
solenberg | 4904fb6 | 2017-02-17 12:01:14 -0800 | [diff] [blame] | 1180 | if (config_.sync_group != sync_group) { |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1181 | RTC_LOG(LS_INFO) << "Recreating AudioReceiveStream for SSRC=" |
| 1182 | << config_.rtp.remote_ssrc |
| 1183 | << " because of sync group change."; |
solenberg | 4904fb6 | 2017-02-17 12:01:14 -0800 | [diff] [blame] | 1184 | config_.sync_group = sync_group; |
| 1185 | RecreateAudioReceiveStream(); |
| 1186 | } |
| 1187 | } |
| 1188 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1189 | webrtc::AudioReceiveStream::Stats GetStats() const { |
| 1190 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1191 | RTC_DCHECK(stream_); |
| 1192 | return stream_->GetStats(); |
| 1193 | } |
| 1194 | |
kwiberg | 686a8ef | 2016-02-26 03:00:35 -0800 | [diff] [blame] | 1195 | void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 1196 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Oskar Sundbom | 4ccc1c4 | 2018-03-16 13:56:27 +0100 | [diff] [blame] | 1197 | // Need to update the stream's sink first; once raw_audio_sink_ is |
| 1198 | // reassigned, whatever was in there before is destroyed. |
| 1199 | stream_->SetSink(sink.get()); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1200 | raw_audio_sink_ = std::move(sink); |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 1201 | } |
| 1202 | |
solenberg | 217fb66 | 2016-06-17 08:30:54 -0700 | [diff] [blame] | 1203 | void SetOutputVolume(double volume) { |
| 1204 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Oskar Sundbom | c668108 | 2018-02-19 14:09:21 +0100 | [diff] [blame] | 1205 | output_volume_ = volume; |
solenberg | 217fb66 | 2016-06-17 08:30:54 -0700 | [diff] [blame] | 1206 | stream_->SetGain(volume); |
| 1207 | } |
| 1208 | |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1209 | void SetPlayout(bool playout) { |
| 1210 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1211 | RTC_DCHECK(stream_); |
| 1212 | if (playout) { |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1213 | stream_->Start(); |
| 1214 | } else { |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1215 | stream_->Stop(); |
| 1216 | } |
aleloi | 18e0b67 | 2016-10-04 02:45:47 -0700 | [diff] [blame] | 1217 | playout_ = playout; |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1218 | } |
| 1219 | |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 1220 | std::vector<webrtc::RtpSource> GetSources() { |
| 1221 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1222 | RTC_DCHECK(stream_); |
| 1223 | return stream_->GetSources(); |
| 1224 | } |
| 1225 | |
Florent Castelli | abe301f | 2018-06-12 18:33:49 +0200 | [diff] [blame] | 1226 | webrtc::RtpParameters GetRtpParameters() const { |
| 1227 | webrtc::RtpParameters rtp_parameters; |
| 1228 | rtp_parameters.encodings.emplace_back(); |
| 1229 | rtp_parameters.encodings[0].ssrc = config_.rtp.remote_ssrc; |
| 1230 | rtp_parameters.header_extensions = config_.rtp.extensions; |
| 1231 | |
| 1232 | return rtp_parameters; |
| 1233 | } |
| 1234 | |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1235 | private: |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1236 | void RecreateAudioReceiveStream() { |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1237 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1238 | if (stream_) { |
| 1239 | call_->DestroyAudioReceiveStream(stream_); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1240 | } |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1241 | stream_ = call_->CreateAudioReceiveStream(config_); |
| 1242 | RTC_CHECK(stream_); |
Oskar Sundbom | c668108 | 2018-02-19 14:09:21 +0100 | [diff] [blame] | 1243 | stream_->SetGain(output_volume_); |
aleloi | 18e0b67 | 2016-10-04 02:45:47 -0700 | [diff] [blame] | 1244 | SetPlayout(playout_); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1245 | stream_->SetSink(raw_audio_sink_.get()); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1246 | } |
| 1247 | |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1248 | void ReconfigureAudioReceiveStream() { |
| 1249 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1250 | RTC_DCHECK(stream_); |
| 1251 | stream_->Reconfigure(config_); |
| 1252 | } |
| 1253 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1254 | rtc::ThreadChecker worker_thread_checker_; |
| 1255 | webrtc::Call* call_ = nullptr; |
| 1256 | webrtc::AudioReceiveStream::Config config_; |
| 1257 | // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if |
| 1258 | // configuration changes. |
| 1259 | webrtc::AudioReceiveStream* stream_ = nullptr; |
aleloi | 18e0b67 | 2016-10-04 02:45:47 -0700 | [diff] [blame] | 1260 | bool playout_ = false; |
Oskar Sundbom | c668108 | 2018-02-19 14:09:21 +0100 | [diff] [blame] | 1261 | float output_volume_ = 1.0; |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1262 | std::unique_ptr<webrtc::AudioSinkInterface> raw_audio_sink_; |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1263 | |
| 1264 | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1265 | }; |
| 1266 | |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 1267 | WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel( |
| 1268 | WebRtcVoiceEngine* engine, |
| 1269 | const MediaConfig& config, |
| 1270 | const AudioOptions& options, |
| 1271 | const webrtc::CryptoOptions& crypto_options, |
| 1272 | webrtc::Call* call) |
| 1273 | : VoiceMediaChannel(config), |
| 1274 | engine_(engine), |
| 1275 | call_(call), |
Jiawei Ou | 5571812 | 2018-11-09 13:17:39 -0800 | [diff] [blame] | 1276 | audio_config_(config.audio), |
Benjamin Wright | bfb444c | 2018-10-15 10:20:24 -0700 | [diff] [blame] | 1277 | crypto_options_(crypto_options) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1278 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel"; |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1279 | RTC_DCHECK(call); |
solenberg | 0a617e2 | 2015-10-20 15:49:38 -0700 | [diff] [blame] | 1280 | engine->RegisterChannel(this); |
Fredrik Solenberg | b071a19 | 2015-09-17 16:42:56 +0200 | [diff] [blame] | 1281 | SetOptions(options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1282 | } |
| 1283 | |
| 1284 | WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1285 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1286 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel"; |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1287 | // TODO(solenberg): Should be able to delete the streams directly, without |
| 1288 | // going through RemoveNnStream(), once stream objects handle |
| 1289 | // all (de)configuration. |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1290 | while (!send_streams_.empty()) { |
| 1291 | RemoveSendStream(send_streams_.begin()->first); |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 1292 | } |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1293 | while (!recv_streams_.empty()) { |
| 1294 | RemoveRecvStream(recv_streams_.begin()->first); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1295 | } |
solenberg | 0a617e2 | 2015-10-20 15:49:38 -0700 | [diff] [blame] | 1296 | engine()->UnregisterChannel(this); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1297 | } |
| 1298 | |
nisse | 51542be | 2016-02-12 02:27:06 -0800 | [diff] [blame] | 1299 | rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const { |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 1300 | return preferred_dscp_; |
nisse | 51542be | 2016-02-12 02:27:06 -0800 | [diff] [blame] | 1301 | } |
| 1302 | |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1303 | bool WebRtcVoiceMediaChannel::SetSendParameters( |
| 1304 | const AudioSendParameters& params) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1305 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1306 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1307 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: " |
| 1308 | << params.ToString(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1309 | // TODO(pthatcher): Refactor this to be more clean now that we have |
| 1310 | // all the information at once. |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1311 | |
| 1312 | if (!SetSendCodecs(params.codecs)) { |
| 1313 | return false; |
| 1314 | } |
| 1315 | |
solenberg | 7e4e01a | 2015-12-02 08:05:01 -0800 | [diff] [blame] | 1316 | if (!ValidateRtpExtensions(params.extensions)) { |
| 1317 | return false; |
| 1318 | } |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 1319 | |
| 1320 | if (ExtmapAllowMixed() != params.extmap_allow_mixed) { |
| 1321 | SetExtmapAllowMixed(params.extmap_allow_mixed); |
| 1322 | for (auto& it : send_streams_) { |
| 1323 | it.second->SetExtmapAllowMixed(params.extmap_allow_mixed); |
| 1324 | } |
| 1325 | } |
| 1326 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 1327 | std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| 1328 | params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true); |
solenberg | 7e4e01a | 2015-12-02 08:05:01 -0800 | [diff] [blame] | 1329 | if (send_rtp_extensions_ != filtered_extensions) { |
| 1330 | send_rtp_extensions_.swap(filtered_extensions); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1331 | for (auto& it : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1332 | it.second->SetRtpExtensions(send_rtp_extensions_); |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1333 | } |
| 1334 | } |
Steve Anton | bb50ce5 | 2018-03-26 10:24:32 -0700 | [diff] [blame] | 1335 | if (!params.mid.empty()) { |
| 1336 | mid_ = params.mid; |
| 1337 | for (auto& it : send_streams_) { |
| 1338 | it.second->SetMid(params.mid); |
| 1339 | } |
| 1340 | } |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1341 | |
deadbeef | 8034614 | 2016-04-27 14:17:10 -0700 | [diff] [blame] | 1342 | if (!SetMaxSendBitrate(params.max_bandwidth_bps)) { |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1343 | return false; |
| 1344 | } |
| 1345 | return SetOptions(params.options); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1346 | } |
| 1347 | |
| 1348 | bool WebRtcVoiceMediaChannel::SetRecvParameters( |
| 1349 | const AudioRecvParameters& params) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1350 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1351 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1352 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: " |
| 1353 | << params.ToString(); |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1354 | // TODO(pthatcher): Refactor this to be more clean now that we have |
| 1355 | // all the information at once. |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1356 | |
| 1357 | if (!SetRecvCodecs(params.codecs)) { |
| 1358 | return false; |
| 1359 | } |
| 1360 | |
solenberg | 7e4e01a | 2015-12-02 08:05:01 -0800 | [diff] [blame] | 1361 | if (!ValidateRtpExtensions(params.extensions)) { |
| 1362 | return false; |
| 1363 | } |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 1364 | std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| 1365 | params.extensions, webrtc::RtpExtension::IsSupportedForAudio, false); |
solenberg | 7e4e01a | 2015-12-02 08:05:01 -0800 | [diff] [blame] | 1366 | if (recv_rtp_extensions_ != filtered_extensions) { |
| 1367 | recv_rtp_extensions_.swap(filtered_extensions); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1368 | for (auto& it : recv_streams_) { |
Fredrik Solenberg | 4613bdf | 2018-01-16 13:32:31 +0100 | [diff] [blame] | 1369 | it.second->SetRtpExtensionsAndRecreateStream(recv_rtp_extensions_); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1370 | } |
| 1371 | } |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1372 | return true; |
Peter Thatcher | c2ee2c8 | 2015-08-07 16:05:34 -0700 | [diff] [blame] | 1373 | } |
| 1374 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1375 | webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters( |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1376 | uint32_t ssrc) const { |
| 1377 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 1378 | auto it = send_streams_.find(ssrc); |
| 1379 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1380 | RTC_LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream " |
| 1381 | << "with ssrc " << ssrc << " which doesn't exist."; |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1382 | return webrtc::RtpParameters(); |
| 1383 | } |
| 1384 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 16:23:10 -0700 | [diff] [blame] | 1385 | webrtc::RtpParameters rtp_params = it->second->rtp_parameters(); |
| 1386 | // Need to add the common list of codecs to the send stream-specific |
| 1387 | // RTP parameters. |
| 1388 | for (const AudioCodec& codec : send_codecs_) { |
| 1389 | rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| 1390 | } |
| 1391 | return rtp_params; |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1392 | } |
| 1393 | |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 1394 | webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters( |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1395 | uint32_t ssrc, |
| 1396 | const webrtc::RtpParameters& parameters) { |
| 1397 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1398 | auto it = send_streams_.find(ssrc); |
| 1399 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1400 | RTC_LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream " |
| 1401 | << "with ssrc " << ssrc << " which doesn't exist."; |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 1402 | return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1403 | } |
| 1404 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1405 | // TODO(deadbeef): Handle setting parameters with a list of codecs in a |
| 1406 | // different order (which should change the send codec). |
| 1407 | webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); |
| 1408 | if (current_parameters.codecs != parameters.codecs) { |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 1409 | RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs " |
| 1410 | << "is not currently supported."; |
Zach Stein | ba37b4b | 2018-01-23 15:02:36 -0800 | [diff] [blame] | 1411 | return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER); |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1412 | } |
| 1413 | |
Tim Haloun | 648d28a | 2018-10-18 16:52:22 -0700 | [diff] [blame] | 1414 | if (!parameters.encodings.empty()) { |
| 1415 | auto& priority = parameters.encodings[0].network_priority; |
| 1416 | rtc::DiffServCodePoint new_dscp = rtc::DSCP_DEFAULT; |
| 1417 | if (priority == 0.5 * webrtc::kDefaultBitratePriority) { |
| 1418 | new_dscp = rtc::DSCP_CS1; |
| 1419 | } else if (priority == 1.0 * webrtc::kDefaultBitratePriority) { |
| 1420 | new_dscp = rtc::DSCP_DEFAULT; |
| 1421 | } else if (priority == 2.0 * webrtc::kDefaultBitratePriority) { |
| 1422 | new_dscp = rtc::DSCP_EF; |
| 1423 | } else if (priority == 4.0 * webrtc::kDefaultBitratePriority) { |
| 1424 | new_dscp = rtc::DSCP_EF; |
| 1425 | } else { |
| 1426 | RTC_LOG(LS_WARNING) << "Received invalid send network priority: " |
| 1427 | << priority; |
| 1428 | return webrtc::RTCError(webrtc::RTCErrorType::INVALID_RANGE); |
| 1429 | } |
| 1430 | |
| 1431 | if (new_dscp != preferred_dscp_) { |
| 1432 | preferred_dscp_ = new_dscp; |
| 1433 | MediaChannel::UpdateDscp(); |
| 1434 | } |
| 1435 | } |
| 1436 | |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 1437 | // TODO(minyue): The following legacy actions go into |
| 1438 | // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end, |
| 1439 | // though there are two difference: |
| 1440 | // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls |
| 1441 | // |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls |
| 1442 | // |SetSendCodecs|. The outcome should be the same. |
| 1443 | // 2. AudioSendStream can be recreated. |
| 1444 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 16:23:10 -0700 | [diff] [blame] | 1445 | // Codecs are handled at the WebRtcVoiceMediaChannel level. |
| 1446 | webrtc::RtpParameters reduced_params = parameters; |
| 1447 | reduced_params.codecs.clear(); |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 1448 | return it->second->SetRtpParameters(reduced_params); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1449 | } |
| 1450 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1451 | webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters( |
| 1452 | uint32_t ssrc) const { |
| 1453 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1454 | webrtc::RtpParameters rtp_params; |
| 1455 | // SSRC of 0 represents the default receive stream. |
| 1456 | if (ssrc == 0) { |
| 1457 | if (!default_sink_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1458 | RTC_LOG(LS_WARNING) |
| 1459 | << "Attempting to get RTP parameters for the default, " |
| 1460 | "unsignaled audio receive stream, but not yet " |
| 1461 | "configured to receive such a stream."; |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1462 | return rtp_params; |
| 1463 | } |
| 1464 | rtp_params.encodings.emplace_back(); |
| 1465 | } else { |
| 1466 | auto it = recv_streams_.find(ssrc); |
| 1467 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1468 | RTC_LOG(LS_WARNING) |
| 1469 | << "Attempting to get RTP receive parameters for stream " |
| 1470 | << "with ssrc " << ssrc << " which doesn't exist."; |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1471 | return webrtc::RtpParameters(); |
| 1472 | } |
Florent Castelli | abe301f | 2018-06-12 18:33:49 +0200 | [diff] [blame] | 1473 | rtp_params = it->second->GetRtpParameters(); |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1474 | } |
| 1475 | |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1476 | for (const AudioCodec& codec : recv_codecs_) { |
| 1477 | rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| 1478 | } |
| 1479 | return rtp_params; |
| 1480 | } |
| 1481 | |
| 1482 | bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters( |
| 1483 | uint32_t ssrc, |
| 1484 | const webrtc::RtpParameters& parameters) { |
| 1485 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1486 | // SSRC of 0 represents the default receive stream. |
| 1487 | if (ssrc == 0) { |
| 1488 | if (!default_sink_) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1489 | RTC_LOG(LS_WARNING) |
| 1490 | << "Attempting to set RTP parameters for the default, " |
| 1491 | "unsignaled audio receive stream, but not yet " |
| 1492 | "configured to receive such a stream."; |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1493 | return false; |
| 1494 | } |
| 1495 | } else { |
| 1496 | auto it = recv_streams_.find(ssrc); |
| 1497 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1498 | RTC_LOG(LS_WARNING) |
| 1499 | << "Attempting to set RTP receive parameters for stream " |
| 1500 | << "with ssrc " << ssrc << " which doesn't exist."; |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1501 | return false; |
| 1502 | } |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1503 | } |
| 1504 | |
| 1505 | webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); |
| 1506 | if (current_parameters != parameters) { |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 1507 | RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently " |
| 1508 | << "unsupported."; |
Taylor Brandstetter | db0cd9e | 2016-05-16 11:40:30 -0700 | [diff] [blame] | 1509 | return false; |
| 1510 | } |
| 1511 | return true; |
| 1512 | } |
| 1513 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1514 | bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1515 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1516 | RTC_LOG(LS_INFO) << "Setting voice channel options: " << options.ToString(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1517 | |
| 1518 | // We retain all of the existing options, and apply the given ones |
| 1519 | // on top. This means there is no way to "clear" options such that |
| 1520 | // they go back to the engine default. |
| 1521 | options_.SetAll(options); |
solenberg | 246b817 | 2015-12-08 09:50:23 -0800 | [diff] [blame] | 1522 | if (!engine()->ApplyOptions(options_)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1523 | RTC_LOG(LS_WARNING) |
| 1524 | << "Failed to apply engine options during channel SetOptions."; |
solenberg | 246b817 | 2015-12-08 09:50:23 -0800 | [diff] [blame] | 1525 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1526 | } |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 1527 | |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1528 | absl::optional<std::string> audio_network_adaptor_config = |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 1529 | GetAudioNetworkAdaptorConfig(options_); |
| 1530 | for (auto& it : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1531 | it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config); |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 1532 | } |
| 1533 | |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1534 | RTC_LOG(LS_INFO) << "Set voice channel options. Current options: " |
| 1535 | << options_.ToString(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1536 | return true; |
| 1537 | } |
| 1538 | |
| 1539 | bool WebRtcVoiceMediaChannel::SetRecvCodecs( |
| 1540 | const std::vector<AudioCodec>& codecs) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1541 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 8fb30c3 | 2015-10-13 03:06:58 -0700 | [diff] [blame] | 1542 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1543 | // Set the payload types to be used for incoming media. |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1544 | RTC_LOG(LS_INFO) << "Setting receive voice codecs."; |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1545 | |
| 1546 | if (!VerifyUniquePayloadTypes(codecs)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1547 | RTC_LOG(LS_ERROR) << "Codec payload types overlap."; |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1548 | return false; |
| 1549 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1550 | |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1551 | // Create a payload type -> SdpAudioFormat map with all the decoders. Fail |
| 1552 | // unless the factory claims to support all decoders. |
| 1553 | std::map<int, webrtc::SdpAudioFormat> decoder_map; |
| 1554 | for (const AudioCodec& codec : codecs) { |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1555 | // Log a warning if a codec's payload type is changing. This used to be |
| 1556 | // treated as an error. It's abnormal, but not really illegal. |
| 1557 | AudioCodec old_codec; |
| 1558 | if (FindCodec(recv_codecs_, codec, &old_codec) && |
| 1559 | old_codec.id != codec.id) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1560 | RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type (" |
| 1561 | << codec.id << ", was already mapped to " |
| 1562 | << old_codec.id << ")"; |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1563 | } |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1564 | auto format = AudioCodecToSdpAudioFormat(codec); |
| 1565 | if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") && |
| 1566 | !engine()->decoder_factory_->IsSupportedDecoder(format)) { |
Jonas Olsson | abbe841 | 2018-04-03 13:40:05 +0200 | [diff] [blame] | 1567 | RTC_LOG(LS_ERROR) << "Unsupported codec: " << rtc::ToString(format); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1568 | return false; |
| 1569 | } |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1570 | // We allow adding new codecs but don't allow changing the payload type of |
| 1571 | // codecs that are already configured since we might already be receiving |
| 1572 | // packets with that payload type. See RFC3264, Section 8.3.2. |
| 1573 | // TODO(deadbeef): Also need to check for clashes with previously mapped |
| 1574 | // payload types, and not just currently mapped ones. For example, this |
| 1575 | // should be illegal: |
| 1576 | // 1. {100: opus/48000/2, 101: ISAC/16000} |
| 1577 | // 2. {100: opus/48000/2} |
| 1578 | // 3. {100: opus/48000/2, 101: ISAC/32000} |
| 1579 | // Though this check really should happen at a higher level, since this |
| 1580 | // conflict could happen between audio and video codecs. |
| 1581 | auto existing = decoder_map_.find(codec.id); |
| 1582 | if (existing != decoder_map_.end() && !existing->second.Matches(format)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1583 | RTC_LOG(LS_ERROR) << "Attempting to use payload type " << codec.id |
| 1584 | << " for " << codec.name |
| 1585 | << ", but it is already used for " |
| 1586 | << existing->second.name; |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1587 | return false; |
| 1588 | } |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1589 | decoder_map.insert({codec.id, std::move(format)}); |
| 1590 | } |
| 1591 | |
deadbeef | cb38367 | 2017-04-26 16:28:42 -0700 | [diff] [blame] | 1592 | if (decoder_map == decoder_map_) { |
| 1593 | // There's nothing new to configure. |
| 1594 | return true; |
| 1595 | } |
| 1596 | |
kwiberg | 37b8b11 | 2016-11-03 02:46:53 -0700 | [diff] [blame] | 1597 | if (playout_) { |
| 1598 | // Receive codecs can not be changed while playing. So we temporarily |
| 1599 | // pause playout. |
| 1600 | ChangePlayout(false); |
| 1601 | } |
| 1602 | |
kwiberg | 1c07c70 | 2017-03-27 07:15:49 -0700 | [diff] [blame] | 1603 | decoder_map_ = std::move(decoder_map); |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1604 | for (auto& kv : recv_streams_) { |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1605 | kv.second->SetDecoderMap(decoder_map_); |
solenberg | 26c8c91 | 2015-11-27 04:00:25 -0800 | [diff] [blame] | 1606 | } |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1607 | recv_codecs_ = codecs; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1608 | |
kwiberg | 37b8b11 | 2016-11-03 02:46:53 -0700 | [diff] [blame] | 1609 | if (desired_playout_ && !playout_) { |
| 1610 | ChangePlayout(desired_playout_); |
| 1611 | } |
kwiberg | d32bf75 | 2017-01-19 07:03:59 -0800 | [diff] [blame] | 1612 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1613 | } |
| 1614 | |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1615 | // Utility function called from SetSendParameters() to extract current send |
| 1616 | // codec settings from the given list of codecs (originally from SDP). Both send |
| 1617 | // and receive streams may be reconfigured based on the new settings. |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 1618 | bool WebRtcVoiceMediaChannel::SetSendCodecs( |
| 1619 | const std::vector<AudioCodec>& codecs) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1620 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1621 | dtmf_payload_type_ = absl::nullopt; |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1622 | dtmf_payload_freq_ = -1; |
| 1623 | |
| 1624 | // Validate supplied codecs list. |
| 1625 | for (const AudioCodec& codec : codecs) { |
| 1626 | // TODO(solenberg): Validate more aspects of input - that payload types |
| 1627 | // don't overlap, remove redundant/unsupported codecs etc - |
| 1628 | // the same way it is done for RtpHeaderExtensions. |
| 1629 | if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1630 | RTC_LOG(LS_WARNING) << "Codec payload type out of range: " |
| 1631 | << ToString(codec); |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1632 | return false; |
| 1633 | } |
| 1634 | } |
| 1635 | |
| 1636 | // Find PT of telephone-event codec with lowest clockrate, as a fallback, in |
| 1637 | // case we don't have a DTMF codec with a rate matching the send codec's, or |
| 1638 | // if this function returns early. |
| 1639 | std::vector<AudioCodec> dtmf_codecs; |
Fredrik Solenberg | af9fb21 | 2015-08-26 10:45:53 +0200 | [diff] [blame] | 1640 | for (const AudioCodec& codec : codecs) { |
Fredrik Solenberg | af9fb21 | 2015-08-26 10:45:53 +0200 | [diff] [blame] | 1641 | if (IsCodec(codec, kDtmfCodecName)) { |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1642 | dtmf_codecs.push_back(codec); |
| 1643 | if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 1644 | dtmf_payload_type_ = codec.id; |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1645 | dtmf_payload_freq_ = codec.clockrate; |
solenberg | 31642aa | 2016-03-14 08:00:37 -0700 | [diff] [blame] | 1646 | } |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 1647 | } |
| 1648 | } |
| 1649 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1650 | // Scan through the list to figure out the codec to use for sending. |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1651 | absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec> |
| 1652 | send_codec_spec; |
Sebastian Jansson | fc8d26b | 2018-02-21 09:52:06 +0100 | [diff] [blame] | 1653 | webrtc::BitrateConstraints bitrate_config; |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1654 | absl::optional<webrtc::AudioCodecInfo> voice_codec_info; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1655 | for (const AudioCodec& voice_codec : codecs) { |
| 1656 | if (!(IsCodec(voice_codec, kCnCodecName) || |
| 1657 | IsCodec(voice_codec, kDtmfCodecName) || |
| 1658 | IsCodec(voice_codec, kRedCodecName))) { |
| 1659 | webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate, |
| 1660 | voice_codec.channels, voice_codec.params); |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1661 | |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1662 | voice_codec_info = engine()->encoder_factory_->QueryAudioEncoder(format); |
| 1663 | if (!voice_codec_info) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1664 | RTC_LOG(LS_WARNING) << "Unknown codec " << ToString(voice_codec); |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1665 | continue; |
| 1666 | } |
| 1667 | |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 1668 | send_codec_spec = webrtc::AudioSendStream::Config::SendCodecSpec( |
| 1669 | voice_codec.id, format); |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1670 | if (voice_codec.bitrate > 0) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 1671 | send_codec_spec->target_bitrate_bps = voice_codec.bitrate; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1672 | } |
| 1673 | send_codec_spec->transport_cc_enabled = HasTransportCc(voice_codec); |
| 1674 | send_codec_spec->nack_enabled = HasNack(voice_codec); |
| 1675 | bitrate_config = GetBitrateConfigForCodec(voice_codec); |
| 1676 | break; |
| 1677 | } |
| 1678 | } |
| 1679 | |
| 1680 | if (!send_codec_spec) { |
| 1681 | return false; |
| 1682 | } |
| 1683 | |
| 1684 | RTC_DCHECK(voice_codec_info); |
| 1685 | if (voice_codec_info->allow_comfort_noise) { |
| 1686 | // Loop through the codecs list again to find the CN codec. |
| 1687 | // TODO(solenberg): Break out into a separate function? |
| 1688 | for (const AudioCodec& cn_codec : codecs) { |
ossu | 0c4b849 | 2017-03-02 11:03:25 -0800 | [diff] [blame] | 1689 | if (IsCodec(cn_codec, kCnCodecName) && |
Karl Wiberg | 20a49f3 | 2018-10-08 12:41:33 +0200 | [diff] [blame] | 1690 | cn_codec.clockrate == send_codec_spec->format.clockrate_hz && |
| 1691 | cn_codec.channels == voice_codec_info->num_channels) { |
| 1692 | if (cn_codec.channels != 1) { |
| 1693 | RTC_LOG(LS_WARNING) |
| 1694 | << "CN #channels " << cn_codec.channels << " not supported."; |
| 1695 | } else if (cn_codec.clockrate != 8000 && cn_codec.clockrate != 16000 && |
| 1696 | cn_codec.clockrate != 32000) { |
| 1697 | RTC_LOG(LS_WARNING) |
| 1698 | << "CN frequency " << cn_codec.clockrate << " not supported."; |
| 1699 | } else { |
| 1700 | send_codec_spec->cng_payload_type = cn_codec.id; |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1701 | } |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1702 | break; |
| 1703 | } |
| 1704 | } |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1705 | |
| 1706 | // Find the telephone-event PT exactly matching the preferred send codec. |
| 1707 | for (const AudioCodec& dtmf_codec : dtmf_codecs) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1708 | if (dtmf_codec.clockrate == send_codec_spec->format.clockrate_hz) { |
Oskar Sundbom | 7880758 | 2017-11-16 11:09:55 +0100 | [diff] [blame] | 1709 | dtmf_payload_type_ = dtmf_codec.id; |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 1710 | dtmf_payload_freq_ = dtmf_codec.clockrate; |
| 1711 | break; |
| 1712 | } |
| 1713 | } |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1714 | } |
| 1715 | |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 1716 | if (send_codec_spec_ != send_codec_spec) { |
| 1717 | send_codec_spec_ = std::move(send_codec_spec); |
stefan | 13f1a0a | 2016-11-30 07:22:58 -0800 | [diff] [blame] | 1718 | // Apply new settings to all streams. |
solenberg | 971cab0 | 2016-06-14 10:02:41 -0700 | [diff] [blame] | 1719 | for (const auto& kv : send_streams_) { |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1720 | kv.second->SetSendCodecSpec(*send_codec_spec_); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 1721 | } |
stefan | 13f1a0a | 2016-11-30 07:22:58 -0800 | [diff] [blame] | 1722 | } else { |
| 1723 | // If the codec isn't changing, set the start bitrate to -1 which means |
| 1724 | // "unchanged" so that BWE isn't affected. |
stefan | 1ccf73f | 2017-03-27 03:51:18 -0700 | [diff] [blame] | 1725 | bitrate_config.start_bitrate_bps = -1; |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 1726 | } |
Sebastian Jansson | 8f83b42 | 2018-02-21 13:07:13 +0100 | [diff] [blame] | 1727 | call_->GetTransportControllerSend()->SetSdpBitrateParameters(bitrate_config); |
wu@webrtc.org | cadf904 | 2013-08-30 21:24:16 +0000 | [diff] [blame] | 1728 | |
solenberg | 8189b02 | 2016-06-14 12:13:00 -0700 | [diff] [blame] | 1729 | // Check if the transport cc feedback or NACK status has changed on the |
| 1730 | // preferred send codec, and in that case reconfigure all receive streams. |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1731 | if (recv_transport_cc_enabled_ != send_codec_spec_->transport_cc_enabled || |
| 1732 | recv_nack_enabled_ != send_codec_spec_->nack_enabled) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1733 | RTC_LOG(LS_INFO) << "Recreate all the receive streams because the send " |
| 1734 | "codec has changed."; |
ossu | 20a4b3f | 2017-04-27 02:08:52 -0700 | [diff] [blame] | 1735 | recv_transport_cc_enabled_ = send_codec_spec_->transport_cc_enabled; |
| 1736 | recv_nack_enabled_ = send_codec_spec_->nack_enabled; |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1737 | for (auto& kv : recv_streams_) { |
Fredrik Solenberg | 4613bdf | 2018-01-16 13:32:31 +0100 | [diff] [blame] | 1738 | kv.second->SetUseTransportCcAndRecreateStream(recv_transport_cc_enabled_, |
| 1739 | recv_nack_enabled_); |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1740 | } |
| 1741 | } |
| 1742 | |
Taylor Brandstetter | 0cd086b | 2016-04-20 16:23:10 -0700 | [diff] [blame] | 1743 | send_codecs_ = codecs; |
solenberg | 72e29d2 | 2016-03-08 06:35:16 -0800 | [diff] [blame] | 1744 | return true; |
| 1745 | } |
| 1746 | |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1747 | void WebRtcVoiceMediaChannel::SetPlayout(bool playout) { |
kwiberg | 37b8b11 | 2016-11-03 02:46:53 -0700 | [diff] [blame] | 1748 | desired_playout_ = playout; |
| 1749 | return ChangePlayout(desired_playout_); |
| 1750 | } |
| 1751 | |
| 1752 | void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) { |
| 1753 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1754 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1755 | if (playout_ == playout) { |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1756 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1757 | } |
| 1758 | |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1759 | for (const auto& kv : recv_streams_) { |
| 1760 | kv.second->SetPlayout(playout); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1761 | } |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1762 | playout_ = playout; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1763 | } |
| 1764 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1765 | void WebRtcVoiceMediaChannel::SetSend(bool send) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1766 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend"); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1767 | if (send_ == send) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1768 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1769 | } |
| 1770 | |
solenberg | d53a3f9 | 2016-04-14 13:56:37 -0700 | [diff] [blame] | 1771 | // Apply channel specific options, and initialize the ADM for recording (this |
| 1772 | // may take time on some platforms, e.g. Android). |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1773 | if (send) { |
solenberg | 63b3454 | 2015-09-29 06:06:31 -0700 | [diff] [blame] | 1774 | engine()->ApplyOptions(options_); |
solenberg | d53a3f9 | 2016-04-14 13:56:37 -0700 | [diff] [blame] | 1775 | |
| 1776 | // InitRecording() may return an error if the ADM is already recording. |
| 1777 | if (!engine()->adm()->RecordingIsInitialized() && |
| 1778 | !engine()->adm()->Recording()) { |
| 1779 | if (engine()->adm()->InitRecording() != 0) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1780 | RTC_LOG(LS_WARNING) << "Failed to initialize recording"; |
solenberg | d53a3f9 | 2016-04-14 13:56:37 -0700 | [diff] [blame] | 1781 | } |
| 1782 | } |
solenberg | 63b3454 | 2015-09-29 06:06:31 -0700 | [diff] [blame] | 1783 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1784 | |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1785 | // Change the settings on each send channel. |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1786 | for (auto& kv : send_streams_) { |
| 1787 | kv.second->SetSend(send); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1788 | } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1789 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1790 | send_ = send; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1791 | } |
| 1792 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1793 | bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, |
| 1794 | bool enable, |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1795 | const AudioOptions* options, |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1796 | AudioSource* source) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1797 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1798 | // TODO(solenberg): The state change should be fully rolled back if any one of |
| 1799 | // these calls fail. |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1800 | if (!SetLocalSource(ssrc, source)) { |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1801 | return false; |
| 1802 | } |
solenberg | dfc8f4f | 2015-10-01 02:31:10 -0700 | [diff] [blame] | 1803 | if (!MuteStream(ssrc, !enable)) { |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1804 | return false; |
| 1805 | } |
solenberg | dfc8f4f | 2015-10-01 02:31:10 -0700 | [diff] [blame] | 1806 | if (enable && options) { |
solenberg | 1dd98f3 | 2015-09-10 01:57:14 -0700 | [diff] [blame] | 1807 | return SetOptions(*options); |
| 1808 | } |
| 1809 | return true; |
| 1810 | } |
| 1811 | |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1812 | bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1813 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1814 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1815 | RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); |
solenberg | 0a617e2 | 2015-10-20 15:49:38 -0700 | [diff] [blame] | 1816 | |
| 1817 | uint32_t ssrc = sp.first_ssrc(); |
| 1818 | RTC_DCHECK(0 != ssrc); |
| 1819 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1820 | if (send_streams_.find(ssrc) != send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1821 | RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1822 | return false; |
| 1823 | } |
| 1824 | |
Danil Chapovalov | 00c7183 | 2018-06-15 15:58:38 +0200 | [diff] [blame] | 1825 | absl::optional<std::string> audio_network_adaptor_config = |
minyue | 6b825df | 2016-10-31 04:08:32 -0700 | [diff] [blame] | 1826 | GetAudioNetworkAdaptorConfig(options_); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1827 | WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 1828 | ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(), |
Jiawei Ou | 5571812 | 2018-11-09 13:17:39 -0800 | [diff] [blame] | 1829 | send_rtp_extensions_, max_send_bitrate_bps_, |
| 1830 | audio_config_.rtcp_report_interval_ms, audio_network_adaptor_config, |
Johannes Kron | 9190b82 | 2018-10-29 11:22:05 +0100 | [diff] [blame] | 1831 | call_, this, media_transport(), engine()->encoder_factory_, |
| 1832 | codec_pair_id_, nullptr, crypto_options_); |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 1833 | send_streams_.insert(std::make_pair(ssrc, stream)); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1834 | |
solenberg | 4a0f7b5 | 2016-06-16 13:07:33 -0700 | [diff] [blame] | 1835 | // At this point the stream's local SSRC has been updated. If it is the first |
| 1836 | // send stream, make sure that all the receive streams are updated with the |
| 1837 | // same SSRC in order to send receiver reports. |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1838 | if (send_streams_.size() == 1) { |
solenberg | 0a617e2 | 2015-10-20 15:49:38 -0700 | [diff] [blame] | 1839 | receiver_reports_ssrc_ = ssrc; |
solenberg | 4a0f7b5 | 2016-06-16 13:07:33 -0700 | [diff] [blame] | 1840 | for (const auto& kv : recv_streams_) { |
| 1841 | // TODO(solenberg): Allow applications to set the RTCP SSRC of receive |
Fredrik Solenberg | 3b903d0 | 2018-01-10 15:17:10 +0100 | [diff] [blame] | 1842 | // streams instead, so we can avoid reconfiguring the streams here. |
| 1843 | kv.second->SetLocalSsrc(ssrc); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1844 | } |
| 1845 | } |
| 1846 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1847 | send_streams_[ssrc]->SetSend(send_); |
| 1848 | return true; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1849 | } |
| 1850 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1851 | bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1852 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1853 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1854 | RTC_LOG(LS_INFO) << "RemoveSendStream: " << ssrc; |
solenberg | 3a94154 | 2015-11-16 07:34:50 -0800 | [diff] [blame] | 1855 | |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1856 | auto it = send_streams_.find(ssrc); |
| 1857 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1858 | RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc |
| 1859 | << " which doesn't exist."; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1860 | return false; |
| 1861 | } |
| 1862 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1863 | it->second->SetSend(false); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1864 | |
solenberg | 7602aab | 2016-11-14 11:30:07 -0800 | [diff] [blame] | 1865 | // TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find |
| 1866 | // the first active send stream and use that instead, reassociating receive |
| 1867 | // streams. |
| 1868 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1869 | delete it->second; |
| 1870 | send_streams_.erase(it); |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1871 | if (send_streams_.empty()) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1872 | SetSend(false); |
solenberg | 0a617e2 | 2015-10-20 15:49:38 -0700 | [diff] [blame] | 1873 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1874 | return true; |
| 1875 | } |
| 1876 | |
| 1877 | bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1878 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1879 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1880 | RTC_LOG(LS_INFO) << "AddRecvStream: " << sp.ToString(); |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 1881 | |
Seth Hampson | 5897a6e | 2018-04-03 11:16:33 -0700 | [diff] [blame] | 1882 | if (!sp.has_ssrcs()) { |
| 1883 | // This is a StreamParam with unsignaled SSRCs. Store it, so it can be used |
| 1884 | // later when we know the SSRCs on the first packet arrival. |
| 1885 | unsignaled_stream_params_ = sp; |
| 1886 | return true; |
| 1887 | } |
| 1888 | |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1889 | if (!ValidateStreamParams(sp)) { |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 1890 | return false; |
| 1891 | } |
| 1892 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1893 | const uint32_t ssrc = sp.first_ssrc(); |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1894 | if (ssrc == 0) { |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 1895 | RTC_DLOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported."; |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1896 | return false; |
| 1897 | } |
| 1898 | |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1899 | // If this stream was previously received unsignaled, we promote it, possibly |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1900 | // recreating the AudioReceiveStream, if stream ids have changed. |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1901 | if (MaybeDeregisterUnsignaledRecvStream(ssrc)) { |
Seth Hampson | 845e878 | 2018-03-02 11:34:10 -0800 | [diff] [blame] | 1902 | recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.stream_ids()); |
solenberg | 4904fb6 | 2017-02-17 12:01:14 -0800 | [diff] [blame] | 1903 | return true; |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1904 | } |
solenberg | 0b67546 | 2015-10-09 01:37:09 -0700 | [diff] [blame] | 1905 | |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1906 | if (recv_streams_.find(ssrc) != recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1907 | RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1908 | return false; |
| 1909 | } |
Fredrik Solenberg | 4b60c73 | 2015-05-07 14:07:48 +0200 | [diff] [blame] | 1910 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1911 | // Create a new channel for receiving audio data. |
stefan | ba4c0e4 | 2016-02-04 04:12:24 -0800 | [diff] [blame] | 1912 | recv_streams_.insert(std::make_pair( |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 1913 | ssrc, |
| 1914 | new WebRtcAudioReceiveStream( |
| 1915 | ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_, |
| 1916 | recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_, call_, |
| 1917 | this, media_transport(), engine()->decoder_factory_, decoder_map_, |
| 1918 | codec_pair_id_, engine()->audio_jitter_buffer_max_packets_, |
| 1919 | engine()->audio_jitter_buffer_fast_accelerate_, |
Jakob Ivarsson | 10403ae | 2018-11-27 15:45:20 +0100 | [diff] [blame] | 1920 | engine()->audio_jitter_buffer_min_delay_ms_, |
Jakob Ivarsson | 53eae87 | 2019-01-10 15:58:36 +0100 | [diff] [blame] | 1921 | engine()->audio_jitter_buffer_enable_rtx_handling_, |
Niels Möller | 7d76a31 | 2018-10-26 12:57:07 +0200 | [diff] [blame] | 1922 | unsignaled_frame_decryptor_, crypto_options_))); |
aleloi | 84ef615 | 2016-08-04 05:28:21 -0700 | [diff] [blame] | 1923 | recv_streams_[ssrc]->SetPlayout(playout_); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1924 | |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1925 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1926 | } |
| 1927 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 1928 | bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 1929 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1930 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1931 | RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 1932 | |
Seth Hampson | 5897a6e | 2018-04-03 11:16:33 -0700 | [diff] [blame] | 1933 | if (ssrc == 0) { |
| 1934 | // This indicates that we need to remove the unsignaled stream parameters |
| 1935 | // that are cached. |
| 1936 | unsignaled_stream_params_ = StreamParams(); |
| 1937 | return true; |
| 1938 | } |
| 1939 | |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1940 | const auto it = recv_streams_.find(ssrc); |
| 1941 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1942 | RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc |
| 1943 | << " which doesn't exist."; |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1944 | return false; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1945 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1946 | |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1947 | MaybeDeregisterUnsignaledRecvStream(ssrc); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1948 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 1949 | it->second->SetRawAudioSink(nullptr); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 1950 | delete it->second; |
| 1951 | recv_streams_.erase(it); |
Fredrik Solenberg | 8f5787a | 2018-01-11 13:52:30 +0100 | [diff] [blame] | 1952 | return true; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1953 | } |
| 1954 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1955 | bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc, |
| 1956 | AudioSource* source) { |
solenberg | c96df77 | 2015-10-21 13:01:53 -0700 | [diff] [blame] | 1957 | auto it = send_streams_.find(ssrc); |
| 1958 | if (it == send_streams_.end()) { |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1959 | if (source) { |
| 1960 | // Return an error if trying to set a valid source with an invalid ssrc. |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1961 | RTC_LOG(LS_ERROR) << "SetLocalSource failed with ssrc " << ssrc; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 1962 | return false; |
| 1963 | } |
| 1964 | |
| 1965 | // The channel likely has gone away, do nothing. |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1966 | return true; |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1967 | } |
| 1968 | |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1969 | if (source) { |
| 1970 | it->second->SetSource(source); |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1971 | } else { |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 1972 | it->second->ClearSource(); |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1973 | } |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 +0000 | [diff] [blame] | 1974 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1975 | return true; |
| 1976 | } |
| 1977 | |
solenberg | 4bac9c5 | 2015-10-09 02:32:53 -0700 | [diff] [blame] | 1978 | bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 1979 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1980 | std::vector<uint32_t> ssrcs(1, ssrc); |
deadbeef | 3bc1510 | 2017-04-20 19:25:07 -0700 | [diff] [blame] | 1981 | // SSRC of 0 represents the default receive stream. |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1982 | if (ssrc == 0) { |
| 1983 | default_recv_volume_ = volume; |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1984 | ssrcs = unsignaled_recv_ssrcs_; |
| 1985 | } |
| 1986 | for (uint32_t ssrc : ssrcs) { |
| 1987 | const auto it = recv_streams_.find(ssrc); |
| 1988 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1989 | RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc; |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1990 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1991 | } |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 1992 | it->second->SetOutputVolume(volume); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 1993 | RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume |
| 1994 | << " for recv stream with ssrc " << ssrc; |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 1995 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1996 | return true; |
| 1997 | } |
| 1998 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1999 | bool WebRtcVoiceMediaChannel::CanInsertDtmf() { |
Harald Alvestrand | a1f6661 | 2018-02-21 11:24:23 +0100 | [diff] [blame] | 2000 | return dtmf_payload_type_.has_value() && send_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2001 | } |
| 2002 | |
Benjamin Wright | 84583f6 | 2018-10-04 14:22:34 -0700 | [diff] [blame] | 2003 | void WebRtcVoiceMediaChannel::SetFrameDecryptor( |
| 2004 | uint32_t ssrc, |
| 2005 | rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) { |
| 2006 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2007 | auto matching_stream = recv_streams_.find(ssrc); |
| 2008 | if (matching_stream != recv_streams_.end()) { |
| 2009 | matching_stream->second->SetFrameDecryptor(frame_decryptor); |
| 2010 | } |
| 2011 | // Handle unsignaled frame decryptors. |
| 2012 | if (ssrc == 0) { |
| 2013 | unsignaled_frame_decryptor_ = frame_decryptor; |
| 2014 | } |
| 2015 | } |
| 2016 | |
| 2017 | void WebRtcVoiceMediaChannel::SetFrameEncryptor( |
| 2018 | uint32_t ssrc, |
| 2019 | rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) { |
| 2020 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2021 | auto matching_stream = send_streams_.find(ssrc); |
| 2022 | if (matching_stream != send_streams_.end()) { |
| 2023 | matching_stream->second->SetFrameEncryptor(frame_encryptor); |
| 2024 | } |
| 2025 | } |
| 2026 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2027 | bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, |
| 2028 | int event, |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 2029 | int duration) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 2030 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2031 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf"; |
Harald Alvestrand | a1f6661 | 2018-02-21 11:24:23 +0100 | [diff] [blame] | 2032 | if (!CanInsertDtmf()) { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2033 | return false; |
| 2034 | } |
| 2035 | |
Fredrik Solenberg | b572768 | 2015-12-04 15:22:19 +0100 | [diff] [blame] | 2036 | // Figure out which WebRtcAudioSendStream to send the event on. |
| 2037 | auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin(); |
| 2038 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2039 | RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 2040 | return false; |
| 2041 | } |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2042 | if (event < kMinTelephoneEventCode || event > kMaxTelephoneEventCode) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2043 | RTC_LOG(LS_WARNING) << "DTMF event code " << event << " out of range."; |
solenberg | 1d63dd0 | 2015-12-02 12:35:09 -0800 | [diff] [blame] | 2044 | return false; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2045 | } |
solenberg | ffbbcac | 2016-11-17 05:25:37 -0800 | [diff] [blame] | 2046 | RTC_DCHECK_NE(-1, dtmf_payload_freq_); |
| 2047 | return it->second->SendTelephoneEvent(*dtmf_payload_type_, dtmf_payload_freq_, |
| 2048 | event, duration); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2049 | } |
| 2050 | |
Niels Möller | e693381 | 2018-11-05 13:01:41 +0100 | [diff] [blame] | 2051 | void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| 2052 | int64_t packet_time_us) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 2053 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 4b60c73 | 2015-05-07 14:07:48 +0200 | [diff] [blame] | 2054 | |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2055 | webrtc::PacketReceiver::DeliveryStatus delivery_result = |
Danil Chapovalov | 292a73e | 2017-12-07 17:00:40 +0100 | [diff] [blame] | 2056 | call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet, |
Niels Möller | e693381 | 2018-11-05 13:01:41 +0100 | [diff] [blame] | 2057 | packet_time_us); |
| 2058 | |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2059 | if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) { |
| 2060 | return; |
| 2061 | } |
| 2062 | |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2063 | // Create an unsignaled receive stream for this previously not received ssrc. |
| 2064 | // If there already is N unsignaled receive streams, delete the oldest. |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2065 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208 |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 2066 | uint32_t ssrc = 0; |
jbauch | eec21bd | 2016-03-20 06:15:43 -0700 | [diff] [blame] | 2067 | if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 2068 | return; |
| 2069 | } |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2070 | RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(), |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2071 | unsignaled_recv_ssrcs_.end(), |
| 2072 | ssrc) == unsignaled_recv_ssrcs_.end()); |
solenberg | 1ac5614 | 2015-10-13 03:58:19 -0700 | [diff] [blame] | 2073 | |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2074 | // Add new stream. |
Seth Hampson | 5897a6e | 2018-04-03 11:16:33 -0700 | [diff] [blame] | 2075 | StreamParams sp = unsignaled_stream_params_; |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2076 | sp.ssrcs.push_back(ssrc); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2077 | RTC_LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc; |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2078 | if (!AddRecvStream(sp)) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2079 | RTC_LOG(LS_WARNING) << "Could not create unsignaled receive stream."; |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2080 | return; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2081 | } |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2082 | unsignaled_recv_ssrcs_.push_back(ssrc); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2083 | RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.NumOfUnsignaledStreams", |
| 2084 | unsignaled_recv_ssrcs_.size(), 1, 100, 101); |
solenberg | f748ca4 | 2017-02-06 13:03:19 -0800 | [diff] [blame] | 2085 | |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2086 | // Remove oldest unsignaled stream, if we have too many. |
| 2087 | if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) { |
| 2088 | uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front(); |
Jonas Olsson | 8544799 | 2018-11-13 14:43:09 +0100 | [diff] [blame] | 2089 | RTC_DLOG(LS_INFO) << "Removing unsignaled receive stream with SSRC=" |
| 2090 | << remove_ssrc; |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2091 | RemoveRecvStream(remove_ssrc); |
| 2092 | } |
| 2093 | RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size()); |
| 2094 | |
| 2095 | SetOutputVolume(ssrc, default_recv_volume_); |
| 2096 | |
| 2097 | // The default sink can only be attached to one stream at a time, so we hook |
| 2098 | // it up to the *latest* unsignaled stream we've seen, in order to support the |
| 2099 | // case where the SSRC of one unsignaled stream changes. |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2100 | if (default_sink_) { |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2101 | for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) { |
| 2102 | auto it = recv_streams_.find(drop_ssrc); |
| 2103 | it->second->SetRawAudioSink(nullptr); |
| 2104 | } |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2105 | std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( |
| 2106 | new ProxySink(default_sink_.get())); |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2107 | SetRawAudioSink(ssrc, std::move(proxy_sink)); |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2108 | } |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2109 | |
Niels Möller | 15ca5a9 | 2018-11-01 14:32:47 +0100 | [diff] [blame] | 2110 | delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
Niels Möller | e693381 | 2018-11-05 13:01:41 +0100 | [diff] [blame] | 2111 | *packet, packet_time_us); |
mflodman | 3d7db26 | 2016-04-29 00:57:13 -0700 | [diff] [blame] | 2112 | RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2113 | } |
| 2114 | |
Niels Möller | e693381 | 2018-11-05 13:01:41 +0100 | [diff] [blame] | 2115 | void WebRtcVoiceMediaChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
| 2116 | int64_t packet_time_us) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 2117 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Fredrik Solenberg | 4b60c73 | 2015-05-07 14:07:48 +0200 | [diff] [blame] | 2118 | |
Fredrik Solenberg | 709ed67 | 2015-09-15 12:26:33 +0200 | [diff] [blame] | 2119 | // Forward packet to Call as well. |
Danil Chapovalov | 292a73e | 2017-12-07 17:00:40 +0100 | [diff] [blame] | 2120 | call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet, |
Niels Möller | e693381 | 2018-11-05 13:01:41 +0100 | [diff] [blame] | 2121 | packet_time_us); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2122 | } |
| 2123 | |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 2124 | void WebRtcVoiceMediaChannel::OnNetworkRouteChanged( |
| 2125 | const std::string& transport_name, |
Honghai Zhang | 0e533ef | 2016-04-19 15:41:36 -0700 | [diff] [blame] | 2126 | const rtc::NetworkRoute& network_route) { |
Zhi Huang | 5f5918f | 2017-11-12 17:26:23 -0800 | [diff] [blame] | 2127 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Sebastian Jansson | 8f83b42 | 2018-02-21 13:07:13 +0100 | [diff] [blame] | 2128 | call_->GetTransportControllerSend()->OnNetworkRouteChanged(transport_name, |
| 2129 | network_route); |
Stefan Holmer | 64be7fa | 2018-10-04 15:21:55 +0200 | [diff] [blame] | 2130 | call_->OnAudioTransportOverheadChanged(network_route.packet_overhead); |
Honghai Zhang | cc411c0 | 2016-03-29 17:27:21 -0700 | [diff] [blame] | 2131 | } |
| 2132 | |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 2133 | bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 2134 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 2135 | const auto it = send_streams_.find(ssrc); |
| 2136 | if (it == send_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2137 | RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2138 | return false; |
| 2139 | } |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 2140 | it->second->SetMuted(muted); |
| 2141 | |
| 2142 | // TODO(solenberg): |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 +0000 | [diff] [blame] | 2143 | // We set the AGC to mute state only when all the channels are muted. |
| 2144 | // This implementation is not ideal, instead we should signal the AGC when |
| 2145 | // the mic channel is muted/unmuted. We can't do it today because there |
| 2146 | // is no good way to know which stream is mapping to the mic channel. |
| 2147 | bool all_muted = muted; |
solenberg | 9421853 | 2016-06-16 10:53:22 -0700 | [diff] [blame] | 2148 | for (const auto& kv : send_streams_) { |
| 2149 | all_muted = all_muted && kv.second->muted(); |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 +0000 | [diff] [blame] | 2150 | } |
solenberg | 059fb44 | 2016-10-26 05:12:24 -0700 | [diff] [blame] | 2151 | engine()->apm()->set_output_will_be_muted(all_muted); |
buildbot@webrtc.org | 6b21b71 | 2014-07-31 15:08:53 +0000 | [diff] [blame] | 2152 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2153 | return true; |
| 2154 | } |
| 2155 | |
deadbeef | 8034614 | 2016-04-27 14:17:10 -0700 | [diff] [blame] | 2156 | bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2157 | RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate."; |
deadbeef | 8034614 | 2016-04-27 14:17:10 -0700 | [diff] [blame] | 2158 | max_send_bitrate_bps_ = bps; |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 2159 | bool success = true; |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 2160 | for (const auto& kv : send_streams_) { |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 2161 | if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) { |
| 2162 | success = false; |
skvlad | e0d4637 | 2016-04-07 22:59:22 -0700 | [diff] [blame] | 2163 | } |
| 2164 | } |
minyue | 7a97344 | 2016-10-20 03:27:12 -0700 | [diff] [blame] | 2165 | return success; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2166 | } |
| 2167 | |
skvlad | 7a43d25 | 2016-03-22 15:32:27 -0700 | [diff] [blame] | 2168 | void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) { |
| 2169 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2170 | RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); |
skvlad | 7a43d25 | 2016-03-22 15:32:27 -0700 | [diff] [blame] | 2171 | call_->SignalChannelNetworkState( |
| 2172 | webrtc::MediaType::AUDIO, |
| 2173 | ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); |
| 2174 | } |
| 2175 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2176 | bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { |
Peter Boström | ca8b404 | 2016-03-08 14:24:13 -0800 | [diff] [blame] | 2177 | TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats"); |
solenberg | 566ef24 | 2015-11-06 15:34:49 -0800 | [diff] [blame] | 2178 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2179 | RTC_DCHECK(info); |
solenberg | d97ec30 | 2015-10-07 01:40:33 -0700 | [diff] [blame] | 2180 | |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2181 | // Get SSRC and stats for each sender. |
hbos | 1acfbd2 | 2016-11-17 23:43:29 -0800 | [diff] [blame] | 2182 | RTC_DCHECK_EQ(info->senders.size(), 0U); |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2183 | for (const auto& stream : send_streams_) { |
Ivo Creusen | 56d4609 | 2017-11-24 17:29:59 +0100 | [diff] [blame] | 2184 | webrtc::AudioSendStream::Stats stats = |
| 2185 | stream.second->GetStats(recv_streams_.size() > 0); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 2186 | VoiceSenderInfo sinfo; |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2187 | sinfo.add_ssrc(stats.local_ssrc); |
| 2188 | sinfo.bytes_sent = stats.bytes_sent; |
| 2189 | sinfo.packets_sent = stats.packets_sent; |
| 2190 | sinfo.packets_lost = stats.packets_lost; |
| 2191 | sinfo.fraction_lost = stats.fraction_lost; |
| 2192 | sinfo.codec_name = stats.codec_name; |
hbos | 1acfbd2 | 2016-11-17 23:43:29 -0800 | [diff] [blame] | 2193 | sinfo.codec_payload_type = stats.codec_payload_type; |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2194 | sinfo.ext_seqnum = stats.ext_seqnum; |
| 2195 | sinfo.jitter_ms = stats.jitter_ms; |
| 2196 | sinfo.rtt_ms = stats.rtt_ms; |
| 2197 | sinfo.audio_level = stats.audio_level; |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 2198 | sinfo.total_input_energy = stats.total_input_energy; |
| 2199 | sinfo.total_input_duration = stats.total_input_duration; |
Taylor Brandstetter | 1a018dc | 2016-03-08 12:37:39 -0800 | [diff] [blame] | 2200 | sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false); |
ivoc | e1198e0 | 2017-09-08 08:13:19 -0700 | [diff] [blame] | 2201 | sinfo.ana_statistics = stats.ana_statistics; |
Ivo Creusen | 56d4609 | 2017-11-24 17:29:59 +0100 | [diff] [blame] | 2202 | sinfo.apm_statistics = stats.apm_statistics; |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 2203 | info->senders.push_back(sinfo); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2204 | } |
| 2205 | |
solenberg | 85a0496 | 2015-10-27 03:35:21 -0700 | [diff] [blame] | 2206 | // Get SSRC and stats for each receiver. |
hbos | 1acfbd2 | 2016-11-17 23:43:29 -0800 | [diff] [blame] | 2207 | RTC_DCHECK_EQ(info->receivers.size(), 0U); |
solenberg | 7add058 | 2015-11-20 09:59:34 -0800 | [diff] [blame] | 2208 | for (const auto& stream : recv_streams_) { |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2209 | uint32_t ssrc = stream.first; |
| 2210 | // When SSRCs are unsignaled, there's only one audio MediaStreamTrack, but |
| 2211 | // multiple RTP streams can be received over time (if the SSRC changes for |
| 2212 | // whatever reason). We only want the RTCMediaStreamTrackStats to represent |
| 2213 | // the stats for the most recent stream (the one whose audio is actually |
| 2214 | // routed to the MediaStreamTrack), so here we ignore any unsignaled SSRCs |
| 2215 | // except for the most recent one (last in the vector). This is somewhat of |
| 2216 | // a hack, and means you don't get *any* stats for these inactive streams, |
| 2217 | // but it's slightly better than the previous behavior, which was "highest |
| 2218 | // SSRC wins". |
| 2219 | // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8158 |
| 2220 | if (!unsignaled_recv_ssrcs_.empty()) { |
| 2221 | auto end_it = --unsignaled_recv_ssrcs_.end(); |
| 2222 | if (std::find(unsignaled_recv_ssrcs_.begin(), end_it, ssrc) != end_it) { |
| 2223 | continue; |
| 2224 | } |
| 2225 | } |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2226 | webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats(); |
| 2227 | VoiceReceiverInfo rinfo; |
| 2228 | rinfo.add_ssrc(stats.remote_ssrc); |
| 2229 | rinfo.bytes_rcvd = stats.bytes_rcvd; |
| 2230 | rinfo.packets_rcvd = stats.packets_rcvd; |
| 2231 | rinfo.packets_lost = stats.packets_lost; |
| 2232 | rinfo.fraction_lost = stats.fraction_lost; |
| 2233 | rinfo.codec_name = stats.codec_name; |
hbos | 1acfbd2 | 2016-11-17 23:43:29 -0800 | [diff] [blame] | 2234 | rinfo.codec_payload_type = stats.codec_payload_type; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2235 | rinfo.ext_seqnum = stats.ext_seqnum; |
| 2236 | rinfo.jitter_ms = stats.jitter_ms; |
| 2237 | rinfo.jitter_buffer_ms = stats.jitter_buffer_ms; |
| 2238 | rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms; |
| 2239 | rinfo.delay_estimate_ms = stats.delay_estimate_ms; |
| 2240 | rinfo.audio_level = stats.audio_level; |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 2241 | rinfo.total_output_energy = stats.total_output_energy; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 2242 | rinfo.total_samples_received = stats.total_samples_received; |
zstein | e76bd3a | 2017-07-14 12:17:49 -0700 | [diff] [blame] | 2243 | rinfo.total_output_duration = stats.total_output_duration; |
Steve Anton | 2dbc69f | 2017-08-24 17:15:13 -0700 | [diff] [blame] | 2244 | rinfo.concealed_samples = stats.concealed_samples; |
Gustaf Ullberg | 9a2e906 | 2017-09-18 09:28:20 +0200 | [diff] [blame] | 2245 | rinfo.concealment_events = stats.concealment_events; |
Gustaf Ullberg | b0a0207 | 2017-10-02 12:00:34 +0200 | [diff] [blame] | 2246 | rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2247 | rinfo.expand_rate = stats.expand_rate; |
| 2248 | rinfo.speech_expand_rate = stats.speech_expand_rate; |
| 2249 | rinfo.secondary_decoded_rate = stats.secondary_decoded_rate; |
minyue-webrtc | 0e320ec | 2017-08-28 13:51:27 +0200 | [diff] [blame] | 2250 | rinfo.secondary_discarded_rate = stats.secondary_discarded_rate; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2251 | rinfo.accelerate_rate = stats.accelerate_rate; |
| 2252 | rinfo.preemptive_expand_rate = stats.preemptive_expand_rate; |
Jakob Ivarsson | 352ce5c | 2018-11-27 12:52:16 +0100 | [diff] [blame] | 2253 | rinfo.delayed_packet_outage_samples = stats.delayed_packet_outage_samples; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2254 | rinfo.decoding_calls_to_silence_generator = |
| 2255 | stats.decoding_calls_to_silence_generator; |
| 2256 | rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq; |
| 2257 | rinfo.decoding_normal = stats.decoding_normal; |
| 2258 | rinfo.decoding_plc = stats.decoding_plc; |
| 2259 | rinfo.decoding_cng = stats.decoding_cng; |
| 2260 | rinfo.decoding_plc_cng = stats.decoding_plc_cng; |
henrik.lundin | 6348978 | 2016-09-20 01:47:12 -0700 | [diff] [blame] | 2261 | rinfo.decoding_muted_output = stats.decoding_muted_output; |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2262 | rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms; |
Ruslan Burakov | 8af8896 | 2018-11-22 17:21:10 +0100 | [diff] [blame] | 2263 | rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes; |
| 2264 | |
Fredrik Solenberg | 4f4ec0a | 2015-10-22 10:49:27 +0200 | [diff] [blame] | 2265 | info->receivers.push_back(rinfo); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2266 | } |
| 2267 | |
hbos | 1acfbd2 | 2016-11-17 23:43:29 -0800 | [diff] [blame] | 2268 | // Get codec info |
| 2269 | for (const AudioCodec& codec : send_codecs_) { |
| 2270 | webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| 2271 | info->send_codecs.insert( |
| 2272 | std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| 2273 | } |
| 2274 | for (const AudioCodec& codec : recv_codecs_) { |
| 2275 | webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| 2276 | info->receive_codecs.insert( |
| 2277 | std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| 2278 | } |
| 2279 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2280 | return true; |
| 2281 | } |
| 2282 | |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 2283 | void WebRtcVoiceMediaChannel::SetRawAudioSink( |
| 2284 | uint32_t ssrc, |
kwiberg | 686a8ef | 2016-02-26 03:00:35 -0800 | [diff] [blame] | 2285 | std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 2286 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2287 | RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" |
| 2288 | << ssrc << " " << (sink ? "(ptr)" : "NULL"); |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 2289 | if (ssrc == 0) { |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2290 | if (!unsignaled_recv_ssrcs_.empty()) { |
kwiberg | 686a8ef | 2016-02-26 03:00:35 -0800 | [diff] [blame] | 2291 | std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 2292 | sink ? new ProxySink(sink.get()) : nullptr); |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2293 | SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink)); |
deadbeef | 884f585 | 2016-01-15 09:20:04 -0800 | [diff] [blame] | 2294 | } |
| 2295 | default_sink_ = std::move(sink); |
| 2296 | return; |
| 2297 | } |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 2298 | const auto it = recv_streams_.find(ssrc); |
| 2299 | if (it == recv_streams_.end()) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2300 | RTC_LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc; |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 2301 | return; |
| 2302 | } |
deadbeef | 2d110be | 2016-01-13 12:00:26 -0800 | [diff] [blame] | 2303 | it->second->SetRawAudioSink(std::move(sink)); |
Tommi | f888bb5 | 2015-12-12 01:37:01 +0100 | [diff] [blame] | 2304 | } |
| 2305 | |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 2306 | std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources( |
| 2307 | uint32_t ssrc) const { |
| 2308 | auto it = recv_streams_.find(ssrc); |
Zhi Huang | fa266ef | 2017-12-13 10:27:46 -0800 | [diff] [blame] | 2309 | if (it == recv_streams_.end()) { |
| 2310 | RTC_LOG(LS_ERROR) << "Attempting to get contributing sources for SSRC:" |
| 2311 | << ssrc << " which doesn't exist."; |
| 2312 | return std::vector<webrtc::RtpSource>(); |
| 2313 | } |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 2314 | return it->second->GetSources(); |
| 2315 | } |
| 2316 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2317 | bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream( |
| 2318 | uint32_t ssrc) { |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2319 | RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 2320 | auto it = std::find(unsignaled_recv_ssrcs_.begin(), |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 2321 | unsignaled_recv_ssrcs_.end(), ssrc); |
solenberg | 2100c0b | 2017-03-01 11:29:29 -0800 | [diff] [blame] | 2322 | if (it != unsignaled_recv_ssrcs_.end()) { |
| 2323 | unsignaled_recv_ssrcs_.erase(it); |
| 2324 | return true; |
| 2325 | } |
| 2326 | return false; |
| 2327 | } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 2328 | } // namespace cricket |
| 2329 | |
| 2330 | #endif // HAVE_WEBRTC_VOICE |