| File: | root/firefox-clang/obj-x86_64-pc-linux-gnu/third_party/libwebrtc/modules/audio_coding/neteq_gn/./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc |
| Warning: | line 265, column 5 Value stored to 'main_timestamp' is never read |
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| 1 | /* |
| 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include "modules/audio_coding/neteq/neteq_impl.h" |
| 12 | |
| 13 | #include <algorithm> |
| 14 | #include <cstdint> |
| 15 | #include <cstring> |
| 16 | #include <list> |
| 17 | #include <map> |
| 18 | #include <memory> |
| 19 | #include <optional> |
| 20 | #include <span> |
| 21 | #include <utility> |
| 22 | #include <vector> |
| 23 | |
| 24 | #include "absl/strings/str_cat.h" |
| 25 | #include "api/audio/audio_frame.h" |
| 26 | #include "api/audio/audio_view.h" |
| 27 | #include "api/audio_codecs/audio_decoder.h" |
| 28 | #include "api/audio_codecs/audio_decoder_factory.h" |
| 29 | #include "api/audio_codecs/audio_format.h" |
| 30 | #include "api/environment/environment.h" |
| 31 | #include "api/neteq/neteq.h" |
| 32 | #include "api/neteq/neteq_controller.h" |
| 33 | #include "api/neteq/neteq_controller_factory.h" |
| 34 | #include "api/neteq/tick_timer.h" |
| 35 | #include "api/rtp_headers.h" |
| 36 | #include "api/rtp_packet_info.h" |
| 37 | #include "api/rtp_packet_infos.h" |
| 38 | #include "api/scoped_refptr.h" |
| 39 | #include "api/units/time_delta.h" |
| 40 | #include "modules/audio_coding/codecs/cng/webrtc_cng.h" |
| 41 | #include "modules/audio_coding/neteq/accelerate.h" |
| 42 | #include "modules/audio_coding/neteq/background_noise.h" |
| 43 | #include "modules/audio_coding/neteq/comfort_noise.h" |
| 44 | #include "modules/audio_coding/neteq/decoder_database.h" |
| 45 | #include "modules/audio_coding/neteq/dtmf_buffer.h" |
| 46 | #include "modules/audio_coding/neteq/dtmf_tone_generator.h" |
| 47 | #include "modules/audio_coding/neteq/expand.h" |
| 48 | #include "modules/audio_coding/neteq/merge.h" |
| 49 | #include "modules/audio_coding/neteq/normal.h" |
| 50 | #include "modules/audio_coding/neteq/packet.h" |
| 51 | #include "modules/audio_coding/neteq/packet_buffer.h" |
| 52 | #include "modules/audio_coding/neteq/preemptive_expand.h" |
| 53 | #include "modules/audio_coding/neteq/red_payload_splitter.h" |
| 54 | #include "modules/audio_coding/neteq/statistics_calculator.h" |
| 55 | #include "modules/audio_coding/neteq/sync_buffer.h" |
| 56 | #include "modules/audio_coding/neteq/time_stretch.h" |
| 57 | #include "modules/audio_coding/neteq/timestamp_scaler.h" |
| 58 | #include "rtc_base/checks.h" |
| 59 | #include "rtc_base/logging.h" |
| 60 | #include "rtc_base/numerics/safe_conversions.h" |
| 61 | #include "rtc_base/sanitizer.h" |
| 62 | #include "rtc_base/trace_event.h" |
| 63 | #include "system_wrappers/include/clock.h" |
| 64 | |
| 65 | namespace webrtc { |
| 66 | namespace { |
| 67 | |
| 68 | AudioFrame::SpeechType ToSpeechType(NetEqImpl::OutputType type) { |
| 69 | switch (type) { |
| 70 | case NetEqImpl::OutputType::kNormalSpeech: { |
| 71 | return AudioFrame::kNormalSpeech; |
| 72 | } |
| 73 | case NetEqImpl::OutputType::kCNG: { |
| 74 | return AudioFrame::kCNG; |
| 75 | } |
| 76 | case NetEqImpl::OutputType::kPLC: { |
| 77 | return AudioFrame::kPLC; |
| 78 | } |
| 79 | case NetEqImpl::OutputType::kPLCCNG: { |
| 80 | return AudioFrame::kPLCCNG; |
| 81 | } |
| 82 | case NetEqImpl::OutputType::kCodecPLC: { |
| 83 | return AudioFrame::kCodecPLC; |
| 84 | } |
| 85 | default: |
| 86 | RTC_DCHECK_NOTREACHED()(false) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 86, "false") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 87 | return AudioFrame::kUndefined; |
| 88 | } |
| 89 | } |
| 90 | |
| 91 | // Returns true if both payload types are known to the decoder database, and |
| 92 | // have the same sample rate. |
| 93 | bool EqualSampleRates(uint8_t pt1, |
| 94 | uint8_t pt2, |
| 95 | const DecoderDatabase& decoder_database) { |
| 96 | auto* di1 = decoder_database.GetDecoderInfo(pt1); |
| 97 | auto* di2 = decoder_database.GetDecoderInfo(pt2); |
| 98 | return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz(); |
| 99 | } |
| 100 | |
| 101 | } // namespace |
| 102 | |
| 103 | NetEqImpl::Dependencies::Dependencies( |
| 104 | const Environment& env, |
| 105 | const NetEq::Config& config, |
| 106 | scoped_refptr<AudioDecoderFactory> decoder_factory, |
| 107 | const NetEqControllerFactory& controller_factory) |
| 108 | : env(env), |
| 109 | tick_timer(new TickTimer), |
| 110 | stats(std::make_unique<StatisticsCalculator>(tick_timer.get())), |
| 111 | decoder_database( |
| 112 | std::make_unique<DecoderDatabase>(env, std::move(decoder_factory))), |
| 113 | dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)), |
| 114 | dtmf_tone_generator(new DtmfToneGenerator), |
| 115 | packet_buffer(new PacketBuffer(env.field_trials(), |
| 116 | config.max_packets_in_buffer, |
| 117 | tick_timer.get(), |
| 118 | stats.get())), |
| 119 | neteq_controller(controller_factory.Create( |
| 120 | env, |
| 121 | {.allow_time_stretching = !config.for_test_no_time_stretching, |
| 122 | .max_packets_in_buffer = |
| 123 | static_cast<int>(config.max_packets_in_buffer), |
| 124 | .base_min_delay_ms = config.min_delay_ms, |
| 125 | .tick_timer = tick_timer.get()})), |
| 126 | red_payload_splitter(new RedPayloadSplitter), |
| 127 | timestamp_scaler(new TimestampScaler(*decoder_database)), |
| 128 | accelerate_factory(new AccelerateFactory), |
| 129 | expand_factory(new ExpandFactory), |
| 130 | preemptive_expand_factory(new PreemptiveExpandFactory) {} |
| 131 | |
| 132 | NetEqImpl::Dependencies::~Dependencies() = default; |
| 133 | |
| 134 | NetEqImpl::NetEqImpl(const NetEq::Config& config, |
| 135 | Dependencies&& deps, |
| 136 | bool create_components) |
| 137 | : env_(deps.env), |
| 138 | tick_timer_(std::move(deps.tick_timer)), |
| 139 | decoder_database_(std::move(deps.decoder_database)), |
| 140 | dtmf_buffer_(std::move(deps.dtmf_buffer)), |
| 141 | dtmf_tone_generator_(std::move(deps.dtmf_tone_generator)), |
| 142 | packet_buffer_(std::move(deps.packet_buffer)), |
| 143 | red_payload_splitter_(std::move(deps.red_payload_splitter)), |
| 144 | timestamp_scaler_(std::move(deps.timestamp_scaler)), |
| 145 | expand_factory_(std::move(deps.expand_factory)), |
| 146 | accelerate_factory_(std::move(deps.accelerate_factory)), |
| 147 | preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)), |
| 148 | stats_(std::move(deps.stats)), |
| 149 | controller_(std::move(deps.neteq_controller)), |
| 150 | last_mode_(Mode::kNormal), |
| 151 | decoded_buffer_length_(kMaxFrameSize), |
| 152 | decoded_buffer_(new int16_t[decoded_buffer_length_]), |
| 153 | playout_timestamp_(0), |
| 154 | new_codec_(false), |
| 155 | timestamp_(0), |
| 156 | reset_decoder_(false), |
| 157 | first_packet_(true), |
| 158 | enable_fast_accelerate_(config.enable_fast_accelerate), |
| 159 | enable_muted_state_(config.enable_muted_state), |
| 160 | no_time_stretching_(config.for_test_no_time_stretching) { |
| 161 | RTC_LOG(LS_INFO)!::webrtc::LogMessage::IsNoop<::webrtc::LS_INFO>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 161, ::webrtc::LS_INFO) << "NetEq config: " << config.ToString(); |
| 162 | int fs = config.sample_rate_hz; |
| 163 | if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) { |
| 164 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 164, ::webrtc::LS_ERROR) << "Sample rate " << fs |
| 165 | << " Hz not supported. " |
| 166 | "Changing to 8000 Hz."; |
| 167 | fs = 8000; |
| 168 | } |
| 169 | controller_->SetMaximumDelay(config.max_delay_ms); |
| 170 | fs_hz_ = fs; |
| 171 | fs_mult_ = fs / 8000; |
| 172 | last_output_sample_rate_hz_ = fs; |
| 173 | output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_); |
| 174 | controller_->SetSampleRate(fs_hz_, output_size_samples_); |
| 175 | decoder_frame_length_ = 2 * output_size_samples_; // 20 ms. |
| 176 | if (create_components) { |
| 177 | SetSampleRateAndChannels(fs, 1); // Default is 1 channel. |
| 178 | } |
| 179 | } |
| 180 | |
| 181 | NetEqImpl::~NetEqImpl() = default; |
| 182 | |
| 183 | int NetEqImpl::InsertPacket(const RTPHeader& rtp_header, |
| 184 | std::span<const uint8_t> payload, |
| 185 | const RtpPacketInfo& packet_info) { |
| 186 | MsanCheckInitialized(payload); |
| 187 | TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket")static const unsigned char* trace_event_unique_catstatic187 = reinterpret_cast<const unsigned char*>("webrtc");; webrtc ::trace_event_internal::TraceEndOnScopeClose trace_event_unique_profileScope187 ; if (*trace_event_unique_catstatic187) { webrtc::trace_event_internal ::AddTraceEvent( ('B'), trace_event_unique_catstatic187, "NetEqImpl::InsertPacket" , webrtc::trace_event_internal::kNoEventId, (static_cast<unsigned char>(0))); trace_event_unique_profileScope187 .Initialize (trace_event_unique_catstatic187, "NetEqImpl::InsertPacket"); }; |
| 188 | if (payload.empty()) { |
| 189 | RTC_LOG_F(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 189, ::webrtc::LS_ERROR) << __PRETTY_FUNCTION__ << ": " << "payload is empty"; |
| 190 | return kFail; |
| 191 | } |
| 192 | stats_->ReceivedPacket(); |
| 193 | |
| 194 | PacketList packet_list; |
| 195 | // Insert packet in a packet list. |
| 196 | packet_list.push_back([&rtp_header, &payload] { |
| 197 | // Convert to Packet. |
| 198 | Packet packet; |
| 199 | packet.payload_type = rtp_header.payloadType; |
| 200 | packet.sequence_number = rtp_header.sequenceNumber; |
| 201 | packet.timestamp = rtp_header.timestamp; |
| 202 | packet.payload.SetData(payload.data(), payload.size()); |
| 203 | // Waiting time will be set upon inserting the packet in the buffer. |
| 204 | RTC_DCHECK(!packet.waiting_time)(!packet.waiting_time) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 204, "!packet.waiting_time") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 205 | return packet; |
| 206 | }()); |
| 207 | |
| 208 | bool update_sample_rate_and_channels = first_packet_; |
| 209 | |
| 210 | if (update_sample_rate_and_channels) { |
| 211 | // Reset timestamp scaling. |
| 212 | timestamp_scaler_->Reset(); |
| 213 | } |
| 214 | |
| 215 | if (!decoder_database_->IsRed(rtp_header.payloadType)) { |
| 216 | // Scale timestamp to internal domain (only for some codecs). |
| 217 | timestamp_scaler_->ToInternal(&packet_list); |
| 218 | } |
| 219 | |
| 220 | // Store these for later use, since the first packet may very well disappear |
| 221 | // before we need these values. |
| 222 | uint32_t main_timestamp = packet_list.front().timestamp; |
| 223 | |
| 224 | // Reinitialize NetEq if it's needed (changed SSRC or first call). |
| 225 | if (update_sample_rate_and_channels) { |
| 226 | // Note: `first_packet_` will be cleared further down in this method, once |
| 227 | // the packet has been successfully inserted into the packet buffer. |
| 228 | |
| 229 | // Flush the packet buffer and DTMF buffer. |
| 230 | packet_buffer_->Flush(); |
| 231 | dtmf_buffer_->Flush(); |
| 232 | |
| 233 | // Update audio buffer timestamp. |
| 234 | sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_); |
| 235 | |
| 236 | // Update codecs. |
| 237 | timestamp_ = main_timestamp; |
| 238 | } |
| 239 | |
| 240 | // Check for RED payload type, and separate payloads into several packets. |
| 241 | if (decoder_database_->IsRed(rtp_header.payloadType)) { |
| 242 | if (!red_payload_splitter_->SplitRed(&packet_list)) { |
| 243 | return kFail; |
| 244 | } |
| 245 | // Only accept a few RED payloads of the same type as the main data, |
| 246 | // DTMF events and CNG. |
| 247 | red_payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_); |
| 248 | if (packet_list.empty()) { |
| 249 | return kFail; |
| 250 | } |
| 251 | } |
| 252 | |
| 253 | // Check payload types. |
| 254 | if (decoder_database_->CheckPayloadTypes(packet_list) == |
| 255 | DecoderDatabase::kDecoderNotFound) { |
| 256 | return kFail; |
| 257 | } |
| 258 | |
| 259 | RTC_DCHECK(!packet_list.empty())(!packet_list.empty()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 259, "!packet_list.empty()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 260 | |
| 261 | // Update main_timestamp, if new packets appear in the list |
| 262 | // after RED splitting. |
| 263 | if (decoder_database_->IsRed(rtp_header.payloadType)) { |
| 264 | timestamp_scaler_->ToInternal(&packet_list); |
| 265 | main_timestamp = packet_list.front().timestamp; |
Value stored to 'main_timestamp' is never read | |
| 266 | } |
| 267 | |
| 268 | // Process DTMF payloads. Cycle through the list of packets, and pick out any |
| 269 | // DTMF payloads found. |
| 270 | PacketList::iterator it = packet_list.begin(); |
| 271 | while (it != packet_list.end()) { |
| 272 | const Packet& current_packet = (*it); |
| 273 | RTC_DCHECK(!current_packet.payload.empty())(!current_packet.payload.empty()) ? static_cast<void>(0 ) : ::webrtc::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 273, "!current_packet.payload.empty()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 274 | if (decoder_database_->IsDtmf(current_packet.payload_type)) { |
| 275 | DtmfEvent event; |
| 276 | int ret = DtmfBuffer::ParseEvent(current_packet.timestamp, |
| 277 | current_packet.payload.data(), |
| 278 | current_packet.payload.size(), &event); |
| 279 | if (ret != DtmfBuffer::kOK) { |
| 280 | return kFail; |
| 281 | } |
| 282 | if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) { |
| 283 | return kFail; |
| 284 | } |
| 285 | it = packet_list.erase(it); |
| 286 | } else { |
| 287 | ++it; |
| 288 | } |
| 289 | } |
| 290 | |
| 291 | PacketList parsed_packet_list; |
| 292 | while (!packet_list.empty()) { |
| 293 | Packet& packet = packet_list.front(); |
| 294 | const DecoderDatabase::DecoderInfo* info = |
| 295 | decoder_database_->GetDecoderInfo(packet.payload_type); |
| 296 | if (!info) { |
| 297 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 297, ::webrtc::LS_WARNING) << "SplitAudio unknown payload type"; |
| 298 | return kFail; |
| 299 | } |
| 300 | |
| 301 | if (info->IsComfortNoise()) { |
| 302 | // Carry comfort noise packets along. |
| 303 | parsed_packet_list.splice(parsed_packet_list.end(), packet_list, |
| 304 | packet_list.begin()); |
| 305 | } else { |
| 306 | const uint16_t sequence_number = packet.sequence_number; |
| 307 | const uint8_t payload_type = packet.payload_type; |
| 308 | const Packet::Priority original_priority = packet.priority; |
| 309 | auto packet_from_result = [&](AudioDecoder::ParseResult& result) { |
| 310 | Packet new_packet; |
| 311 | new_packet.sequence_number = sequence_number; |
| 312 | new_packet.payload_type = payload_type; |
| 313 | new_packet.timestamp = result.timestamp; |
| 314 | new_packet.priority.codec_level = result.priority; |
| 315 | new_packet.priority.red_level = original_priority.red_level; |
| 316 | // Only associate the header information with the primary packet. |
| 317 | if (new_packet.timestamp == packet_info.rtp_timestamp()) { |
| 318 | new_packet.packet_info = packet_info; |
| 319 | } |
| 320 | new_packet.frame = std::move(result.frame); |
| 321 | return new_packet; |
| 322 | }; |
| 323 | |
| 324 | std::vector<AudioDecoder::ParseResult> results = |
| 325 | info->GetDecoder()->ParsePayload(std::move(packet.payload), |
| 326 | packet.timestamp); |
| 327 | if (results.empty()) { |
| 328 | packet_list.pop_front(); |
| 329 | } else { |
| 330 | bool first = true; |
| 331 | for (auto& result : results) { |
| 332 | RTC_DCHECK(result.frame)(result.frame) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 332, "result.frame") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 333 | RTC_DCHECK_GE(result.priority, 0)::webrtc::SafeGe((result.priority), (0)) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 333, "result.priority" " " ">=" " " "0") & ::webrtc:: webrtc_checks_impl::LogStreamer<>() << (result.priority ) << (0); |
| 334 | if (first) { |
| 335 | // Re-use the node and move it to parsed_packet_list. |
| 336 | packet_list.front() = packet_from_result(result); |
| 337 | parsed_packet_list.splice(parsed_packet_list.end(), packet_list, |
| 338 | packet_list.begin()); |
| 339 | first = false; |
| 340 | } else { |
| 341 | parsed_packet_list.push_back(packet_from_result(result)); |
| 342 | } |
| 343 | } |
| 344 | } |
| 345 | } |
| 346 | } |
| 347 | |
| 348 | // Calculate the number of primary (non-FEC/RED) packets. |
| 349 | const size_t number_of_primary_packets = std::count_if( |
| 350 | parsed_packet_list.begin(), parsed_packet_list.end(), |
| 351 | [](const Packet& in) { return in.priority.codec_level == 0; }); |
| 352 | if (number_of_primary_packets < parsed_packet_list.size()) { |
| 353 | stats_->SecondaryPacketsReceived(parsed_packet_list.size() - |
| 354 | number_of_primary_packets); |
| 355 | } |
| 356 | |
| 357 | const int target_level_ms = controller_->TargetLevelMs(); |
| 358 | for (Packet& packet : parsed_packet_list) { |
| 359 | NetEqController::PacketArrivedInfo info = ToPacketArrivedInfo(packet); |
| 360 | if (MaybeChangePayloadType(packet.payload_type)) { |
| 361 | packet_buffer_->Flush(); |
| 362 | new_codec_ = true; |
| 363 | update_sample_rate_and_channels = true; |
| 364 | info.buffer_flush = true; |
| 365 | } |
| 366 | int return_val = packet_buffer_->InsertPacket( |
| 367 | std::move(packet), decoder_frame_length_, last_output_sample_rate_hz_, |
| 368 | target_level_ms); |
| 369 | if (return_val == PacketBuffer::kFlushed) { |
| 370 | new_codec_ = true; |
| 371 | update_sample_rate_and_channels = true; |
| 372 | info.buffer_flush = true; |
| 373 | } else if (return_val == PacketBuffer::kPartialFlush) { |
| 374 | // Forward sync buffer timestamp. |
| 375 | const Packet* next_packet = packet_buffer_->PeekNextPacket(); |
| 376 | if (next_packet) { |
| 377 | timestamp_ = next_packet->timestamp; |
| 378 | sync_buffer_->IncreaseEndTimestamp(timestamp_ - |
| 379 | sync_buffer_->end_timestamp()); |
| 380 | } |
| 381 | info.buffer_flush = true; |
| 382 | } else if (return_val != PacketBuffer::kOK) { |
| 383 | // An error occurred. |
| 384 | return kFail; |
| 385 | } |
| 386 | |
| 387 | const bool should_update_stats = !new_codec_; |
| 388 | auto relative_delay = |
| 389 | controller_->PacketArrived(fs_hz_, should_update_stats, info); |
| 390 | if (relative_delay) { |
| 391 | stats_->RelativePacketArrivalDelay(relative_delay.value()); |
| 392 | } |
| 393 | } |
| 394 | |
| 395 | if (first_packet_) { |
| 396 | first_packet_ = false; |
| 397 | // Update the codec on the next GetAudio call. |
| 398 | new_codec_ = true; |
| 399 | } |
| 400 | |
| 401 | if (current_rtp_payload_type_) { |
| 402 | RTC_DCHECK(decoder_database_->GetDecoderInfo(*current_rtp_payload_type_))(decoder_database_->GetDecoderInfo(*current_rtp_payload_type_ )) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 402, "decoder_database_->GetDecoderInfo(*current_rtp_payload_type_)" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 403 | << "Payload type " << static_cast<int>(*current_rtp_payload_type_) |
| 404 | << " is unknown where it shouldn't be"; |
| 405 | } |
| 406 | |
| 407 | if (update_sample_rate_and_channels && !packet_buffer_->Empty()) { |
| 408 | // We do not use `current_rtp_payload_type_` to |set payload_type|, but |
| 409 | // get the next RTP header from `packet_buffer_` to obtain the payload type. |
| 410 | // The reason for it is the following corner case. If NetEq receives a |
| 411 | // CNG packet with a sample rate different than the current CNG then it |
| 412 | // flushes its buffer, assuming send codec must have been changed. However, |
| 413 | // payload type of the hypothetically new send codec is not known. |
| 414 | const Packet* next_packet = packet_buffer_->PeekNextPacket(); |
| 415 | RTC_DCHECK(next_packet)(next_packet) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 415, "next_packet") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 416 | const int payload_type = next_packet->payload_type; |
| 417 | size_t channels = 1; |
| 418 | if (!decoder_database_->IsComfortNoise(payload_type)) { |
| 419 | AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type); |
| 420 | RTC_DCHECK(decoder)(decoder) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 420, "decoder") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); // Payloads are already checked to be valid. |
| 421 | channels = decoder->Channels(); |
| 422 | RTC_DCHECK_LE(channels, kMaxNumberOfAudioChannels)::webrtc::SafeLe((channels), (kMaxNumberOfAudioChannels)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 422, "channels" " " "<=" " " "kMaxNumberOfAudioChannels" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (channels) << (kMaxNumberOfAudioChannels); |
| 423 | } |
| 424 | const DecoderDatabase::DecoderInfo* decoder_info = |
| 425 | decoder_database_->GetDecoderInfo(payload_type); |
| 426 | RTC_DCHECK(decoder_info)(decoder_info) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 426, "decoder_info") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 427 | if (decoder_info->SampleRateHz() != fs_hz_ || |
| 428 | channels != algorithm_buffer_->Channels()) { |
| 429 | RTC_DCHECK_LE(channels, kMaxNumberOfAudioChannels)::webrtc::SafeLe((channels), (kMaxNumberOfAudioChannels)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 429, "channels" " " "<=" " " "kMaxNumberOfAudioChannels" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (channels) << (kMaxNumberOfAudioChannels); |
| 430 | SetSampleRateAndChannels(decoder_info->SampleRateHz(), channels); |
| 431 | } |
| 432 | } |
| 433 | |
| 434 | return kOK; |
| 435 | } |
| 436 | |
| 437 | int NetEqImpl::GetAudio(AudioFrame* audio_frame, |
| 438 | bool* muted, |
| 439 | int* current_sample_rate_hz, |
| 440 | std::optional<Operation> action_override) { |
| 441 | TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio")static const unsigned char* trace_event_unique_catstatic441 = reinterpret_cast<const unsigned char*>("webrtc");; webrtc ::trace_event_internal::TraceEndOnScopeClose trace_event_unique_profileScope441 ; if (*trace_event_unique_catstatic441) { webrtc::trace_event_internal ::AddTraceEvent( ('B'), trace_event_unique_catstatic441, "NetEqImpl::GetAudio" , webrtc::trace_event_internal::kNoEventId, (static_cast<unsigned char>(0))); trace_event_unique_profileScope441 .Initialize (trace_event_unique_catstatic441, "NetEqImpl::GetAudio"); }; |
| 442 | if (GetAudioInternal(audio_frame, action_override) != kNoError) { |
| 443 | return kFail; |
| 444 | } |
| 445 | stats_->IncreaseCounter(output_size_samples_, fs_hz_); |
| 446 | RTC_DCHECK_EQ(audio_frame->sample_rate_hz_,::webrtc::SafeEq((audio_frame->sample_rate_hz_), (dchecked_cast <int>(audio_frame->samples_per_channel_ * 100))) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 447, "audio_frame->sample_rate_hz_" " " "==" " " "dchecked_cast<int>(audio_frame->samples_per_channel_ * 100)" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (audio_frame->sample_rate_hz_) << (dchecked_cast< int>(audio_frame->samples_per_channel_ * 100)) |
| 447 | dchecked_cast<int>(audio_frame->samples_per_channel_ * 100))::webrtc::SafeEq((audio_frame->sample_rate_hz_), (dchecked_cast <int>(audio_frame->samples_per_channel_ * 100))) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 447, "audio_frame->sample_rate_hz_" " " "==" " " "dchecked_cast<int>(audio_frame->samples_per_channel_ * 100)" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (audio_frame->sample_rate_hz_) << (dchecked_cast< int>(audio_frame->samples_per_channel_ * 100)); |
| 448 | if (muted != nullptr) { |
| 449 | *muted = audio_frame->muted(); |
| 450 | } |
| 451 | audio_frame->speech_type_ = ToSpeechType(LastOutputType()); |
| 452 | last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_; |
| 453 | RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||(last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 456, "last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 454 | last_output_sample_rate_hz_ == 16000 ||(last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 456, "last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 455 | last_output_sample_rate_hz_ == 32000 ||(last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 456, "last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 456 | last_output_sample_rate_hz_ == 48000)(last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 456, "last_output_sample_rate_hz_ == 8000 || last_output_sample_rate_hz_ == 16000 || last_output_sample_rate_hz_ == 32000 || last_output_sample_rate_hz_ == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 457 | << "Unexpected sample rate " << last_output_sample_rate_hz_; |
| 458 | |
| 459 | if (current_sample_rate_hz) { |
| 460 | *current_sample_rate_hz = last_output_sample_rate_hz_; |
| 461 | } |
| 462 | |
| 463 | return kOK; |
| 464 | } |
| 465 | |
| 466 | void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) { |
| 467 | const std::vector<int> changed_payload_types = |
| 468 | decoder_database_->SetCodecs(codecs); |
| 469 | for (const int pt : changed_payload_types) { |
| 470 | packet_buffer_->DiscardPacketsWithPayloadType(pt); |
| 471 | } |
| 472 | } |
| 473 | |
| 474 | bool NetEqImpl::RegisterPayloadType(int rtp_payload_type, |
| 475 | const SdpAudioFormat& audio_format) { |
| 476 | RTC_LOG(LS_VERBOSE)!::webrtc::LogMessage::IsNoop<::webrtc::LS_VERBOSE>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 476, ::webrtc::LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type " |
| 477 | << rtp_payload_type << ", codec " |
| 478 | << absl::StrCat(audio_format); |
| 479 | return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) == |
| 480 | DecoderDatabase::kOK; |
| 481 | } |
| 482 | |
| 483 | bool NetEqImpl::CreateDecoder(int rtp_payload_type) { |
| 484 | return decoder_database_->GetDecoder(rtp_payload_type) != nullptr; |
| 485 | } |
| 486 | |
| 487 | int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) { |
| 488 | int ret = decoder_database_->Remove(rtp_payload_type); |
| 489 | if (ret == DecoderDatabase::kOK || ret == DecoderDatabase::kDecoderNotFound) { |
| 490 | packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type); |
| 491 | return kOK; |
| 492 | } |
| 493 | return kFail; |
| 494 | } |
| 495 | |
| 496 | void NetEqImpl::RemoveAllPayloadTypes() { |
| 497 | decoder_database_->RemoveAll(); |
| 498 | } |
| 499 | |
| 500 | bool NetEqImpl::SetMinimumDelay(int delay_ms) { |
| 501 | if (delay_ms >= 0 && delay_ms <= 10000) { |
| 502 | RTC_DCHECK(controller_.get())(controller_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 502, "controller_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 503 | return controller_->SetMinimumDelay(delay_ms); |
| 504 | } |
| 505 | return false; |
| 506 | } |
| 507 | |
| 508 | bool NetEqImpl::SetMaximumDelay(int delay_ms) { |
| 509 | if (delay_ms >= 0 && delay_ms <= 10000) { |
| 510 | RTC_DCHECK(controller_.get())(controller_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 510, "controller_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 511 | return controller_->SetMaximumDelay(delay_ms); |
| 512 | } |
| 513 | return false; |
| 514 | } |
| 515 | |
| 516 | bool NetEqImpl::SetBaseMinimumDelayMs(int delay_ms) { |
| 517 | if (delay_ms >= 0 && delay_ms <= 10000) { |
| 518 | return controller_->SetBaseMinimumDelay(delay_ms); |
| 519 | } |
| 520 | return false; |
| 521 | } |
| 522 | |
| 523 | int NetEqImpl::GetBaseMinimumDelayMs() const { |
| 524 | return controller_->GetBaseMinimumDelay(); |
| 525 | } |
| 526 | |
| 527 | int NetEqImpl::TargetDelayMs() const { |
| 528 | RTC_DCHECK(controller_.get())(controller_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 528, "controller_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 529 | return controller_->TargetLevelMs(); |
| 530 | } |
| 531 | |
| 532 | int NetEqImpl::FilteredCurrentDelayMs() const { |
| 533 | // Sum up the filtered packet buffer level with the future length of the sync |
| 534 | // buffer. |
| 535 | const int delay_samples = |
| 536 | controller_->GetFilteredBufferLevel() + sync_buffer_->FutureLength(); |
| 537 | // The division below will truncate. The return value is in ms. |
| 538 | return delay_samples / CheckedDivExact(fs_hz_, 1000); |
| 539 | } |
| 540 | |
| 541 | int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) { |
| 542 | RTC_DCHECK(decoder_database_.get())(decoder_database_.get()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 542, "decoder_database_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 543 | *stats = CurrentNetworkStatisticsInternal(); |
| 544 | stats_->GetNetworkStatistics(decoder_frame_length_, stats); |
| 545 | return 0; |
| 546 | } |
| 547 | |
| 548 | NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatistics() const { |
| 549 | return CurrentNetworkStatisticsInternal(); |
| 550 | } |
| 551 | |
| 552 | NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatisticsInternal() const { |
| 553 | RTC_DCHECK(decoder_database_.get())(decoder_database_.get()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 553, "decoder_database_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 554 | NetEqNetworkStatistics stats; |
| 555 | const size_t total_samples_in_buffers = |
| 556 | packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) + |
| 557 | sync_buffer_->FutureLength(); |
| 558 | |
| 559 | RTC_DCHECK(controller_.get())(controller_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 559, "controller_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 560 | stats.preferred_buffer_size_ms = controller_->TargetLevelMs(); |
| 561 | stats.jitter_peaks_found = controller_->PeakFound(); |
| 562 | RTC_DCHECK_GT(fs_hz_, 0)::webrtc::SafeGt((fs_hz_), (0)) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 562, "fs_hz_" " " ">" " " "0") & ::webrtc::webrtc_checks_impl ::LogStreamer<>() << (fs_hz_) << (0); |
| 563 | stats.current_buffer_size_ms = |
| 564 | static_cast<uint16_t>(total_samples_in_buffers * 1000 / fs_hz_); |
| 565 | return stats; |
| 566 | } |
| 567 | |
| 568 | NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const { |
| 569 | return stats_->GetLifetimeStatistics(); |
| 570 | } |
| 571 | |
| 572 | NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const { |
| 573 | auto result = stats_->GetOperationsAndState(); |
| 574 | result.current_buffer_size_ms = |
| 575 | (packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) + |
| 576 | sync_buffer_->FutureLength()) * |
| 577 | 1000 / fs_hz_; |
| 578 | result.current_frame_size_ms = decoder_frame_length_ * 1000 / fs_hz_; |
| 579 | result.next_packet_available = packet_buffer_->PeekNextPacket() && |
| 580 | packet_buffer_->PeekNextPacket()->timestamp == |
| 581 | sync_buffer_->end_timestamp(); |
| 582 | return result; |
| 583 | } |
| 584 | |
| 585 | std::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const { |
| 586 | if (first_packet_ || last_mode_ == Mode::kRfc3389Cng || |
| 587 | last_mode_ == Mode::kCodecInternalCng) { |
| 588 | // We don't have a valid RTP timestamp until we have decoded our first |
| 589 | // RTP packet. Also, the RTP timestamp is not accurate while playing CNG, |
| 590 | // which is indicated by returning an empty value. |
| 591 | return std::nullopt; |
| 592 | } |
| 593 | return timestamp_scaler_->ToExternal(playout_timestamp_); |
| 594 | } |
| 595 | |
| 596 | int NetEqImpl::last_output_sample_rate_hz() const { |
| 597 | return last_output_sample_rate_hz_; |
| 598 | } |
| 599 | |
| 600 | std::optional<NetEq::DecoderFormat> NetEqImpl::GetCurrentDecoderFormat() const { |
| 601 | if (!current_rtp_payload_type_.has_value()) { |
| 602 | return std::nullopt; |
| 603 | } |
| 604 | const DecoderDatabase::DecoderInfo* di = |
| 605 | decoder_database_->GetDecoderInfo(*current_rtp_payload_type_); |
| 606 | if (di == nullptr) { |
| 607 | return std::nullopt; |
| 608 | } |
| 609 | return DecoderFormat{ |
| 610 | .payload_type = *current_rtp_payload_type_, |
| 611 | .sample_rate_hz = di->SampleRateHz(), |
| 612 | .num_channels = dchecked_cast<int>(di->GetDecoder()->Channels()), |
| 613 | .sdp_format = di->GetFormat()}; |
| 614 | } |
| 615 | |
| 616 | void NetEqImpl::FlushBuffers() { |
| 617 | RTC_LOG(LS_VERBOSE)!::webrtc::LogMessage::IsNoop<::webrtc::LS_VERBOSE>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 617, ::webrtc::LS_VERBOSE) << "FlushBuffers"; |
| 618 | packet_buffer_->Flush(); |
| 619 | RTC_DCHECK(sync_buffer_.get())(sync_buffer_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 619, "sync_buffer_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 620 | RTC_DCHECK(expand_.get())(expand_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 620, "expand_.get()") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 621 | sync_buffer_->Flush(); |
| 622 | sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| 623 | expand_->overlap_length()); |
| 624 | // Set to wait for new codec. |
| 625 | first_packet_ = true; |
| 626 | } |
| 627 | |
| 628 | int NetEqImpl::SyncBufferSizeMs() const { |
| 629 | return dchecked_cast<int>(sync_buffer_->FutureLength() / |
| 630 | CheckedDivExact(fs_hz_, 1000)); |
| 631 | } |
| 632 | |
| 633 | const SyncBuffer* NetEqImpl::sync_buffer_for_test() const { |
| 634 | return sync_buffer_.get(); |
| 635 | } |
| 636 | |
| 637 | NetEq::Operation NetEqImpl::last_operation_for_test() const { |
| 638 | return last_operation_; |
| 639 | } |
| 640 | |
| 641 | // Methods below this line are private. |
| 642 | |
| 643 | bool NetEqImpl::MaybeChangePayloadType(uint8_t payload_type) { |
| 644 | bool changed = false; |
| 645 | if (decoder_database_->IsComfortNoise(payload_type)) { |
| 646 | if (current_cng_rtp_payload_type_ && |
| 647 | *current_cng_rtp_payload_type_ != payload_type) { |
| 648 | // New CNG payload type implies new codec type. |
| 649 | current_rtp_payload_type_ = std::nullopt; |
| 650 | changed = true; |
| 651 | } |
| 652 | current_cng_rtp_payload_type_ = payload_type; |
| 653 | } else if (!decoder_database_->IsDtmf(payload_type)) { |
| 654 | // This must be speech. |
| 655 | if ((current_rtp_payload_type_ && |
| 656 | *current_rtp_payload_type_ != payload_type) || |
| 657 | (current_cng_rtp_payload_type_ && |
| 658 | !EqualSampleRates(payload_type, *current_cng_rtp_payload_type_, |
| 659 | *decoder_database_))) { |
| 660 | current_cng_rtp_payload_type_ = std::nullopt; |
| 661 | changed = true; |
| 662 | } |
| 663 | current_rtp_payload_type_ = payload_type; |
| 664 | } |
| 665 | return changed; |
| 666 | } |
| 667 | |
| 668 | int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, |
| 669 | std::optional<Operation> action_override) { |
| 670 | PacketList packet_list; |
| 671 | DtmfEvent dtmf_event; |
| 672 | Operation operation; |
| 673 | bool play_dtmf; |
| 674 | last_decoded_packet_infos_.clear(); |
| 675 | tick_timer_->Increment(); |
| 676 | |
| 677 | // Sanity check - should already be taken care of when setting |
| 678 | // output_size_samples_. |
| 679 | RTC_DCHECK_LE(output_size_samples_ * sync_buffer_->Channels(),::webrtc::SafeLe((output_size_samples_ * sync_buffer_->Channels ()), (AudioFrame::kMaxDataSizeSamples)) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 680, "output_size_samples_ * sync_buffer_->Channels()" " " "<=" " " "AudioFrame::kMaxDataSizeSamples") & ::webrtc ::webrtc_checks_impl::LogStreamer<>() << (output_size_samples_ * sync_buffer_->Channels()) << (AudioFrame::kMaxDataSizeSamples ) |
| 680 | AudioFrame::kMaxDataSizeSamples)::webrtc::SafeLe((output_size_samples_ * sync_buffer_->Channels ()), (AudioFrame::kMaxDataSizeSamples)) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 680, "output_size_samples_ * sync_buffer_->Channels()" " " "<=" " " "AudioFrame::kMaxDataSizeSamples") & ::webrtc ::webrtc_checks_impl::LogStreamer<>() << (output_size_samples_ * sync_buffer_->Channels()) << (AudioFrame::kMaxDataSizeSamples ); |
| 681 | |
| 682 | // Check for muted state. |
| 683 | if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) { |
| 684 | RTC_DCHECK_EQ(last_mode_, Mode::kExpand)::webrtc::SafeEq((last_mode_), (Mode::kExpand)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 684, "last_mode_" " " "==" " " "Mode::kExpand") & ::webrtc ::webrtc_checks_impl::LogStreamer<>() << (last_mode_ ) << (Mode::kExpand); |
| 685 | audio_frame->Reset(); |
| 686 | RTC_DCHECK(audio_frame->muted())(audio_frame->muted()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 686, "audio_frame->muted()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); // Reset() should mute the frame. |
| 687 | playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); |
| 688 | audio_frame->sample_rate_hz_ = fs_hz_; |
| 689 | audio_frame->samples_per_channel_ = output_size_samples_; |
| 690 | audio_frame->timestamp_ = |
| 691 | first_packet_ |
| 692 | ? 0 |
| 693 | : timestamp_scaler_->ToExternal(playout_timestamp_) - |
| 694 | static_cast<uint32_t>(audio_frame->samples_per_channel_); |
| 695 | audio_frame->num_channels_ = sync_buffer_->Channels(); |
| 696 | stats_->ExpandedNoiseSamples(output_size_samples_, false); |
| 697 | controller_->NotifyMutedState(); |
| 698 | return 0; |
| 699 | } |
| 700 | int return_value = GetDecision(&operation, &packet_list, &dtmf_event, |
| 701 | &play_dtmf, action_override); |
| 702 | if (return_value != 0) { |
| 703 | last_mode_ = Mode::kError; |
| 704 | return return_value; |
| 705 | } |
| 706 | |
| 707 | AudioDecoder::SpeechType speech_type; |
| 708 | int length = 0; |
| 709 | const size_t start_num_packets = packet_list.size(); |
| 710 | int decode_return_value = |
| 711 | Decode(&packet_list, &operation, &length, &speech_type); |
| 712 | if (length > 0) { |
| 713 | last_decoded_type_ = speech_type; |
| 714 | } |
| 715 | |
| 716 | bool sid_frame_available = |
| 717 | (operation == Operation::kRfc3389Cng && !packet_list.empty()); |
| 718 | |
| 719 | // This is the criterion that we did decode some data through the speech |
| 720 | // decoder, and the operation resulted in comfort noise. |
| 721 | const bool codec_internal_sid_frame = |
| 722 | (speech_type == AudioDecoder::kComfortNoise && |
| 723 | start_num_packets > packet_list.size()); |
| 724 | |
| 725 | if (sid_frame_available || codec_internal_sid_frame) { |
| 726 | // Start a new stopwatch since we are decoding a new CNG packet. |
| 727 | generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch(); |
| 728 | } |
| 729 | |
| 730 | algorithm_buffer_->Clear(); |
| 731 | switch (operation) { |
| 732 | case Operation::kNormal: { |
| 733 | DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf); |
| 734 | if (length > 0) { |
| 735 | stats_->DecodedOutputPlayed(); |
| 736 | } |
| 737 | break; |
| 738 | } |
| 739 | case Operation::kMerge: { |
| 740 | DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf); |
| 741 | break; |
| 742 | } |
| 743 | case Operation::kExpand: { |
| 744 | RTC_DCHECK_EQ(return_value, 0)::webrtc::SafeEq((return_value), (0)) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 744, "return_value" " " "==" " " "0") & ::webrtc::webrtc_checks_impl ::LogStreamer<>() << (return_value) << (0); |
| 745 | if (!current_rtp_payload_type_ || !DoCodecPlc()) { |
| 746 | return_value = DoExpand(play_dtmf); |
| 747 | } |
| 748 | RTC_DCHECK_GE(sync_buffer_->FutureLength() - expand_->overlap_length(),::webrtc::SafeGe((sync_buffer_->FutureLength() - expand_-> overlap_length()), (output_size_samples_)) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 749, "sync_buffer_->FutureLength() - expand_->overlap_length()" " " ">=" " " "output_size_samples_") & ::webrtc::webrtc_checks_impl ::LogStreamer<>() << (sync_buffer_->FutureLength () - expand_->overlap_length()) << (output_size_samples_ ) |
| 749 | output_size_samples_)::webrtc::SafeGe((sync_buffer_->FutureLength() - expand_-> overlap_length()), (output_size_samples_)) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 749, "sync_buffer_->FutureLength() - expand_->overlap_length()" " " ">=" " " "output_size_samples_") & ::webrtc::webrtc_checks_impl ::LogStreamer<>() << (sync_buffer_->FutureLength () - expand_->overlap_length()) << (output_size_samples_ ); |
| 750 | break; |
| 751 | } |
| 752 | case Operation::kAccelerate: |
| 753 | case Operation::kFastAccelerate: { |
| 754 | const bool fast_accelerate = |
| 755 | enable_fast_accelerate_ && (operation == Operation::kFastAccelerate); |
| 756 | return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type, |
| 757 | play_dtmf, fast_accelerate); |
| 758 | break; |
| 759 | } |
| 760 | case Operation::kPreemptiveExpand: { |
| 761 | return_value = DoPreemptiveExpand(decoded_buffer_.get(), length, |
| 762 | speech_type, play_dtmf); |
| 763 | break; |
| 764 | } |
| 765 | case Operation::kRfc3389Cng: |
| 766 | case Operation::kRfc3389CngNoPacket: { |
| 767 | return_value = DoRfc3389Cng(&packet_list, play_dtmf); |
| 768 | break; |
| 769 | } |
| 770 | case Operation::kCodecInternalCng: { |
| 771 | // This handles the case when there is no transmission and the decoder |
| 772 | // should produce internal comfort noise. |
| 773 | // TODO(hlundin): Write test for codec-internal CNG. |
| 774 | DoCodecInternalCng(decoded_buffer_.get(), length); |
| 775 | break; |
| 776 | } |
| 777 | case Operation::kDtmf: { |
| 778 | // TODO(hlundin): Write test for this. |
| 779 | return_value = DoDtmf(dtmf_event, &play_dtmf); |
| 780 | break; |
| 781 | } |
| 782 | case Operation::kUndefined: { |
| 783 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 783, ::webrtc::LS_ERROR) << "Invalid operation kUndefined."; |
| 784 | RTC_DCHECK_NOTREACHED()(false) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 784, "false") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); // This should not happen. |
| 785 | last_mode_ = Mode::kError; |
| 786 | return kInvalidOperation; |
| 787 | } |
| 788 | } // End of switch. |
| 789 | last_operation_ = operation; |
| 790 | if (return_value < 0) { |
| 791 | return return_value; |
| 792 | } |
| 793 | |
| 794 | if (last_mode_ != Mode::kRfc3389Cng) { |
| 795 | comfort_noise_->Reset(); |
| 796 | } |
| 797 | |
| 798 | // We treat it as if all packets referenced to by `last_decoded_packet_infos_` |
| 799 | // were mashed together when creating the samples in `algorithm_buffer_`. |
| 800 | RtpPacketInfos packet_infos(last_decoded_packet_infos_); |
| 801 | |
| 802 | // Copy samples from `algorithm_buffer_` to `sync_buffer_`. |
| 803 | // |
| 804 | // TODO(bugs.webrtc.org/10757): |
| 805 | // We would in the future also like to pass `packet_infos` so that we can do |
| 806 | // sample-perfect tracking of that information across `sync_buffer_`. |
| 807 | sync_buffer_->PushBack(*algorithm_buffer_); |
| 808 | |
| 809 | // Extract data from `sync_buffer_` to `output`. |
| 810 | audio_frame->ResetWithoutMuting(); |
| 811 | audio_frame->SetSampleRateAndChannelSize(fs_hz_); |
| 812 | InterleavedView<int16_t> view = |
| 813 | audio_frame->mutable_data(output_size_samples_, sync_buffer_->Channels()); |
| 814 | bool got_audio = sync_buffer_->GetNextAudioInterleaved(view); |
| 815 | |
| 816 | // TODO(bugs.webrtc.org/10757): |
| 817 | // We don't have the ability to properly track individual packets once their |
| 818 | // audio samples have entered `sync_buffer_`. So for now, treat it as if |
| 819 | // `packet_infos` from packets decoded by the current `GetAudioInternal()` |
| 820 | // call were all consumed assembling the current audio frame and the current |
| 821 | // audio frame only. |
| 822 | audio_frame->packet_infos_ = std::move(packet_infos); |
| 823 | if (sync_buffer_->FutureLength() < expand_->overlap_length()) { |
| 824 | // The sync buffer should always contain `overlap_length` samples, but now |
| 825 | // too many samples have been extracted. Reinstall the `overlap_length` |
| 826 | // lookahead by moving the index. |
| 827 | const size_t missing_lookahead_samples = |
| 828 | expand_->overlap_length() - sync_buffer_->FutureLength(); |
| 829 | RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples)::webrtc::SafeGe((sync_buffer_->next_index()), (missing_lookahead_samples )) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 829, "sync_buffer_->next_index()" " " ">=" " " "missing_lookahead_samples" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (sync_buffer_->next_index()) << (missing_lookahead_samples ); |
| 830 | sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| 831 | missing_lookahead_samples); |
| 832 | } |
| 833 | |
| 834 | if (!got_audio) { |
| 835 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 835, ::webrtc::LS_ERROR) << "audio_frame->samples_per_channel_ (" |
| 836 | << audio_frame->samples_per_channel_ |
| 837 | << ") != output_size_samples_ (" << output_size_samples_ |
| 838 | << ")"; |
| 839 | // TODO(minyue): treatment of under-run, filling zeros |
| 840 | audio_frame->Mute(); |
| 841 | return kSampleUnderrun; |
| 842 | } |
| 843 | |
| 844 | // Should always have overlap samples left in the `sync_buffer_`. |
| 845 | RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length())::webrtc::SafeGe((sync_buffer_->FutureLength()), (expand_-> overlap_length())) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 845, "sync_buffer_->FutureLength()" " " ">=" " " "expand_->overlap_length()" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (sync_buffer_->FutureLength()) << (expand_->overlap_length ()); |
| 846 | |
| 847 | // TODO(yujo): For muted frames, this can be a copy rather than an addition. |
| 848 | if (play_dtmf) { |
| 849 | return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), |
| 850 | audio_frame->mutable_data()); |
| 851 | } |
| 852 | |
| 853 | // Update the background noise parameters if last operation wrote data |
| 854 | // straight from the decoder to the `sync_buffer_`. That is, none of the |
| 855 | // operations that modify the signal can be followed by a parameter update. |
| 856 | if ((last_mode_ == Mode::kNormal) || (last_mode_ == Mode::kAccelerateFail) || |
| 857 | (last_mode_ == Mode::kPreemptiveExpandFail) || |
| 858 | (last_mode_ == Mode::kRfc3389Cng) || |
| 859 | (last_mode_ == Mode::kCodecInternalCng)) { |
| 860 | background_noise_->Update(*sync_buffer_); |
| 861 | } |
| 862 | |
| 863 | if (operation == Operation::kDtmf) { |
| 864 | // DTMF data was written the end of `sync_buffer_`. |
| 865 | // Update index to end of DTMF data in `sync_buffer_`. |
| 866 | sync_buffer_->set_dtmf_index(sync_buffer_->Size()); |
| 867 | } |
| 868 | |
| 869 | if (last_mode_ != Mode::kExpand && last_mode_ != Mode::kCodecPlc) { |
| 870 | // If last operation was not expand, calculate the `playout_timestamp_` from |
| 871 | // the `sync_buffer_`. However, do not update the `playout_timestamp_` if it |
| 872 | // would be moved "backwards". |
| 873 | uint32_t temp_timestamp = |
| 874 | sync_buffer_->end_timestamp() - |
| 875 | static_cast<uint32_t>(sync_buffer_->FutureLength()); |
| 876 | if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) { |
| 877 | playout_timestamp_ = temp_timestamp; |
| 878 | } |
| 879 | } else { |
| 880 | // Use dead reckoning to estimate the `playout_timestamp_`. |
| 881 | playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); |
| 882 | } |
| 883 | // Set the timestamp in the audio frame to zero before the first packet has |
| 884 | // been inserted. Otherwise, subtract the frame size in samples to get the |
| 885 | // timestamp of the first sample in the frame (playout_timestamp_ is the |
| 886 | // last + 1). |
| 887 | audio_frame->timestamp_ = |
| 888 | first_packet_ |
| 889 | ? 0 |
| 890 | : timestamp_scaler_->ToExternal(playout_timestamp_) - |
| 891 | static_cast<uint32_t>(audio_frame->samples_per_channel_); |
| 892 | |
| 893 | if (!(last_mode_ == Mode::kRfc3389Cng || |
| 894 | last_mode_ == Mode::kCodecInternalCng || last_mode_ == Mode::kExpand || |
| 895 | last_mode_ == Mode::kCodecPlc)) { |
| 896 | generated_noise_stopwatch_.reset(); |
| 897 | } |
| 898 | |
| 899 | if (decode_return_value) |
| 900 | return decode_return_value; |
| 901 | return return_value; |
| 902 | } |
| 903 | |
| 904 | int NetEqImpl::GetDecision(Operation* operation, |
| 905 | PacketList* packet_list, |
| 906 | DtmfEvent* dtmf_event, |
| 907 | bool* play_dtmf, |
| 908 | std::optional<Operation> action_override) { |
| 909 | // Initialize output variables. |
| 910 | *play_dtmf = false; |
| 911 | *operation = Operation::kUndefined; |
| 912 | |
| 913 | RTC_DCHECK(sync_buffer_.get())(sync_buffer_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 913, "sync_buffer_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 914 | uint32_t end_timestamp = sync_buffer_->end_timestamp(); |
| 915 | if (!new_codec_) { |
| 916 | const uint32_t five_seconds_samples = 5 * fs_hz_; |
| 917 | packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples); |
| 918 | } |
| 919 | const Packet* packet = packet_buffer_->PeekNextPacket(); |
| 920 | |
| 921 | RTC_DCHECK(!generated_noise_stopwatch_ ||(!generated_noise_stopwatch_ || generated_noise_stopwatch_-> ElapsedTicks() >= 1) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 922, "!generated_noise_stopwatch_ || generated_noise_stopwatch_->ElapsedTicks() >= 1" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 922 | generated_noise_stopwatch_->ElapsedTicks() >= 1)(!generated_noise_stopwatch_ || generated_noise_stopwatch_-> ElapsedTicks() >= 1) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 922, "!generated_noise_stopwatch_ || generated_noise_stopwatch_->ElapsedTicks() >= 1" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>(); |
| 923 | uint64_t generated_noise_samples = |
| 924 | generated_noise_stopwatch_ ? (generated_noise_stopwatch_->ElapsedTicks() - |
| 925 | 1) * output_size_samples_ + |
| 926 | controller_->noise_fast_forward() |
| 927 | : 0; |
| 928 | |
| 929 | if (last_mode_ == Mode::kRfc3389Cng) { |
| 930 | // Because of timestamp peculiarities, we have to "manually" disallow using |
| 931 | // a CNG packet with the same timestamp as the one that was last played. |
| 932 | // This can happen when using redundancy and will cause the timing to shift. |
| 933 | while (packet && decoder_database_->IsComfortNoise(packet->payload_type) && |
| 934 | (end_timestamp >= packet->timestamp || |
| 935 | end_timestamp + generated_noise_samples > packet->timestamp)) { |
| 936 | // Don't use this packet, discard it. |
| 937 | if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) { |
| 938 | RTC_DCHECK_NOTREACHED()(false) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 938, "false") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); // Must be ok by design. |
| 939 | } |
| 940 | // Check buffer again. |
| 941 | if (!new_codec_) { |
| 942 | packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_); |
| 943 | } |
| 944 | packet = packet_buffer_->PeekNextPacket(); |
| 945 | } |
| 946 | } |
| 947 | |
| 948 | RTC_DCHECK(expand_.get())(expand_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 948, "expand_.get()") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 949 | const int samples_left = static_cast<int>(sync_buffer_->FutureLength() - |
| 950 | expand_->overlap_length()); |
| 951 | if (last_mode_ == Mode::kAccelerateSuccess || |
| 952 | last_mode_ == Mode::kAccelerateLowEnergy || |
| 953 | last_mode_ == Mode::kPreemptiveExpandSuccess || |
| 954 | last_mode_ == Mode::kPreemptiveExpandLowEnergy) { |
| 955 | // Subtract (samples_left + output_size_samples_) from sampleMemory. |
| 956 | controller_->AddSampleMemory( |
| 957 | -(samples_left + dchecked_cast<int>(output_size_samples_))); |
| 958 | } |
| 959 | |
| 960 | // Check if it is time to play a DTMF event. |
| 961 | if (dtmf_buffer_->GetEvent( |
| 962 | static_cast<uint32_t>(end_timestamp + generated_noise_samples), |
| 963 | dtmf_event)) { |
| 964 | *play_dtmf = true; |
| 965 | } |
| 966 | |
| 967 | // Get instruction. |
| 968 | RTC_DCHECK(sync_buffer_.get())(sync_buffer_.get()) ? static_cast<void>(0) : ::webrtc:: webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 968, "sync_buffer_.get()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 969 | RTC_DCHECK(expand_.get())(expand_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 969, "expand_.get()") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 970 | generated_noise_samples = |
| 971 | generated_noise_stopwatch_ |
| 972 | ? generated_noise_stopwatch_->ElapsedTicks() * output_size_samples_ + |
| 973 | controller_->noise_fast_forward() |
| 974 | : 0; |
| 975 | NetEqController::NetEqStatus status; |
| 976 | status.packet_buffer_info.dtx_or_cng = |
| 977 | packet_buffer_->ContainsDtxOrCngPacket(decoder_database_.get()); |
| 978 | status.packet_buffer_info.num_samples = |
| 979 | packet_buffer_->NumSamplesInBuffer(decoder_frame_length_); |
| 980 | status.packet_buffer_info.span_samples = packet_buffer_->GetSpanSamples( |
| 981 | decoder_frame_length_, last_output_sample_rate_hz_, false); |
| 982 | status.packet_buffer_info.span_samples_wait_time = |
| 983 | packet_buffer_->GetSpanSamples(decoder_frame_length_, |
| 984 | last_output_sample_rate_hz_, true); |
| 985 | status.packet_buffer_info.num_packets = packet_buffer_->NumPacketsInBuffer(); |
| 986 | status.target_timestamp = sync_buffer_->end_timestamp(); |
| 987 | status.expand_mutefactor = expand_->MuteFactor(0); |
| 988 | status.last_packet_samples = decoder_frame_length_; |
| 989 | status.last_mode = last_mode_; |
| 990 | status.play_dtmf = *play_dtmf; |
| 991 | status.generated_noise_samples = generated_noise_samples; |
| 992 | status.sync_buffer_samples = sync_buffer_->FutureLength(); |
| 993 | if (packet) { |
| 994 | status.next_packet = { |
| 995 | .timestamp = packet->timestamp, |
| 996 | .is_dtx = packet->frame && packet->frame->IsDtxPacket(), |
| 997 | .is_cng = decoder_database_->IsComfortNoise(packet->payload_type)}; |
| 998 | } |
| 999 | *operation = controller_->GetDecision(status, &reset_decoder_); |
| 1000 | |
| 1001 | // Disallow time stretching if this packet is DTX, because such a decision may |
| 1002 | // be based on earlier buffer level estimate, as we do not update buffer level |
| 1003 | // during DTX. When we have a better way to update buffer level during DTX, |
| 1004 | // this can be discarded. |
| 1005 | if (packet && packet->frame && packet->frame->IsDtxPacket() && |
| 1006 | (*operation == Operation::kMerge || |
| 1007 | *operation == Operation::kAccelerate || |
| 1008 | *operation == Operation::kFastAccelerate || |
| 1009 | *operation == Operation::kPreemptiveExpand)) { |
| 1010 | *operation = Operation::kNormal; |
| 1011 | } |
| 1012 | |
| 1013 | if (action_override) { |
| 1014 | // Use the provided action instead of the decision NetEq decided on. |
| 1015 | *operation = *action_override; |
| 1016 | } |
| 1017 | // Check if we already have enough samples in the `sync_buffer_`. If so, |
| 1018 | // change decision to normal, unless the decision was merge, accelerate, or |
| 1019 | // preemptive expand. |
| 1020 | if (samples_left >= dchecked_cast<int>(output_size_samples_) && |
| 1021 | *operation != Operation::kMerge && *operation != Operation::kAccelerate && |
| 1022 | *operation != Operation::kFastAccelerate && |
| 1023 | *operation != Operation::kPreemptiveExpand) { |
| 1024 | *operation = Operation::kNormal; |
| 1025 | return 0; |
| 1026 | } |
| 1027 | |
| 1028 | controller_->ExpandDecision(*operation); |
| 1029 | if ((last_mode_ == Mode::kCodecPlc) && (*operation != Operation::kExpand)) { |
| 1030 | // Getting out of the PLC expand mode, reporting interruptions. |
| 1031 | // NetEq PLC reports this metrics in expand.cc |
| 1032 | stats_->EndExpandEvent(fs_hz_); |
| 1033 | } |
| 1034 | |
| 1035 | // Check conditions for reset. |
| 1036 | if (new_codec_ || *operation == Operation::kUndefined) { |
| 1037 | // The only valid reason to get kUndefined is that new_codec_ is set. |
| 1038 | RTC_DCHECK(new_codec_)(new_codec_) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1038, "new_codec_") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1039 | if (*play_dtmf && !packet) { |
| 1040 | timestamp_ = dtmf_event->timestamp; |
| 1041 | } else { |
| 1042 | if (!packet) { |
| 1043 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1043, ::webrtc::LS_ERROR) << "Packet missing where it shouldn't."; |
| 1044 | return -1; |
| 1045 | } |
| 1046 | timestamp_ = packet->timestamp; |
| 1047 | if (*operation == Operation::kRfc3389CngNoPacket && |
| 1048 | decoder_database_->IsComfortNoise(packet->payload_type)) { |
| 1049 | // Change decision to CNG packet, since we do have a CNG packet, but it |
| 1050 | // was considered too early to use. Now, use it anyway. |
| 1051 | *operation = Operation::kRfc3389Cng; |
| 1052 | } else if (*operation != Operation::kRfc3389Cng) { |
| 1053 | *operation = Operation::kNormal; |
| 1054 | } |
| 1055 | } |
| 1056 | // Adjust `sync_buffer_` timestamp before setting `end_timestamp` to the |
| 1057 | // new value. |
| 1058 | sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp); |
| 1059 | end_timestamp = timestamp_; |
| 1060 | new_codec_ = false; |
| 1061 | controller_->SoftReset(); |
| 1062 | stats_->ResetMcu(); |
| 1063 | } |
| 1064 | |
| 1065 | size_t required_samples = output_size_samples_; |
| 1066 | const size_t samples_10_ms = static_cast<size_t>(80 * fs_mult_); |
| 1067 | const size_t samples_20_ms = 2 * samples_10_ms; |
| 1068 | const size_t samples_30_ms = 3 * samples_10_ms; |
| 1069 | |
| 1070 | switch (*operation) { |
| 1071 | case Operation::kExpand: { |
| 1072 | timestamp_ = end_timestamp; |
| 1073 | return 0; |
| 1074 | } |
| 1075 | case Operation::kRfc3389CngNoPacket: |
| 1076 | case Operation::kCodecInternalCng: { |
| 1077 | return 0; |
| 1078 | } |
| 1079 | case Operation::kDtmf: { |
| 1080 | // TODO(hlundin): Write test for this. |
| 1081 | // Update timestamp. |
| 1082 | timestamp_ = end_timestamp; |
| 1083 | generated_noise_samples = |
| 1084 | generated_noise_stopwatch_ |
| 1085 | ? generated_noise_stopwatch_->ElapsedTicks() * |
| 1086 | output_size_samples_ + |
| 1087 | controller_->noise_fast_forward() |
| 1088 | : 0; |
| 1089 | if (generated_noise_samples > 0 && last_mode_ != Mode::kDtmf) { |
| 1090 | // Make a jump in timestamp due to the recently played comfort noise. |
| 1091 | uint32_t timestamp_jump = |
| 1092 | static_cast<uint32_t>(generated_noise_samples); |
| 1093 | sync_buffer_->IncreaseEndTimestamp(timestamp_jump); |
| 1094 | timestamp_ += timestamp_jump; |
| 1095 | } |
| 1096 | return 0; |
| 1097 | } |
| 1098 | case Operation::kAccelerate: |
| 1099 | case Operation::kFastAccelerate: { |
| 1100 | // In order to do an accelerate we need at least 30 ms of audio data. |
| 1101 | if (samples_left >= static_cast<int>(samples_30_ms)) { |
| 1102 | // Already have enough data, so we do not need to extract any more. |
| 1103 | controller_->set_sample_memory(samples_left); |
| 1104 | controller_->set_prev_time_scale(true); |
| 1105 | return 0; |
| 1106 | } else if (samples_left >= static_cast<int>(samples_10_ms) && |
| 1107 | decoder_frame_length_ >= samples_30_ms) { |
| 1108 | // Avoid decoding more data as it might overflow the playout buffer. |
| 1109 | *operation = Operation::kNormal; |
| 1110 | return 0; |
| 1111 | } else if (samples_left < static_cast<int>(samples_20_ms) && |
| 1112 | decoder_frame_length_ < samples_30_ms) { |
| 1113 | // Build up decoded data by decoding at least 20 ms of audio data. Do |
| 1114 | // not perform accelerate yet, but wait until we only need to do one |
| 1115 | // decoding. |
| 1116 | required_samples = 2 * output_size_samples_; |
| 1117 | *operation = Operation::kNormal; |
| 1118 | } |
| 1119 | // If none of the above is true, we have one of two possible situations: |
| 1120 | // (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or |
| 1121 | // (2) samples_left < 10 ms and decoder_frame_length_ >= 30 ms. |
| 1122 | // In either case, we move on with the accelerate decision, and decode one |
| 1123 | // frame now. |
| 1124 | break; |
| 1125 | } |
| 1126 | case Operation::kPreemptiveExpand: { |
| 1127 | // In order to do a preemptive expand we need at least 30 ms of decoded |
| 1128 | // audio data. |
| 1129 | if ((samples_left >= static_cast<int>(samples_30_ms)) || |
| 1130 | (samples_left >= static_cast<int>(samples_10_ms) && |
| 1131 | decoder_frame_length_ >= samples_30_ms)) { |
| 1132 | // Already have enough data, so we do not need to extract any more. |
| 1133 | // Or, avoid decoding more data as it might overflow the playout buffer. |
| 1134 | // Still try preemptive expand, though. |
| 1135 | controller_->set_sample_memory(samples_left); |
| 1136 | controller_->set_prev_time_scale(true); |
| 1137 | return 0; |
| 1138 | } |
| 1139 | if (samples_left < static_cast<int>(samples_20_ms) && |
| 1140 | decoder_frame_length_ < samples_30_ms) { |
| 1141 | // Build up decoded data by decoding at least 20 ms of audio data. |
| 1142 | // Still try to perform preemptive expand. |
| 1143 | required_samples = 2 * output_size_samples_; |
| 1144 | } |
| 1145 | // Move on with the preemptive expand decision. |
| 1146 | break; |
| 1147 | } |
| 1148 | case Operation::kMerge: { |
| 1149 | required_samples = |
| 1150 | std::max(merge_->RequiredFutureSamples(), required_samples); |
| 1151 | break; |
| 1152 | } |
| 1153 | default: { |
| 1154 | // Do nothing. |
| 1155 | } |
| 1156 | } |
| 1157 | |
| 1158 | // Get packets from buffer. |
| 1159 | int extracted_samples = 0; |
| 1160 | if (packet) { |
| 1161 | sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp); |
| 1162 | extracted_samples = ExtractPackets(required_samples, packet_list); |
| 1163 | if (extracted_samples < 0) { |
| 1164 | return kPacketBufferCorruption; |
| 1165 | } |
| 1166 | } |
| 1167 | |
| 1168 | if (*operation == Operation::kAccelerate || |
| 1169 | *operation == Operation::kFastAccelerate || |
| 1170 | *operation == Operation::kPreemptiveExpand) { |
| 1171 | controller_->set_sample_memory(samples_left + extracted_samples); |
| 1172 | controller_->set_prev_time_scale(true); |
| 1173 | } |
| 1174 | |
| 1175 | if (*operation == Operation::kAccelerate || |
| 1176 | *operation == Operation::kFastAccelerate) { |
| 1177 | // Check that we have enough data (30ms) to do accelerate. |
| 1178 | if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) { |
| 1179 | // TODO(hlundin): Write test for this. |
| 1180 | // Not enough, do normal operation instead. |
| 1181 | *operation = Operation::kNormal; |
| 1182 | } |
| 1183 | } |
| 1184 | |
| 1185 | timestamp_ = sync_buffer_->end_timestamp(); |
| 1186 | return 0; |
| 1187 | } |
| 1188 | |
| 1189 | int NetEqImpl::Decode(PacketList* packet_list, |
| 1190 | Operation* operation, |
| 1191 | int* decoded_length, |
| 1192 | AudioDecoder::SpeechType* speech_type) { |
| 1193 | *speech_type = AudioDecoder::kSpeech; |
| 1194 | |
| 1195 | // When packet_list is empty, we may be in kCodecInternalCng mode, and for |
| 1196 | // that we use current active decoder. |
| 1197 | AudioDecoder* decoder = decoder_database_->GetActiveDecoder(); |
| 1198 | |
| 1199 | if (!packet_list->empty()) { |
| 1200 | const Packet& packet = packet_list->front(); |
| 1201 | uint8_t payload_type = packet.payload_type; |
| 1202 | if (!decoder_database_->IsComfortNoise(payload_type)) { |
| 1203 | decoder = decoder_database_->GetDecoder(payload_type); |
| 1204 | RTC_DCHECK(decoder)(decoder) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1204, "decoder") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1205 | if (!decoder) { |
| 1206 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1206, ::webrtc::LS_WARNING) |
| 1207 | << "Unknown payload type " << static_cast<int>(payload_type); |
| 1208 | packet_list->clear(); |
| 1209 | return kDecoderNotFound; |
| 1210 | } |
| 1211 | bool decoder_changed; |
| 1212 | decoder_database_->SetActiveDecoder(payload_type, &decoder_changed); |
| 1213 | if (decoder_changed) { |
| 1214 | // We have a new decoder. Re-init some values. |
| 1215 | const DecoderDatabase::DecoderInfo* decoder_info = |
| 1216 | decoder_database_->GetDecoderInfo(payload_type); |
| 1217 | RTC_DCHECK(decoder_info)(decoder_info) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1217, "decoder_info") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1218 | if (!decoder_info) { |
| 1219 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1219, ::webrtc::LS_WARNING) |
| 1220 | << "Unknown payload type " << static_cast<int>(payload_type); |
| 1221 | packet_list->clear(); |
| 1222 | return kDecoderNotFound; |
| 1223 | } |
| 1224 | // If sampling rate or number of channels has changed, we need to make |
| 1225 | // a reset. |
| 1226 | if (decoder_info->SampleRateHz() != fs_hz_ || |
| 1227 | decoder->Channels() != algorithm_buffer_->Channels()) { |
| 1228 | SetSampleRateAndChannels(decoder_info->SampleRateHz(), |
| 1229 | decoder->Channels()); |
| 1230 | } |
| 1231 | sync_buffer_->set_end_timestamp(timestamp_); |
| 1232 | playout_timestamp_ = timestamp_; |
| 1233 | } |
| 1234 | } |
| 1235 | } |
| 1236 | |
| 1237 | if (reset_decoder_) { |
| 1238 | // TODO(hlundin): Write test for this. |
| 1239 | if (decoder) |
| 1240 | decoder->Reset(); |
| 1241 | |
| 1242 | // Reset comfort noise decoder. |
| 1243 | ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder(); |
| 1244 | if (cng_decoder) |
| 1245 | cng_decoder->Reset(); |
| 1246 | |
| 1247 | reset_decoder_ = false; |
| 1248 | } |
| 1249 | |
| 1250 | *decoded_length = 0; |
| 1251 | // Update codec-internal PLC state. |
| 1252 | if ((*operation == Operation::kMerge) && decoder && decoder->HasDecodePlc()) { |
| 1253 | decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]); |
| 1254 | } |
| 1255 | |
| 1256 | int return_value; |
| 1257 | if (*operation == Operation::kCodecInternalCng) { |
| 1258 | RTC_DCHECK(packet_list->empty())(packet_list->empty()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1258, "packet_list->empty()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 1259 | return_value = DecodeCng(decoder, decoded_length, speech_type); |
| 1260 | } else { |
| 1261 | return_value = DecodeLoop(packet_list, *operation, decoder, decoded_length, |
| 1262 | speech_type); |
| 1263 | } |
| 1264 | |
| 1265 | if (*decoded_length < 0) { |
| 1266 | // Error returned from the decoder. |
| 1267 | *decoded_length = 0; |
| 1268 | sync_buffer_->IncreaseEndTimestamp( |
| 1269 | static_cast<uint32_t>(decoder_frame_length_)); |
| 1270 | int error_code = 0; |
| 1271 | if (decoder) |
| 1272 | error_code = decoder->ErrorCode(); |
| 1273 | if (error_code != 0) { |
| 1274 | // Got some error code from the decoder. |
| 1275 | return_value = kDecoderErrorCode; |
| 1276 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1276, ::webrtc::LS_WARNING) << "Decoder returned error code: " << error_code; |
| 1277 | } else { |
| 1278 | // Decoder does not implement error codes. Return generic error. |
| 1279 | return_value = kOtherDecoderError; |
| 1280 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1280, ::webrtc::LS_WARNING) << "Decoder error (no error code)"; |
| 1281 | } |
| 1282 | *operation = Operation::kExpand; // Do expansion to get data instead. |
| 1283 | } |
| 1284 | if (*speech_type != AudioDecoder::kComfortNoise) { |
| 1285 | // Don't increment timestamp if codec returned CNG speech type |
| 1286 | // since in this case, the we will increment the CNGplayedTS counter. |
| 1287 | // Increase with number of samples per channel. |
| 1288 | RTC_DCHECK(*decoded_length == 0 ||(*decoded_length == 0 || (decoder && decoder->Channels () == sync_buffer_->Channels())) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1289, "*decoded_length == 0 || (decoder && decoder->Channels() == sync_buffer_->Channels())" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1289 | (decoder && decoder->Channels() == sync_buffer_->Channels()))(*decoded_length == 0 || (decoder && decoder->Channels () == sync_buffer_->Channels())) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1289, "*decoded_length == 0 || (decoder && decoder->Channels() == sync_buffer_->Channels())" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>(); |
| 1290 | sync_buffer_->IncreaseEndTimestamp( |
| 1291 | *decoded_length / static_cast<int>(sync_buffer_->Channels())); |
| 1292 | } |
| 1293 | return return_value; |
| 1294 | } |
| 1295 | |
| 1296 | int NetEqImpl::DecodeCng(AudioDecoder* decoder, |
| 1297 | int* decoded_length, |
| 1298 | AudioDecoder::SpeechType* speech_type) { |
| 1299 | if (!decoder) { |
| 1300 | // This happens when active decoder is not defined. |
| 1301 | *decoded_length = -1; |
| 1302 | return 0; |
| 1303 | } |
| 1304 | |
| 1305 | while (*decoded_length < dchecked_cast<int>(output_size_samples_)) { |
| 1306 | const int length = decoder->Decode( |
| 1307 | nullptr, 0, fs_hz_, |
| 1308 | (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), |
| 1309 | &decoded_buffer_[*decoded_length], speech_type); |
| 1310 | if (length > 0) { |
| 1311 | *decoded_length += length; |
| 1312 | } else { |
| 1313 | // Error. |
| 1314 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1314, ::webrtc::LS_WARNING) << "Failed to decode CNG"; |
| 1315 | *decoded_length = -1; |
| 1316 | break; |
| 1317 | } |
| 1318 | if (*decoded_length > static_cast<int>(decoded_buffer_length_)) { |
| 1319 | // Guard against overflow. |
| 1320 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1320, ::webrtc::LS_WARNING) << "Decoded too much CNG."; |
| 1321 | return kDecodedTooMuch; |
| 1322 | } |
| 1323 | } |
| 1324 | stats_->GeneratedNoiseSamples(*decoded_length); |
| 1325 | return 0; |
| 1326 | } |
| 1327 | |
| 1328 | int NetEqImpl::DecodeLoop(PacketList* packet_list, |
| 1329 | const Operation& operation, |
| 1330 | AudioDecoder* decoder, |
| 1331 | int* decoded_length, |
| 1332 | AudioDecoder::SpeechType* speech_type) { |
| 1333 | RTC_DCHECK(last_decoded_packet_infos_.empty())(last_decoded_packet_infos_.empty()) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false> ( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1333, "last_decoded_packet_infos_.empty()") & ::webrtc:: webrtc_checks_impl::LogStreamer<>(); |
| 1334 | |
| 1335 | // Do decoding. |
| 1336 | while (!packet_list->empty() && !decoder_database_->IsComfortNoise( |
| 1337 | packet_list->front().payload_type)) { |
| 1338 | RTC_DCHECK(decoder)(decoder) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1338, "decoder") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); // At this point, we must have a decoder object. |
| 1339 | // The number of channels in the `sync_buffer_` should be the same as the |
| 1340 | // number decoder channels. |
| 1341 | RTC_DCHECK_EQ(sync_buffer_->Channels(), decoder->Channels())::webrtc::SafeEq((sync_buffer_->Channels()), (decoder-> Channels())) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1341, "sync_buffer_->Channels()" " " "==" " " "decoder->Channels()" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (sync_buffer_->Channels()) << (decoder->Channels ()); |
| 1342 | RTC_DCHECK_GE(decoded_buffer_length_, kMaxFrameSize * decoder->Channels())::webrtc::SafeGe((decoded_buffer_length_), (kMaxFrameSize * decoder ->Channels())) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1342, "decoded_buffer_length_" " " ">=" " " "kMaxFrameSize * decoder->Channels()" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (decoded_buffer_length_) << (kMaxFrameSize * decoder-> Channels()); |
| 1343 | RTC_DCHECK(operation == Operation::kNormal ||(operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation ::kMerge || operation == Operation::kPreemptiveExpand) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1347, "operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation::kMerge || operation == Operation::kPreemptiveExpand" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1344 | operation == Operation::kAccelerate ||(operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation ::kMerge || operation == Operation::kPreemptiveExpand) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1347, "operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation::kMerge || operation == Operation::kPreemptiveExpand" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1345 | operation == Operation::kFastAccelerate ||(operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation ::kMerge || operation == Operation::kPreemptiveExpand) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1347, "operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation::kMerge || operation == Operation::kPreemptiveExpand" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1346 | operation == Operation::kMerge ||(operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation ::kMerge || operation == Operation::kPreemptiveExpand) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1347, "operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation::kMerge || operation == Operation::kPreemptiveExpand" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1347 | operation == Operation::kPreemptiveExpand)(operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation ::kMerge || operation == Operation::kPreemptiveExpand) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1347, "operation == Operation::kNormal || operation == Operation::kAccelerate || operation == Operation::kFastAccelerate || operation == Operation::kMerge || operation == Operation::kPreemptiveExpand" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>(); |
| 1348 | |
| 1349 | auto opt_result = packet_list->front().frame->Decode( |
| 1350 | std::span<int16_t>(&decoded_buffer_[*decoded_length], |
| 1351 | decoded_buffer_length_ - *decoded_length)); |
| 1352 | if (packet_list->front().packet_info) { |
| 1353 | last_decoded_packet_infos_.push_back(*packet_list->front().packet_info); |
| 1354 | } |
| 1355 | packet_list->pop_front(); |
| 1356 | if (opt_result) { |
| 1357 | const auto& result = *opt_result; |
| 1358 | *speech_type = result.speech_type; |
| 1359 | if (result.num_decoded_samples > 0) { |
| 1360 | *decoded_length += dchecked_cast<int>(result.num_decoded_samples); |
| 1361 | // Update `decoder_frame_length_` with number of samples per channel. |
| 1362 | decoder_frame_length_ = |
| 1363 | result.num_decoded_samples / decoder->Channels(); |
| 1364 | } |
| 1365 | } else { |
| 1366 | // Error. |
| 1367 | // TODO(ossu): What to put here? |
| 1368 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1368, ::webrtc::LS_WARNING) << "Decode error"; |
| 1369 | *decoded_length = -1; |
| 1370 | last_decoded_packet_infos_.clear(); |
| 1371 | packet_list->clear(); |
| 1372 | break; |
| 1373 | } |
| 1374 | if (*decoded_length > dchecked_cast<int>(decoded_buffer_length_)) { |
| 1375 | // Guard against overflow. |
| 1376 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1376, ::webrtc::LS_WARNING) << "Decoded too much."; |
| 1377 | packet_list->clear(); |
| 1378 | return kDecodedTooMuch; |
| 1379 | } |
| 1380 | } // End of decode loop. |
| 1381 | |
| 1382 | // If the list is not empty at this point, either a decoding error terminated |
| 1383 | // the while-loop, or list must hold exactly one CNG packet. |
| 1384 | RTC_DCHECK((packet_list->empty() || *decoded_length < 0 || (packet_list ->size() == 1 && decoder_database_->IsComfortNoise (packet_list->front().payload_type))) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false >( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1387, "packet_list->empty() || *decoded_length < 0 || (packet_list->size() == 1 && decoder_database_->IsComfortNoise(packet_list->front().payload_type))" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1385 | packet_list->empty() || *decoded_length < 0 ||(packet_list->empty() || *decoded_length < 0 || (packet_list ->size() == 1 && decoder_database_->IsComfortNoise (packet_list->front().payload_type))) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false >( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1387, "packet_list->empty() || *decoded_length < 0 || (packet_list->size() == 1 && decoder_database_->IsComfortNoise(packet_list->front().payload_type))" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1386 | (packet_list->size() == 1 &&(packet_list->empty() || *decoded_length < 0 || (packet_list ->size() == 1 && decoder_database_->IsComfortNoise (packet_list->front().payload_type))) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false >( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1387, "packet_list->empty() || *decoded_length < 0 || (packet_list->size() == 1 && decoder_database_->IsComfortNoise(packet_list->front().payload_type))" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1387 | decoder_database_->IsComfortNoise(packet_list->front().payload_type)))(packet_list->empty() || *decoded_length < 0 || (packet_list ->size() == 1 && decoder_database_->IsComfortNoise (packet_list->front().payload_type))) ? static_cast<void >(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false >( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1387, "packet_list->empty() || *decoded_length < 0 || (packet_list->size() == 1 && decoder_database_->IsComfortNoise(packet_list->front().payload_type))" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>(); |
| 1388 | return 0; |
| 1389 | } |
| 1390 | |
| 1391 | void NetEqImpl::DoNormal(const int16_t* decoded_buffer, |
| 1392 | size_t decoded_length, |
| 1393 | AudioDecoder::SpeechType speech_type, |
| 1394 | bool play_dtmf) { |
| 1395 | RTC_DCHECK(normal_.get())(normal_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1395, "normal_.get()") & ::webrtc::webrtc_checks_impl:: LogStreamer<>(); |
| 1396 | normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| 1397 | algorithm_buffer_.get()); |
| 1398 | if (decoded_length != 0) { |
| 1399 | last_mode_ = Mode::kNormal; |
| 1400 | } |
| 1401 | |
| 1402 | // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| 1403 | if ((speech_type == AudioDecoder::kComfortNoise) || |
| 1404 | ((last_mode_ == Mode::kCodecInternalCng) && (decoded_length == 0))) { |
| 1405 | // TODO(hlundin): Remove second part of || statement above. |
| 1406 | last_mode_ = Mode::kCodecInternalCng; |
| 1407 | } |
| 1408 | |
| 1409 | if (!play_dtmf) { |
| 1410 | dtmf_tone_generator_->Reset(); |
| 1411 | } |
| 1412 | } |
| 1413 | |
| 1414 | void NetEqImpl::DoMerge(int16_t* decoded_buffer, |
| 1415 | size_t decoded_length, |
| 1416 | AudioDecoder::SpeechType speech_type, |
| 1417 | bool play_dtmf) { |
| 1418 | RTC_DCHECK(merge_.get())(merge_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1418, "merge_.get()") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1419 | size_t new_length = |
| 1420 | merge_->Process(decoded_buffer, decoded_length, algorithm_buffer_.get()); |
| 1421 | // Correction can be negative. |
| 1422 | int expand_length_correction = |
| 1423 | dchecked_cast<int>(new_length) - |
| 1424 | dchecked_cast<int>(decoded_length / algorithm_buffer_->Channels()); |
| 1425 | |
| 1426 | // Update in-call and post-call statistics. |
| 1427 | if (expand_->Muted() || last_decoded_type_ == AudioDecoder::kComfortNoise) { |
| 1428 | // Expand generates only noise. |
| 1429 | stats_->ExpandedNoiseSamplesCorrection(expand_length_correction); |
| 1430 | } else { |
| 1431 | // Expansion generates more than only noise. |
| 1432 | stats_->ExpandedVoiceSamplesCorrection(expand_length_correction); |
| 1433 | } |
| 1434 | |
| 1435 | last_mode_ = Mode::kMerge; |
| 1436 | // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| 1437 | if (speech_type == AudioDecoder::kComfortNoise) { |
| 1438 | last_mode_ = Mode::kCodecInternalCng; |
| 1439 | } |
| 1440 | expand_->Reset(); |
| 1441 | if (!play_dtmf) { |
| 1442 | dtmf_tone_generator_->Reset(); |
| 1443 | } |
| 1444 | } |
| 1445 | |
| 1446 | bool NetEqImpl::DoCodecPlc() { |
| 1447 | AudioDecoder* decoder = decoder_database_->GetActiveDecoder(); |
| 1448 | if (!decoder) { |
| 1449 | return false; |
| 1450 | } |
| 1451 | const size_t channels = algorithm_buffer_->Channels(); |
| 1452 | const size_t requested_samples_per_channel = |
| 1453 | output_size_samples_ - |
| 1454 | (sync_buffer_->FutureLength() - expand_->overlap_length()); |
| 1455 | concealment_audio_.Clear(); |
| 1456 | decoder->GeneratePlc(requested_samples_per_channel, &concealment_audio_); |
| 1457 | if (concealment_audio_.empty()) { |
| 1458 | // Nothing produced. Resort to regular expand. |
| 1459 | return false; |
| 1460 | } |
| 1461 | RTC_CHECK_GE(concealment_audio_.size(),::webrtc::SafeGe((concealment_audio_.size()), (requested_samples_per_channel * channels)) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1462, "concealment_audio_.size()" " " ">=" " " "requested_samples_per_channel * channels" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (concealment_audio_.size()) << (requested_samples_per_channel * channels) |
| 1462 | requested_samples_per_channel * channels)::webrtc::SafeGe((concealment_audio_.size()), (requested_samples_per_channel * channels)) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1462, "concealment_audio_.size()" " " ">=" " " "requested_samples_per_channel * channels" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (concealment_audio_.size()) << (requested_samples_per_channel * channels); |
| 1463 | sync_buffer_->PushBackInterleaved(concealment_audio_); |
| 1464 | RTC_DCHECK_NE(algorithm_buffer_->Channels(), 0)::webrtc::SafeNe((algorithm_buffer_->Channels()), (0)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1464, "algorithm_buffer_->Channels()" " " "!=" " " "0") & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (algorithm_buffer_->Channels()) << (0); |
| 1465 | const size_t concealed_samples_per_channel = |
| 1466 | concealment_audio_.size() / channels; |
| 1467 | |
| 1468 | // Update in-call and post-call statistics. |
| 1469 | const bool is_new_concealment_event = (last_mode_ != Mode::kCodecPlc); |
| 1470 | if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(), |
| 1471 | [](int16_t i) { return i == 0; })) { |
| 1472 | // Expand operation generates only noise. |
| 1473 | stats_->ExpandedNoiseSamples(concealed_samples_per_channel, |
| 1474 | is_new_concealment_event); |
| 1475 | } else { |
| 1476 | // Expand operation generates more than only noise. |
| 1477 | stats_->ExpandedVoiceSamples(concealed_samples_per_channel, |
| 1478 | is_new_concealment_event); |
| 1479 | } |
| 1480 | last_mode_ = Mode::kCodecPlc; |
| 1481 | if (!generated_noise_stopwatch_) { |
| 1482 | // Start a new stopwatch since we may be covering for a lost CNG packet. |
| 1483 | generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch(); |
| 1484 | } |
| 1485 | return true; |
| 1486 | } |
| 1487 | |
| 1488 | int NetEqImpl::DoExpand(bool play_dtmf) { |
| 1489 | while ((sync_buffer_->FutureLength() - expand_->overlap_length()) < |
| 1490 | output_size_samples_) { |
| 1491 | algorithm_buffer_->Clear(); |
| 1492 | int return_value = expand_->Process(algorithm_buffer_.get()); |
| 1493 | size_t length = algorithm_buffer_->Size(); |
| 1494 | bool is_new_concealment_event = (last_mode_ != Mode::kExpand); |
| 1495 | |
| 1496 | // Update in-call and post-call statistics. |
| 1497 | if (expand_->Muted() || last_decoded_type_ == AudioDecoder::kComfortNoise) { |
| 1498 | // Expand operation generates only noise. |
| 1499 | stats_->ExpandedNoiseSamples(length, is_new_concealment_event); |
| 1500 | } else { |
| 1501 | // Expand operation generates more than only noise. |
| 1502 | stats_->ExpandedVoiceSamples(length, is_new_concealment_event); |
| 1503 | } |
| 1504 | |
| 1505 | last_mode_ = Mode::kExpand; |
| 1506 | |
| 1507 | if (return_value < 0) { |
| 1508 | return return_value; |
| 1509 | } |
| 1510 | |
| 1511 | sync_buffer_->PushBack(*algorithm_buffer_); |
| 1512 | algorithm_buffer_->Clear(); |
| 1513 | } |
| 1514 | if (!play_dtmf) { |
| 1515 | dtmf_tone_generator_->Reset(); |
| 1516 | } |
| 1517 | |
| 1518 | if (!generated_noise_stopwatch_) { |
| 1519 | // Start a new stopwatch since we may be covering for a lost CNG packet. |
| 1520 | generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch(); |
| 1521 | } |
| 1522 | |
| 1523 | return 0; |
| 1524 | } |
| 1525 | |
| 1526 | int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, |
| 1527 | size_t decoded_length, |
| 1528 | AudioDecoder::SpeechType speech_type, |
| 1529 | bool play_dtmf, |
| 1530 | bool fast_accelerate) { |
| 1531 | const size_t required_samples = |
| 1532 | static_cast<size_t>(240 * fs_mult_); // Must have 30 ms. |
| 1533 | size_t borrowed_samples_per_channel = 0; |
| 1534 | size_t num_channels = algorithm_buffer_->Channels(); |
| 1535 | size_t decoded_length_per_channel = decoded_length / num_channels; |
| 1536 | if (decoded_length_per_channel < required_samples) { |
| 1537 | // Must move data from the `sync_buffer_` in order to get 30 ms. |
| 1538 | borrowed_samples_per_channel = |
| 1539 | static_cast<int>(required_samples - decoded_length_per_channel); |
| 1540 | memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels], |
| 1541 | decoded_buffer, sizeof(int16_t) * decoded_length); |
| 1542 | sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel, |
| 1543 | decoded_buffer); |
| 1544 | decoded_length = required_samples * num_channels; |
| 1545 | } |
| 1546 | |
| 1547 | size_t samples_removed = 0; |
| 1548 | Accelerate::ReturnCodes return_code = |
| 1549 | accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate, |
| 1550 | algorithm_buffer_.get(), &samples_removed); |
| 1551 | stats_->AcceleratedSamples(samples_removed); |
| 1552 | switch (return_code) { |
| 1553 | case Accelerate::kSuccess: |
| 1554 | last_mode_ = Mode::kAccelerateSuccess; |
| 1555 | break; |
| 1556 | case Accelerate::kSuccessLowEnergy: |
| 1557 | last_mode_ = Mode::kAccelerateLowEnergy; |
| 1558 | break; |
| 1559 | case Accelerate::kNoStretch: |
| 1560 | last_mode_ = Mode::kAccelerateFail; |
| 1561 | break; |
| 1562 | case Accelerate::kError: |
| 1563 | // TODO(hlundin): Map to Modes::kError instead? |
| 1564 | last_mode_ = Mode::kAccelerateFail; |
| 1565 | return kAccelerateError; |
| 1566 | } |
| 1567 | |
| 1568 | if (borrowed_samples_per_channel > 0) { |
| 1569 | // Copy borrowed samples back to the `sync_buffer_`. |
| 1570 | size_t length = algorithm_buffer_->Size(); |
| 1571 | if (length < borrowed_samples_per_channel) { |
| 1572 | // This destroys the beginning of the buffer, but will not cause any |
| 1573 | // problems. |
| 1574 | sync_buffer_->ReplaceAtIndex( |
| 1575 | *algorithm_buffer_, |
| 1576 | sync_buffer_->Size() - borrowed_samples_per_channel); |
| 1577 | sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length); |
| 1578 | algorithm_buffer_->PopFront(length); |
| 1579 | RTC_DCHECK(algorithm_buffer_->Empty())(algorithm_buffer_->Empty()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1579, "algorithm_buffer_->Empty()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 1580 | } else { |
| 1581 | sync_buffer_->ReplaceAtIndex( |
| 1582 | *algorithm_buffer_, borrowed_samples_per_channel, |
| 1583 | sync_buffer_->Size() - borrowed_samples_per_channel); |
| 1584 | algorithm_buffer_->PopFront(borrowed_samples_per_channel); |
| 1585 | } |
| 1586 | } |
| 1587 | |
| 1588 | // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| 1589 | if (speech_type == AudioDecoder::kComfortNoise) { |
| 1590 | last_mode_ = Mode::kCodecInternalCng; |
| 1591 | } |
| 1592 | if (!play_dtmf) { |
| 1593 | dtmf_tone_generator_->Reset(); |
| 1594 | } |
| 1595 | expand_->Reset(); |
| 1596 | return 0; |
| 1597 | } |
| 1598 | |
| 1599 | int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer, |
| 1600 | size_t decoded_length, |
| 1601 | AudioDecoder::SpeechType speech_type, |
| 1602 | bool play_dtmf) { |
| 1603 | const size_t required_samples = |
| 1604 | static_cast<size_t>(240 * fs_mult_); // Must have 30 ms. |
| 1605 | size_t num_channels = algorithm_buffer_->Channels(); |
| 1606 | size_t borrowed_samples_per_channel = 0; |
| 1607 | size_t old_borrowed_samples_per_channel = 0; |
| 1608 | size_t decoded_length_per_channel = decoded_length / num_channels; |
| 1609 | if (decoded_length_per_channel < required_samples) { |
| 1610 | // Must move data from the `sync_buffer_` in order to get 30 ms. |
| 1611 | borrowed_samples_per_channel = |
| 1612 | required_samples - decoded_length_per_channel; |
| 1613 | // Calculate how many of these were already played out. |
| 1614 | old_borrowed_samples_per_channel = |
| 1615 | (borrowed_samples_per_channel > sync_buffer_->FutureLength()) |
| 1616 | ? (borrowed_samples_per_channel - sync_buffer_->FutureLength()) |
| 1617 | : 0; |
| 1618 | memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels], |
| 1619 | decoded_buffer, sizeof(int16_t) * decoded_length); |
| 1620 | sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel, |
| 1621 | decoded_buffer); |
| 1622 | decoded_length = required_samples * num_channels; |
| 1623 | } |
| 1624 | |
| 1625 | size_t samples_added = 0; |
| 1626 | PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process( |
| 1627 | decoded_buffer, decoded_length, old_borrowed_samples_per_channel, |
| 1628 | algorithm_buffer_.get(), &samples_added); |
| 1629 | stats_->PreemptiveExpandedSamples(samples_added); |
| 1630 | switch (return_code) { |
| 1631 | case PreemptiveExpand::kSuccess: |
| 1632 | last_mode_ = Mode::kPreemptiveExpandSuccess; |
| 1633 | break; |
| 1634 | case PreemptiveExpand::kSuccessLowEnergy: |
| 1635 | last_mode_ = Mode::kPreemptiveExpandLowEnergy; |
| 1636 | break; |
| 1637 | case PreemptiveExpand::kNoStretch: |
| 1638 | last_mode_ = Mode::kPreemptiveExpandFail; |
| 1639 | break; |
| 1640 | case PreemptiveExpand::kError: |
| 1641 | // TODO(hlundin): Map to Modes::kError instead? |
| 1642 | last_mode_ = Mode::kPreemptiveExpandFail; |
| 1643 | return kPreemptiveExpandError; |
| 1644 | } |
| 1645 | |
| 1646 | if (borrowed_samples_per_channel > 0) { |
| 1647 | // Copy borrowed samples back to the `sync_buffer_`. |
| 1648 | sync_buffer_->ReplaceAtIndex( |
| 1649 | *algorithm_buffer_, borrowed_samples_per_channel, |
| 1650 | sync_buffer_->Size() - borrowed_samples_per_channel); |
| 1651 | algorithm_buffer_->PopFront(borrowed_samples_per_channel); |
| 1652 | } |
| 1653 | |
| 1654 | // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| 1655 | if (speech_type == AudioDecoder::kComfortNoise) { |
| 1656 | last_mode_ = Mode::kCodecInternalCng; |
| 1657 | } |
| 1658 | if (!play_dtmf) { |
| 1659 | dtmf_tone_generator_->Reset(); |
| 1660 | } |
| 1661 | expand_->Reset(); |
| 1662 | return 0; |
| 1663 | } |
| 1664 | |
| 1665 | int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) { |
| 1666 | if (!packet_list->empty()) { |
| 1667 | // Must have exactly one SID frame at this point. |
| 1668 | RTC_DCHECK_EQ(packet_list->size(), 1)::webrtc::SafeEq((packet_list->size()), (1)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1668, "packet_list->size()" " " "==" " " "1") & ::webrtc ::webrtc_checks_impl::LogStreamer<>() << (packet_list ->size()) << (1); |
| 1669 | const Packet& packet = packet_list->front(); |
| 1670 | if (!decoder_database_->IsComfortNoise(packet.payload_type)) { |
| 1671 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1671, ::webrtc::LS_ERROR) << "Trying to decode non-CNG payload as CNG."; |
| 1672 | return kOtherError; |
| 1673 | } |
| 1674 | if (comfort_noise_->UpdateParameters(packet) == |
| 1675 | ComfortNoise::kInternalError) { |
| 1676 | algorithm_buffer_->Zeros(output_size_samples_); |
| 1677 | return -comfort_noise_->internal_error_code(); |
| 1678 | } |
| 1679 | } |
| 1680 | int cn_return = |
| 1681 | comfort_noise_->Generate(output_size_samples_, algorithm_buffer_.get()); |
| 1682 | expand_->Reset(); |
| 1683 | last_mode_ = Mode::kRfc3389Cng; |
| 1684 | if (!play_dtmf) { |
| 1685 | dtmf_tone_generator_->Reset(); |
| 1686 | } |
| 1687 | if (cn_return == ComfortNoise::kInternalError) { |
| 1688 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1688, ::webrtc::LS_WARNING) << "Comfort noise generator returned error code: " |
| 1689 | << comfort_noise_->internal_error_code(); |
| 1690 | return kComfortNoiseErrorCode; |
| 1691 | } else if (cn_return == ComfortNoise::kUnknownPayloadType) { |
| 1692 | return kUnknownRtpPayloadType; |
| 1693 | } |
| 1694 | return 0; |
| 1695 | } |
| 1696 | |
| 1697 | void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer, |
| 1698 | size_t decoded_length) { |
| 1699 | RTC_DCHECK(normal_.get())(normal_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1699, "normal_.get()") & ::webrtc::webrtc_checks_impl:: LogStreamer<>(); |
| 1700 | normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| 1701 | algorithm_buffer_.get()); |
| 1702 | last_mode_ = Mode::kCodecInternalCng; |
| 1703 | expand_->Reset(); |
| 1704 | } |
| 1705 | |
| 1706 | int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) { |
| 1707 | // This block of the code and the block further down, handling `dtmf_switch` |
| 1708 | // are commented out. Otherwise playing out-of-band DTMF would fail in VoE |
| 1709 | // test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is |
| 1710 | // equivalent to `dtmf_switch` always be false. |
| 1711 | // |
| 1712 | // See http://webrtc-codereview.appspot.com/1195004/ for discussion |
| 1713 | // On this issue. This change might cause some glitches at the point of |
| 1714 | // switch from audio to DTMF. Issue 1545 is filed to track this. |
| 1715 | // |
| 1716 | // bool dtmf_switch = false; |
| 1717 | // if ((last_mode_ != Modes::kDtmf) && |
| 1718 | // dtmf_tone_generator_->initialized()) { |
| 1719 | // // Special case; see below. |
| 1720 | // // We must catch this before calling Generate, since `initialized` is |
| 1721 | // // modified in that call. |
| 1722 | // dtmf_switch = true; |
| 1723 | // } |
| 1724 | |
| 1725 | int dtmf_return_value = 0; |
| 1726 | if (!dtmf_tone_generator_->initialized()) { |
| 1727 | // Initialize if not already done. |
| 1728 | dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no, |
| 1729 | dtmf_event.volume); |
| 1730 | } |
| 1731 | |
| 1732 | if (dtmf_return_value == 0) { |
| 1733 | // Generate DTMF signal. |
| 1734 | dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_, |
| 1735 | algorithm_buffer_.get()); |
| 1736 | } |
| 1737 | |
| 1738 | if (dtmf_return_value < 0) { |
| 1739 | algorithm_buffer_->Zeros(output_size_samples_); |
| 1740 | return dtmf_return_value; |
| 1741 | } |
| 1742 | |
| 1743 | // if (dtmf_switch) { |
| 1744 | // // This is the special case where the previous operation was DTMF |
| 1745 | // // overdub, but the current instruction is "regular" DTMF. We must make |
| 1746 | // // sure that the DTMF does not have any discontinuities. The first DTMF |
| 1747 | // // sample that we generate now must be played out immediately, therefore |
| 1748 | // // it must be copied to the speech buffer. |
| 1749 | // // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and |
| 1750 | // // verify correct operation. |
| 1751 | // RTC_DCHECK_NOTREACHED(); |
| 1752 | // // Must generate enough data to replace all of the `sync_buffer_` |
| 1753 | // // "future". |
| 1754 | // int required_length = sync_buffer_->FutureLength(); |
| 1755 | // RTC_DCHECK(dtmf_tone_generator_->initialized()); |
| 1756 | // dtmf_return_value = dtmf_tone_generator_->Generate(required_length, |
| 1757 | // algorithm_buffer_); |
| 1758 | // RTC_DCHECK((size_t) required_length == algorithm_buffer_->Size()); |
| 1759 | // if (dtmf_return_value < 0) { |
| 1760 | // algorithm_buffer_->Zeros(output_size_samples_); |
| 1761 | // return dtmf_return_value; |
| 1762 | // } |
| 1763 | // |
| 1764 | // // Overwrite the "future" part of the speech buffer with the new DTMF |
| 1765 | // // data. |
| 1766 | // // TODO(hlundin): It seems that this overwriting has gone lost. |
| 1767 | // // Not adapted for multi-channel yet. |
| 1768 | // RTC_DCHECK(algorithm_buffer_->Channels() == 1); |
| 1769 | // if (algorithm_buffer_->Channels() != 1) { |
| 1770 | // RTC_LOG(LS_WARNING) << "DTMF not supported for more than one channel"; |
| 1771 | // return kStereoNotSupported; |
| 1772 | // } |
| 1773 | // // Shuffle the remaining data to the beginning of algorithm buffer. |
| 1774 | // algorithm_buffer_->PopFront(sync_buffer_->FutureLength()); |
| 1775 | // } |
| 1776 | |
| 1777 | sync_buffer_->IncreaseEndTimestamp( |
| 1778 | static_cast<uint32_t>(output_size_samples_)); |
| 1779 | expand_->Reset(); |
| 1780 | last_mode_ = Mode::kDtmf; |
| 1781 | |
| 1782 | // Set to false because the DTMF is already in the algorithm buffer. |
| 1783 | *play_dtmf = false; |
| 1784 | return 0; |
| 1785 | } |
| 1786 | |
| 1787 | int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, |
| 1788 | size_t num_channels, |
| 1789 | int16_t* output) const { |
| 1790 | size_t out_index = 0; |
| 1791 | size_t overdub_length = output_size_samples_; // Default value. |
| 1792 | |
| 1793 | if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { |
| 1794 | // Special operation for transition from "DTMF only" to "DTMF overdub". |
| 1795 | out_index = |
| 1796 | std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(), |
| 1797 | output_size_samples_); |
| 1798 | overdub_length = output_size_samples_ - out_index; |
| 1799 | } |
| 1800 | |
| 1801 | AudioMultiVector dtmf_output(num_channels); |
| 1802 | int dtmf_return_value = 0; |
| 1803 | if (!dtmf_tone_generator_->initialized()) { |
| 1804 | dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no, |
| 1805 | dtmf_event.volume); |
| 1806 | } |
| 1807 | if (dtmf_return_value == 0) { |
| 1808 | dtmf_return_value = |
| 1809 | dtmf_tone_generator_->Generate(overdub_length, &dtmf_output); |
| 1810 | RTC_DCHECK_EQ(overdub_length, dtmf_output.Size())::webrtc::SafeEq((overdub_length), (dtmf_output.Size())) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1810, "overdub_length" " " "==" " " "dtmf_output.Size()") & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (overdub_length) << (dtmf_output.Size()); |
| 1811 | } |
| 1812 | dtmf_output.ReadInterleaved(overdub_length, &output[out_index]); |
| 1813 | return dtmf_return_value < 0 ? dtmf_return_value : 0; |
| 1814 | } |
| 1815 | |
| 1816 | int NetEqImpl::ExtractPackets(size_t required_samples, |
| 1817 | PacketList* packet_list) { |
| 1818 | bool next_packet_available = false; |
| 1819 | |
| 1820 | const Packet* next_packet = packet_buffer_->PeekNextPacket(); |
| 1821 | RTC_DCHECK(next_packet)(next_packet) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1821, "next_packet") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1822 | if (!next_packet) { |
| 1823 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1823, ::webrtc::LS_ERROR) << "Packet buffer unexpectedly empty."; |
| 1824 | return -1; |
| 1825 | } |
| 1826 | uint32_t first_timestamp = next_packet->timestamp; |
| 1827 | size_t extracted_samples = 0; |
| 1828 | |
| 1829 | // Packet extraction loop. |
| 1830 | do { |
| 1831 | timestamp_ = next_packet->timestamp; |
| 1832 | std::optional<Packet> packet = packet_buffer_->GetNextPacket(); |
| 1833 | // `next_packet` may be invalid after the `packet_buffer_` operation. |
| 1834 | next_packet = nullptr; |
| 1835 | if (!packet) { |
| 1836 | RTC_LOG(LS_ERROR)!::webrtc::LogMessage::IsNoop<::webrtc::LS_ERROR>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1836, ::webrtc::LS_ERROR) << "Should always be able to extract a packet here"; |
| 1837 | RTC_DCHECK_NOTREACHED()(false) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1837, "false") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); // Should always be able to extract a packet |
| 1838 | // here. |
| 1839 | return -1; |
| 1840 | } |
| 1841 | const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs(); |
| 1842 | stats_->StoreWaitingTime(waiting_time_ms); |
| 1843 | RTC_DCHECK(!packet->empty())(!packet->empty()) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1843, "!packet->empty()") & ::webrtc::webrtc_checks_impl ::LogStreamer<>(); |
| 1844 | |
| 1845 | const bool has_cng_packet = |
| 1846 | decoder_database_->IsComfortNoise(packet->payload_type); |
| 1847 | // Store number of extracted samples. |
| 1848 | size_t packet_duration = 0; |
| 1849 | if (packet->frame) { |
| 1850 | packet_duration = packet->frame->Duration(); |
| 1851 | // TODO(ossu): Is this the correct way to track Opus FEC packets? |
| 1852 | if (packet->priority.codec_level > 0) { |
| 1853 | stats_->SecondaryDecodedSamples(dchecked_cast<int>(packet_duration)); |
| 1854 | } |
| 1855 | } else if (!has_cng_packet) { |
| 1856 | RTC_LOG(LS_WARNING)!::webrtc::LogMessage::IsNoop<::webrtc::LS_WARNING>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1856, ::webrtc::LS_WARNING) << "Unknown payload type " |
| 1857 | << static_cast<int>(packet->payload_type); |
| 1858 | RTC_DCHECK_NOTREACHED()(false) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1858, "false") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1859 | } |
| 1860 | |
| 1861 | if (packet_duration == 0) { |
| 1862 | // Decoder did not return a packet duration. Assume that the packet |
| 1863 | // contains the same number of samples as the previous one. |
| 1864 | packet_duration = decoder_frame_length_; |
| 1865 | } |
| 1866 | extracted_samples = packet->timestamp - first_timestamp + packet_duration; |
| 1867 | |
| 1868 | RTC_DCHECK(controller_)(controller_) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1868, "controller_") & ::webrtc::webrtc_checks_impl::LogStreamer <>(); |
| 1869 | TimeDelta processing_time = TimeDelta::Zero(); |
| 1870 | |
| 1871 | if (packet->packet_info.has_value() && |
| 1872 | !packet->packet_info->receive_time().IsMinusInfinity()) { |
| 1873 | processing_time = |
| 1874 | env_.clock().CurrentTime() - packet->packet_info->receive_time(); |
| 1875 | } |
| 1876 | |
| 1877 | stats_->JitterBufferDelay( |
| 1878 | packet_duration, waiting_time_ms, controller_->TargetLevelMs(), |
| 1879 | controller_->UnlimitedTargetLevelMs(), processing_time.us()); |
| 1880 | |
| 1881 | // Check what packet is available next. |
| 1882 | next_packet = packet_buffer_->PeekNextPacket(); |
| 1883 | next_packet_available = |
| 1884 | next_packet && next_packet->payload_type == packet->payload_type && |
| 1885 | next_packet->timestamp == packet->timestamp + packet_duration && |
| 1886 | !has_cng_packet; |
| 1887 | |
| 1888 | packet_list->push_back(std::move(*packet)); // Store packet in list. |
| 1889 | packet = std::nullopt; // Ensure it's never used after the move. |
| 1890 | } while (extracted_samples < required_samples && next_packet_available); |
| 1891 | |
| 1892 | if (extracted_samples > 0) { |
| 1893 | // Delete old packets only when we are going to decode something. Otherwise, |
| 1894 | // we could end up in the situation where we never decode anything, since |
| 1895 | // all incoming packets are considered too old but the buffer will also |
| 1896 | // never be flooded and flushed. |
| 1897 | packet_buffer_->DiscardAllOldPackets(timestamp_); |
| 1898 | } |
| 1899 | |
| 1900 | return dchecked_cast<int>(extracted_samples); |
| 1901 | } |
| 1902 | |
| 1903 | void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { |
| 1904 | // Delete objects and create new ones. |
| 1905 | expand_.reset(expand_factory_->Create(background_noise_.get(), |
| 1906 | sync_buffer_.get(), &random_vector_, |
| 1907 | stats_.get(), fs_hz, channels)); |
| 1908 | merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); |
| 1909 | } |
| 1910 | |
| 1911 | void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { |
| 1912 | RTC_LOG(LS_VERBOSE)!::webrtc::LogMessage::IsNoop<::webrtc::LS_VERBOSE>() && ::webrtc::webrtc_logging_impl::LogCall() & ::webrtc::webrtc_logging_impl ::LogStreamer<>() << ::webrtc::webrtc_logging_impl ::LogMetadata("./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1912, ::webrtc::LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " " |
| 1913 | << channels; |
| 1914 | RTC_CHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 ||(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1915, "fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() |
| 1915 | fs_hz == 48000)(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1915, "fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>(); |
| 1916 | RTC_CHECK_GT(channels, 0)::webrtc::SafeGt((channels), (0)) ? static_cast<void>(0 ) : ::webrtc::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1916, "channels" " " ">" " " "0") & ::webrtc::webrtc_checks_impl ::LogStreamer<>() << (channels) << (0); |
| 1917 | RTC_CHECK_LE(channels, kMaxNumberOfAudioChannels)::webrtc::SafeLe((channels), (kMaxNumberOfAudioChannels)) ? static_cast <void>(0) : ::webrtc::webrtc_checks_impl::FatalLogCall< true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1917, "channels" " " "<=" " " "kMaxNumberOfAudioChannels" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (channels) << (kMaxNumberOfAudioChannels); |
| 1918 | |
| 1919 | // The format must fit in an AudioFrame. Situations where this could |
| 1920 | // theoratically happen but aren't supported is e.g. if receiving 24 channels |
| 1921 | // of 10ms 48 kHz buffers. |
| 1922 | output_size_samples_ = SampleRateToDefaultChannelSize(fs_hz); |
| 1923 | RTC_CHECK_LE(channels * output_size_samples_,::webrtc::SafeLe((channels * output_size_samples_), (AudioFrame ::kMaxDataSizeSamples)) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1924, "channels * output_size_samples_" " " "<=" " " "AudioFrame::kMaxDataSizeSamples" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (channels * output_size_samples_) << (AudioFrame::kMaxDataSizeSamples ) |
| 1924 | AudioFrame::kMaxDataSizeSamples)::webrtc::SafeLe((channels * output_size_samples_), (AudioFrame ::kMaxDataSizeSamples)) ? static_cast<void>(0) : ::webrtc ::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1924, "channels * output_size_samples_" " " "<=" " " "AudioFrame::kMaxDataSizeSamples" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (channels * output_size_samples_) << (AudioFrame::kMaxDataSizeSamples ); |
| 1925 | |
| 1926 | // Before changing the sample rate, end and report any ongoing expand event. |
| 1927 | stats_->EndExpandEvent(fs_hz_); |
| 1928 | fs_hz_ = fs_hz; |
| 1929 | fs_mult_ = fs_hz / 8000; |
| 1930 | RTC_DCHECK_EQ(output_size_samples_,::webrtc::SafeEq((output_size_samples_), (static_cast<size_t >(kOutputSizeMs * 8 * fs_mult_))) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1931, "output_size_samples_" " " "==" " " "static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_)" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (output_size_samples_) << (static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_)) |
| 1931 | static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_))::webrtc::SafeEq((output_size_samples_), (static_cast<size_t >(kOutputSizeMs * 8 * fs_mult_))) ? static_cast<void> (0) : ::webrtc::webrtc_checks_impl::FatalLogCall<true>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1931, "output_size_samples_" " " "==" " " "static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_)" ) & ::webrtc::webrtc_checks_impl::LogStreamer<>() << (output_size_samples_) << (static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_)); |
| 1932 | decoder_frame_length_ = 3 * output_size_samples_; // Initialize to 30ms. |
| 1933 | |
| 1934 | last_mode_ = Mode::kNormal; |
| 1935 | |
| 1936 | ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder(); |
| 1937 | if (cng_decoder) |
| 1938 | cng_decoder->Reset(); |
| 1939 | |
| 1940 | // Delete algorithm buffer and create a new one. |
| 1941 | algorithm_buffer_.reset(new AudioMultiVector(channels)); |
| 1942 | |
| 1943 | // Delete sync buffer and create a new one. |
| 1944 | sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_)); |
| 1945 | |
| 1946 | // Delete BackgroundNoise object and create a new one. |
| 1947 | background_noise_.reset(new BackgroundNoise(channels)); |
| 1948 | |
| 1949 | // Reset random vector. |
| 1950 | random_vector_.Reset(); |
| 1951 | |
| 1952 | UpdatePlcComponents(fs_hz, channels); |
| 1953 | |
| 1954 | // Move index so that we create a small set of future samples (all 0). |
| 1955 | sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| 1956 | expand_->overlap_length()); |
| 1957 | |
| 1958 | normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_, |
| 1959 | expand_.get(), stats_.get())); |
| 1960 | accelerate_.reset( |
| 1961 | accelerate_factory_->Create(fs_hz, channels, *background_noise_)); |
| 1962 | preemptive_expand_.reset(preemptive_expand_factory_->Create( |
| 1963 | fs_hz, channels, *background_noise_, expand_->overlap_length())); |
| 1964 | |
| 1965 | // Delete ComfortNoise object and create a new one. |
| 1966 | comfort_noise_.reset( |
| 1967 | new ComfortNoise(fs_hz, decoder_database_.get(), sync_buffer_.get())); |
| 1968 | |
| 1969 | // Verify that `decoded_buffer_` is long enough. |
| 1970 | if (decoded_buffer_length_ < kMaxFrameSize * channels) { |
| 1971 | // Reallocate to larger size. |
| 1972 | decoded_buffer_length_ = kMaxFrameSize * channels; |
| 1973 | decoded_buffer_.reset(new int16_t[decoded_buffer_length_]); |
| 1974 | } |
| 1975 | RTC_CHECK(controller_)(controller_) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1975, "controller_") & ::webrtc::webrtc_checks_impl::LogStreamer <>() << "Unexpectedly found no NetEqController"; |
| 1976 | controller_->SetSampleRate(fs_hz_, output_size_samples_); |
| 1977 | } |
| 1978 | |
| 1979 | NetEqImpl::OutputType NetEqImpl::LastOutputType() { |
| 1980 | RTC_DCHECK(expand_.get())(expand_.get()) ? static_cast<void>(0) : ::webrtc::webrtc_checks_impl ::FatalLogCall<false>( "./../../../../../../third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc" , 1980, "expand_.get()") & ::webrtc::webrtc_checks_impl:: LogStreamer<>(); |
| 1981 | if (last_mode_ == Mode::kCodecInternalCng || |
| 1982 | last_mode_ == Mode::kRfc3389Cng) { |
| 1983 | return OutputType::kCNG; |
| 1984 | } else if (last_mode_ == Mode::kExpand && expand_->MuteFactor(0) == 0) { |
| 1985 | // Expand mode has faded down to background noise only (very long expand). |
| 1986 | return OutputType::kPLCCNG; |
| 1987 | } else if (last_mode_ == Mode::kExpand) { |
| 1988 | return OutputType::kPLC; |
| 1989 | } else if (last_mode_ == Mode::kCodecPlc) { |
| 1990 | return OutputType::kCodecPLC; |
| 1991 | } else { |
| 1992 | return OutputType::kNormalSpeech; |
| 1993 | } |
| 1994 | } |
| 1995 | |
| 1996 | NetEqController::PacketArrivedInfo NetEqImpl::ToPacketArrivedInfo( |
| 1997 | const Packet& packet) const { |
| 1998 | const DecoderDatabase::DecoderInfo* dec_info = |
| 1999 | decoder_database_->GetDecoderInfo(packet.payload_type); |
| 2000 | |
| 2001 | NetEqController::PacketArrivedInfo info; |
| 2002 | info.is_cng_or_dtmf = |
| 2003 | dec_info && (dec_info->IsComfortNoise() || dec_info->IsDtmf()); |
| 2004 | info.packet_length_samples = |
| 2005 | packet.frame ? packet.frame->Duration() : decoder_frame_length_; |
| 2006 | info.main_timestamp = packet.timestamp; |
| 2007 | info.main_sequence_number = packet.sequence_number; |
| 2008 | info.is_dtx = packet.frame && packet.frame->IsDtxPacket(); |
| 2009 | info.buffer_flush = false; |
| 2010 | return info; |
| 2011 | } |
| 2012 | |
| 2013 | void NetEqImpl::SetMaximumBufferPackets(size_t max_packets) { |
| 2014 | packet_buffer_->SetMaxNumberOfPackets(max_packets); |
| 2015 | } |
| 2016 | |
| 2017 | void NetEqImpl::SetFastAccelerate(bool enable) { |
| 2018 | enable_fast_accelerate_ = enable; |
| 2019 | } |
| 2020 | |
| 2021 | } // namespace webrtc |