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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 "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h" | |
| 12 | |
| 13 #include <assert.h> | |
| 14 #include <stdlib.h> | |
| 15 #include <vector> | |
| 16 | |
| 17 #include "webrtc/base/checks.h" | |
| 18 #include "webrtc/base/safe_conversions.h" | |
| 19 #include "webrtc/engine_configurations.h" | |
| 20 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" | |
| 21 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" | |
| 22 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | |
| 23 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" | |
| 24 #include "webrtc/system_wrappers/include/logging.h" | |
| 25 #include "webrtc/system_wrappers/include/metrics.h" | |
| 26 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" | |
| 27 #include "webrtc/system_wrappers/include/trace.h" | |
| 28 #include "webrtc/typedefs.h" | |
| 29 | |
| 30 namespace webrtc { | |
| 31 | |
| 32 namespace { | |
| 33 | |
| 34 // Adds a codec usage sample to the histogram. | |
| 35 void UpdateCodecTypeHistogram(size_t codec_type) { | |
| 36 RTC_HISTOGRAM_ENUMERATION( | |
| 37 "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type), | |
| 38 static_cast<int>( | |
| 39 webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)); | |
| 40 } | |
| 41 | |
| 42 } // namespace | |
| 43 | |
| 44 namespace acm2 { | |
| 45 | |
| 46 struct EncoderFactory { | |
| 47 AudioEncoder* external_speech_encoder = nullptr; | |
| 48 CodecManager codec_manager; | |
| 49 RentACodec rent_a_codec; | |
| 50 }; | |
| 51 | |
| 52 namespace { | |
| 53 | |
| 54 // TODO(turajs): the same functionality is used in NetEq. If both classes | |
| 55 // need them, make it a static function in ACMCodecDB. | |
| 56 bool IsCodecRED(const CodecInst& codec) { | |
| 57 return (STR_CASE_CMP(codec.plname, "RED") == 0); | |
| 58 } | |
| 59 | |
| 60 bool IsCodecCN(const CodecInst& codec) { | |
| 61 return (STR_CASE_CMP(codec.plname, "CN") == 0); | |
| 62 } | |
| 63 | |
| 64 // Stereo-to-mono can be used as in-place. | |
| 65 int DownMix(const AudioFrame& frame, | |
| 66 size_t length_out_buff, | |
| 67 int16_t* out_buff) { | |
| 68 if (length_out_buff < frame.samples_per_channel_) { | |
| 69 return -1; | |
| 70 } | |
| 71 for (size_t n = 0; n < frame.samples_per_channel_; ++n) | |
| 72 out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1; | |
| 73 return 0; | |
| 74 } | |
| 75 | |
| 76 // Mono-to-stereo can be used as in-place. | |
| 77 int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) { | |
| 78 if (length_out_buff < frame.samples_per_channel_) { | |
| 79 return -1; | |
| 80 } | |
| 81 for (size_t n = frame.samples_per_channel_; n != 0; --n) { | |
| 82 size_t i = n - 1; | |
| 83 int16_t sample = frame.data_[i]; | |
| 84 out_buff[2 * i + 1] = sample; | |
| 85 out_buff[2 * i] = sample; | |
| 86 } | |
| 87 return 0; | |
| 88 } | |
| 89 | |
| 90 void ConvertEncodedInfoToFragmentationHeader( | |
| 91 const AudioEncoder::EncodedInfo& info, | |
| 92 RTPFragmentationHeader* frag) { | |
| 93 if (info.redundant.empty()) { | |
| 94 frag->fragmentationVectorSize = 0; | |
| 95 return; | |
| 96 } | |
| 97 | |
| 98 frag->VerifyAndAllocateFragmentationHeader( | |
| 99 static_cast<uint16_t>(info.redundant.size())); | |
| 100 frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size()); | |
| 101 size_t offset = 0; | |
| 102 for (size_t i = 0; i < info.redundant.size(); ++i) { | |
| 103 frag->fragmentationOffset[i] = offset; | |
| 104 offset += info.redundant[i].encoded_bytes; | |
| 105 frag->fragmentationLength[i] = info.redundant[i].encoded_bytes; | |
| 106 frag->fragmentationTimeDiff[i] = rtc::checked_cast<uint16_t>( | |
| 107 info.encoded_timestamp - info.redundant[i].encoded_timestamp); | |
| 108 frag->fragmentationPlType[i] = info.redundant[i].payload_type; | |
| 109 } | |
| 110 } | |
| 111 | |
| 112 // Wraps a raw AudioEncoder pointer. The idea is that you can put one of these | |
| 113 // in a unique_ptr, to protect the contained raw pointer from being deleted | |
| 114 // when the unique_ptr expires. (This is of course a bad idea in general, but | |
| 115 // backwards compatibility.) | |
| 116 class RawAudioEncoderWrapper final : public AudioEncoder { | |
| 117 public: | |
| 118 RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {} | |
| 119 int SampleRateHz() const override { return enc_->SampleRateHz(); } | |
| 120 size_t NumChannels() const override { return enc_->NumChannels(); } | |
| 121 int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); } | |
| 122 size_t Num10MsFramesInNextPacket() const override { | |
| 123 return enc_->Num10MsFramesInNextPacket(); | |
| 124 } | |
| 125 size_t Max10MsFramesInAPacket() const override { | |
| 126 return enc_->Max10MsFramesInAPacket(); | |
| 127 } | |
| 128 int GetTargetBitrate() const override { return enc_->GetTargetBitrate(); } | |
| 129 EncodedInfo EncodeImpl(uint32_t rtp_timestamp, | |
| 130 rtc::ArrayView<const int16_t> audio, | |
| 131 rtc::Buffer* encoded) override { | |
| 132 return enc_->Encode(rtp_timestamp, audio, encoded); | |
| 133 } | |
| 134 void Reset() override { return enc_->Reset(); } | |
| 135 bool SetFec(bool enable) override { return enc_->SetFec(enable); } | |
| 136 bool SetDtx(bool enable) override { return enc_->SetDtx(enable); } | |
| 137 bool SetApplication(Application application) override { | |
| 138 return enc_->SetApplication(application); | |
| 139 } | |
| 140 void SetMaxPlaybackRate(int frequency_hz) override { | |
| 141 return enc_->SetMaxPlaybackRate(frequency_hz); | |
| 142 } | |
| 143 void SetProjectedPacketLossRate(double fraction) override { | |
| 144 return enc_->SetProjectedPacketLossRate(fraction); | |
| 145 } | |
| 146 void SetTargetBitrate(int target_bps) override { | |
| 147 return enc_->SetTargetBitrate(target_bps); | |
| 148 } | |
| 149 | |
| 150 private: | |
| 151 AudioEncoder* enc_; | |
| 152 }; | |
| 153 | |
| 154 // Return false on error. | |
| 155 bool CreateSpeechEncoderIfNecessary(EncoderFactory* ef) { | |
| 156 auto* sp = ef->codec_manager.GetStackParams(); | |
| 157 if (sp->speech_encoder) { | |
| 158 // Do nothing; we already have a speech encoder. | |
| 159 } else if (ef->codec_manager.GetCodecInst()) { | |
| 160 RTC_DCHECK(!ef->external_speech_encoder); | |
| 161 // We have no speech encoder, but we have a specification for making one. | |
| 162 std::unique_ptr<AudioEncoder> enc = | |
| 163 ef->rent_a_codec.RentEncoder(*ef->codec_manager.GetCodecInst()); | |
| 164 if (!enc) | |
| 165 return false; // Encoder spec was bad. | |
| 166 sp->speech_encoder = std::move(enc); | |
| 167 } else if (ef->external_speech_encoder) { | |
| 168 RTC_DCHECK(!ef->codec_manager.GetCodecInst()); | |
| 169 // We have an external speech encoder. | |
| 170 sp->speech_encoder = std::unique_ptr<AudioEncoder>( | |
| 171 new RawAudioEncoderWrapper(ef->external_speech_encoder)); | |
| 172 } | |
| 173 return true; | |
| 174 } | |
| 175 | |
| 176 } // namespace | |
| 177 | |
| 178 void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) { | |
| 179 if (value != last_value_ || first_time_) { | |
| 180 first_time_ = false; | |
| 181 last_value_ = value; | |
| 182 RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value); | |
| 183 } | |
| 184 } | |
| 185 | |
| 186 AudioCodingModuleImpl::AudioCodingModuleImpl( | |
| 187 const AudioCodingModule::Config& config) | |
| 188 : id_(config.id), | |
| 189 expected_codec_ts_(0xD87F3F9F), | |
| 190 expected_in_ts_(0xD87F3F9F), | |
| 191 receiver_(config), | |
| 192 bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"), | |
| 193 encoder_factory_(new EncoderFactory), | |
| 194 encoder_stack_(nullptr), | |
| 195 previous_pltype_(255), | |
| 196 receiver_initialized_(false), | |
| 197 first_10ms_data_(false), | |
| 198 first_frame_(true), | |
| 199 packetization_callback_(NULL), | |
| 200 vad_callback_(NULL), | |
| 201 codec_histogram_bins_log_(), | |
| 202 number_of_consecutive_empty_packets_(0) { | |
| 203 if (InitializeReceiverSafe() < 0) { | |
| 204 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 205 "Cannot initialize receiver"); | |
| 206 } | |
| 207 WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created"); | |
| 208 } | |
| 209 | |
| 210 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default; | |
| 211 | |
| 212 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) { | |
| 213 AudioEncoder::EncodedInfo encoded_info; | |
| 214 uint8_t previous_pltype; | |
| 215 | |
| 216 // Check if there is an encoder before. | |
| 217 if (!HaveValidEncoder("Process")) | |
| 218 return -1; | |
| 219 | |
| 220 // Scale the timestamp to the codec's RTP timestamp rate. | |
| 221 uint32_t rtp_timestamp = | |
| 222 first_frame_ ? input_data.input_timestamp | |
| 223 : last_rtp_timestamp_ + | |
| 224 rtc::CheckedDivExact( | |
| 225 input_data.input_timestamp - last_timestamp_, | |
| 226 static_cast<uint32_t>(rtc::CheckedDivExact( | |
| 227 encoder_stack_->SampleRateHz(), | |
| 228 encoder_stack_->RtpTimestampRateHz()))); | |
| 229 last_timestamp_ = input_data.input_timestamp; | |
| 230 last_rtp_timestamp_ = rtp_timestamp; | |
| 231 first_frame_ = false; | |
| 232 | |
| 233 // Clear the buffer before reuse - encoded data will get appended. | |
| 234 encode_buffer_.Clear(); | |
| 235 encoded_info = encoder_stack_->Encode( | |
| 236 rtp_timestamp, rtc::ArrayView<const int16_t>( | |
| 237 input_data.audio, input_data.audio_channel * | |
| 238 input_data.length_per_channel), | |
| 239 &encode_buffer_); | |
| 240 | |
| 241 bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000); | |
| 242 if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) { | |
| 243 // Not enough data. | |
| 244 return 0; | |
| 245 } | |
| 246 previous_pltype = previous_pltype_; // Read it while we have the critsect. | |
| 247 | |
| 248 // Log codec type to histogram once every 500 packets. | |
| 249 if (encoded_info.encoded_bytes == 0) { | |
| 250 ++number_of_consecutive_empty_packets_; | |
| 251 } else { | |
| 252 size_t codec_type = static_cast<size_t>(encoded_info.encoder_type); | |
| 253 codec_histogram_bins_log_[codec_type] += | |
| 254 number_of_consecutive_empty_packets_ + 1; | |
| 255 number_of_consecutive_empty_packets_ = 0; | |
| 256 if (codec_histogram_bins_log_[codec_type] >= 500) { | |
| 257 codec_histogram_bins_log_[codec_type] -= 500; | |
| 258 UpdateCodecTypeHistogram(codec_type); | |
| 259 } | |
| 260 } | |
| 261 | |
| 262 RTPFragmentationHeader my_fragmentation; | |
| 263 ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation); | |
| 264 FrameType frame_type; | |
| 265 if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) { | |
| 266 frame_type = kEmptyFrame; | |
| 267 encoded_info.payload_type = previous_pltype; | |
| 268 } else { | |
| 269 RTC_DCHECK_GT(encode_buffer_.size(), 0u); | |
| 270 frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN; | |
| 271 } | |
| 272 | |
| 273 { | |
| 274 rtc::CritScope lock(&callback_crit_sect_); | |
| 275 if (packetization_callback_) { | |
| 276 packetization_callback_->SendData( | |
| 277 frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp, | |
| 278 encode_buffer_.data(), encode_buffer_.size(), | |
| 279 my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation | |
| 280 : nullptr); | |
| 281 } | |
| 282 | |
| 283 if (vad_callback_) { | |
| 284 // Callback with VAD decision. | |
| 285 vad_callback_->InFrameType(frame_type); | |
| 286 } | |
| 287 } | |
| 288 previous_pltype_ = encoded_info.payload_type; | |
| 289 return static_cast<int32_t>(encode_buffer_.size()); | |
| 290 } | |
| 291 | |
| 292 ///////////////////////////////////////// | |
| 293 // Sender | |
| 294 // | |
| 295 | |
| 296 // Can be called multiple times for Codec, CNG, RED. | |
| 297 int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) { | |
| 298 rtc::CritScope lock(&acm_crit_sect_); | |
| 299 if (!encoder_factory_->codec_manager.RegisterEncoder(send_codec)) { | |
| 300 return -1; | |
| 301 } | |
| 302 if (encoder_factory_->codec_manager.GetCodecInst()) { | |
| 303 encoder_factory_->external_speech_encoder = nullptr; | |
| 304 } | |
| 305 if (!CreateSpeechEncoderIfNecessary(encoder_factory_.get())) { | |
| 306 return -1; | |
| 307 } | |
| 308 auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 309 if (sp->speech_encoder) | |
| 310 encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp); | |
| 311 return 0; | |
| 312 } | |
| 313 | |
| 314 void AudioCodingModuleImpl::RegisterExternalSendCodec( | |
| 315 AudioEncoder* external_speech_encoder) { | |
| 316 rtc::CritScope lock(&acm_crit_sect_); | |
| 317 encoder_factory_->codec_manager.UnsetCodecInst(); | |
| 318 encoder_factory_->external_speech_encoder = external_speech_encoder; | |
| 319 RTC_CHECK(CreateSpeechEncoderIfNecessary(encoder_factory_.get())); | |
| 320 auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 321 RTC_CHECK(sp->speech_encoder); | |
| 322 encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp); | |
| 323 } | |
| 324 | |
| 325 void AudioCodingModuleImpl::ModifyEncoder( | |
| 326 FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { | |
| 327 rtc::CritScope lock(&acm_crit_sect_); | |
| 328 | |
| 329 // Wipe the encoder factory, so that everything that relies on it will fail. | |
| 330 // We don't want the complexity of supporting swapping back and forth. | |
| 331 if (encoder_factory_) { | |
| 332 encoder_factory_.reset(); | |
| 333 RTC_CHECK(!encoder_stack_); // Ensure we hadn't started using the factory. | |
| 334 } | |
| 335 | |
| 336 modifier(&encoder_stack_); | |
| 337 } | |
| 338 | |
| 339 // Get current send codec. | |
| 340 rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const { | |
| 341 rtc::CritScope lock(&acm_crit_sect_); | |
| 342 if (encoder_factory_) { | |
| 343 auto* ci = encoder_factory_->codec_manager.GetCodecInst(); | |
| 344 if (ci) { | |
| 345 return rtc::Optional<CodecInst>(*ci); | |
| 346 } | |
| 347 CreateSpeechEncoderIfNecessary(encoder_factory_.get()); | |
| 348 const std::unique_ptr<AudioEncoder>& enc = | |
| 349 encoder_factory_->codec_manager.GetStackParams()->speech_encoder; | |
| 350 if (enc) { | |
| 351 return rtc::Optional<CodecInst>(CodecManager::ForgeCodecInst(enc.get())); | |
| 352 } | |
| 353 return rtc::Optional<CodecInst>(); | |
| 354 } else { | |
| 355 return encoder_stack_ | |
| 356 ? rtc::Optional<CodecInst>( | |
| 357 CodecManager::ForgeCodecInst(encoder_stack_.get())) | |
| 358 : rtc::Optional<CodecInst>(); | |
| 359 } | |
| 360 } | |
| 361 | |
| 362 // Get current send frequency. | |
| 363 int AudioCodingModuleImpl::SendFrequency() const { | |
| 364 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, | |
| 365 "SendFrequency()"); | |
| 366 rtc::CritScope lock(&acm_crit_sect_); | |
| 367 | |
| 368 if (!encoder_stack_) { | |
| 369 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, | |
| 370 "SendFrequency Failed, no codec is registered"); | |
| 371 return -1; | |
| 372 } | |
| 373 | |
| 374 return encoder_stack_->SampleRateHz(); | |
| 375 } | |
| 376 | |
| 377 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) { | |
| 378 rtc::CritScope lock(&acm_crit_sect_); | |
| 379 if (encoder_stack_) { | |
| 380 encoder_stack_->SetTargetBitrate(bitrate_bps); | |
| 381 } | |
| 382 } | |
| 383 | |
| 384 // Register a transport callback which will be called to deliver | |
| 385 // the encoded buffers. | |
| 386 int AudioCodingModuleImpl::RegisterTransportCallback( | |
| 387 AudioPacketizationCallback* transport) { | |
| 388 rtc::CritScope lock(&callback_crit_sect_); | |
| 389 packetization_callback_ = transport; | |
| 390 return 0; | |
| 391 } | |
| 392 | |
| 393 // Add 10MS of raw (PCM) audio data to the encoder. | |
| 394 int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) { | |
| 395 InputData input_data; | |
| 396 rtc::CritScope lock(&acm_crit_sect_); | |
| 397 int r = Add10MsDataInternal(audio_frame, &input_data); | |
| 398 return r < 0 ? r : Encode(input_data); | |
| 399 } | |
| 400 | |
| 401 int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame, | |
| 402 InputData* input_data) { | |
| 403 if (audio_frame.samples_per_channel_ == 0) { | |
| 404 assert(false); | |
| 405 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 406 "Cannot Add 10 ms audio, payload length is zero"); | |
| 407 return -1; | |
| 408 } | |
| 409 | |
| 410 if (audio_frame.sample_rate_hz_ > 48000) { | |
| 411 assert(false); | |
| 412 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 413 "Cannot Add 10 ms audio, input frequency not valid"); | |
| 414 return -1; | |
| 415 } | |
| 416 | |
| 417 // If the length and frequency matches. We currently just support raw PCM. | |
| 418 if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) != | |
| 419 audio_frame.samples_per_channel_) { | |
| 420 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 421 "Cannot Add 10 ms audio, input frequency and length doesn't" | |
| 422 " match"); | |
| 423 return -1; | |
| 424 } | |
| 425 | |
| 426 if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) { | |
| 427 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 428 "Cannot Add 10 ms audio, invalid number of channels."); | |
| 429 return -1; | |
| 430 } | |
| 431 | |
| 432 // Do we have a codec registered? | |
| 433 if (!HaveValidEncoder("Add10MsData")) { | |
| 434 return -1; | |
| 435 } | |
| 436 | |
| 437 const AudioFrame* ptr_frame; | |
| 438 // Perform a resampling, also down-mix if it is required and can be | |
| 439 // performed before resampling (a down mix prior to resampling will take | |
| 440 // place if both primary and secondary encoders are mono and input is in | |
| 441 // stereo). | |
| 442 if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) { | |
| 443 return -1; | |
| 444 } | |
| 445 | |
| 446 // Check whether we need an up-mix or down-mix? | |
| 447 const size_t current_num_channels = encoder_stack_->NumChannels(); | |
| 448 const bool same_num_channels = | |
| 449 ptr_frame->num_channels_ == current_num_channels; | |
| 450 | |
| 451 if (!same_num_channels) { | |
| 452 if (ptr_frame->num_channels_ == 1) { | |
| 453 if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) | |
| 454 return -1; | |
| 455 } else { | |
| 456 if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) | |
| 457 return -1; | |
| 458 } | |
| 459 } | |
| 460 | |
| 461 // When adding data to encoders this pointer is pointing to an audio buffer | |
| 462 // with correct number of channels. | |
| 463 const int16_t* ptr_audio = ptr_frame->data_; | |
| 464 | |
| 465 // For pushing data to primary, point the |ptr_audio| to correct buffer. | |
| 466 if (!same_num_channels) | |
| 467 ptr_audio = input_data->buffer; | |
| 468 | |
| 469 input_data->input_timestamp = ptr_frame->timestamp_; | |
| 470 input_data->audio = ptr_audio; | |
| 471 input_data->length_per_channel = ptr_frame->samples_per_channel_; | |
| 472 input_data->audio_channel = current_num_channels; | |
| 473 | |
| 474 return 0; | |
| 475 } | |
| 476 | |
| 477 // Perform a resampling and down-mix if required. We down-mix only if | |
| 478 // encoder is mono and input is stereo. In case of dual-streaming, both | |
| 479 // encoders has to be mono for down-mix to take place. | |
| 480 // |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing | |
| 481 // is required, |*ptr_out| points to |in_frame|. | |
| 482 int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, | |
| 483 const AudioFrame** ptr_out) { | |
| 484 const bool resample = | |
| 485 in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz(); | |
| 486 | |
| 487 // This variable is true if primary codec and secondary codec (if exists) | |
| 488 // are both mono and input is stereo. | |
| 489 // TODO(henrik.lundin): This condition should probably be | |
| 490 // in_frame.num_channels_ > encoder_stack_->NumChannels() | |
| 491 const bool down_mix = | |
| 492 in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1; | |
| 493 | |
| 494 if (!first_10ms_data_) { | |
| 495 expected_in_ts_ = in_frame.timestamp_; | |
| 496 expected_codec_ts_ = in_frame.timestamp_; | |
| 497 first_10ms_data_ = true; | |
| 498 } else if (in_frame.timestamp_ != expected_in_ts_) { | |
| 499 // TODO(turajs): Do we need a warning here. | |
| 500 expected_codec_ts_ += | |
| 501 (in_frame.timestamp_ - expected_in_ts_) * | |
| 502 static_cast<uint32_t>( | |
| 503 static_cast<double>(encoder_stack_->SampleRateHz()) / | |
| 504 static_cast<double>(in_frame.sample_rate_hz_)); | |
| 505 expected_in_ts_ = in_frame.timestamp_; | |
| 506 } | |
| 507 | |
| 508 | |
| 509 if (!down_mix && !resample) { | |
| 510 // No pre-processing is required. | |
| 511 expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_); | |
| 512 expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_); | |
| 513 *ptr_out = &in_frame; | |
| 514 return 0; | |
| 515 } | |
| 516 | |
| 517 *ptr_out = &preprocess_frame_; | |
| 518 preprocess_frame_.num_channels_ = in_frame.num_channels_; | |
| 519 int16_t audio[WEBRTC_10MS_PCM_AUDIO]; | |
| 520 const int16_t* src_ptr_audio = in_frame.data_; | |
| 521 int16_t* dest_ptr_audio = preprocess_frame_.data_; | |
| 522 if (down_mix) { | |
| 523 // If a resampling is required the output of a down-mix is written into a | |
| 524 // local buffer, otherwise, it will be written to the output frame. | |
| 525 if (resample) | |
| 526 dest_ptr_audio = audio; | |
| 527 if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0) | |
| 528 return -1; | |
| 529 preprocess_frame_.num_channels_ = 1; | |
| 530 // Set the input of the resampler is the down-mixed signal. | |
| 531 src_ptr_audio = audio; | |
| 532 } | |
| 533 | |
| 534 preprocess_frame_.timestamp_ = expected_codec_ts_; | |
| 535 preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; | |
| 536 preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; | |
| 537 // If it is required, we have to do a resampling. | |
| 538 if (resample) { | |
| 539 // The result of the resampler is written to output frame. | |
| 540 dest_ptr_audio = preprocess_frame_.data_; | |
| 541 | |
| 542 int samples_per_channel = resampler_.Resample10Msec( | |
| 543 src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(), | |
| 544 preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples, | |
| 545 dest_ptr_audio); | |
| 546 | |
| 547 if (samples_per_channel < 0) { | |
| 548 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 549 "Cannot add 10 ms audio, resampling failed"); | |
| 550 return -1; | |
| 551 } | |
| 552 preprocess_frame_.samples_per_channel_ = | |
| 553 static_cast<size_t>(samples_per_channel); | |
| 554 preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz(); | |
| 555 } | |
| 556 | |
| 557 expected_codec_ts_ += | |
| 558 static_cast<uint32_t>(preprocess_frame_.samples_per_channel_); | |
| 559 expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_); | |
| 560 | |
| 561 return 0; | |
| 562 } | |
| 563 | |
| 564 ///////////////////////////////////////// | |
| 565 // (RED) Redundant Coding | |
| 566 // | |
| 567 | |
| 568 bool AudioCodingModuleImpl::REDStatus() const { | |
| 569 rtc::CritScope lock(&acm_crit_sect_); | |
| 570 return encoder_factory_->codec_manager.GetStackParams()->use_red; | |
| 571 } | |
| 572 | |
| 573 // Configure RED status i.e on/off. | |
| 574 int AudioCodingModuleImpl::SetREDStatus(bool enable_red) { | |
| 575 #ifdef WEBRTC_CODEC_RED | |
| 576 rtc::CritScope lock(&acm_crit_sect_); | |
| 577 CreateSpeechEncoderIfNecessary(encoder_factory_.get()); | |
| 578 if (!encoder_factory_->codec_manager.SetCopyRed(enable_red)) { | |
| 579 return -1; | |
| 580 } | |
| 581 auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 582 if (sp->speech_encoder) | |
| 583 encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp); | |
| 584 return 0; | |
| 585 #else | |
| 586 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_, | |
| 587 " WEBRTC_CODEC_RED is undefined"); | |
| 588 return -1; | |
| 589 #endif | |
| 590 } | |
| 591 | |
| 592 ///////////////////////////////////////// | |
| 593 // (FEC) Forward Error Correction (codec internal) | |
| 594 // | |
| 595 | |
| 596 bool AudioCodingModuleImpl::CodecFEC() const { | |
| 597 rtc::CritScope lock(&acm_crit_sect_); | |
| 598 return encoder_factory_->codec_manager.GetStackParams()->use_codec_fec; | |
| 599 } | |
| 600 | |
| 601 int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) { | |
| 602 rtc::CritScope lock(&acm_crit_sect_); | |
| 603 CreateSpeechEncoderIfNecessary(encoder_factory_.get()); | |
| 604 if (!encoder_factory_->codec_manager.SetCodecFEC(enable_codec_fec)) { | |
| 605 return -1; | |
| 606 } | |
| 607 auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 608 if (sp->speech_encoder) | |
| 609 encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp); | |
| 610 if (enable_codec_fec) { | |
| 611 return sp->use_codec_fec ? 0 : -1; | |
| 612 } else { | |
| 613 RTC_DCHECK(!sp->use_codec_fec); | |
| 614 return 0; | |
| 615 } | |
| 616 } | |
| 617 | |
| 618 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { | |
| 619 rtc::CritScope lock(&acm_crit_sect_); | |
| 620 if (HaveValidEncoder("SetPacketLossRate")) { | |
| 621 encoder_stack_->SetProjectedPacketLossRate(loss_rate / 100.0); | |
| 622 } | |
| 623 return 0; | |
| 624 } | |
| 625 | |
| 626 ///////////////////////////////////////// | |
| 627 // (VAD) Voice Activity Detection | |
| 628 // | |
| 629 int AudioCodingModuleImpl::SetVAD(bool enable_dtx, | |
| 630 bool enable_vad, | |
| 631 ACMVADMode mode) { | |
| 632 // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting. | |
| 633 RTC_DCHECK_EQ(enable_dtx, enable_vad); | |
| 634 rtc::CritScope lock(&acm_crit_sect_); | |
| 635 CreateSpeechEncoderIfNecessary(encoder_factory_.get()); | |
| 636 if (!encoder_factory_->codec_manager.SetVAD(enable_dtx, mode)) { | |
| 637 return -1; | |
| 638 } | |
| 639 auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 640 if (sp->speech_encoder) | |
| 641 encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp); | |
| 642 return 0; | |
| 643 } | |
| 644 | |
| 645 // Get VAD/DTX settings. | |
| 646 int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled, | |
| 647 ACMVADMode* mode) const { | |
| 648 rtc::CritScope lock(&acm_crit_sect_); | |
| 649 const auto* sp = encoder_factory_->codec_manager.GetStackParams(); | |
| 650 *dtx_enabled = *vad_enabled = sp->use_cng; | |
| 651 *mode = sp->vad_mode; | |
| 652 return 0; | |
| 653 } | |
| 654 | |
| 655 ///////////////////////////////////////// | |
| 656 // Receiver | |
| 657 // | |
| 658 | |
| 659 int AudioCodingModuleImpl::InitializeReceiver() { | |
| 660 rtc::CritScope lock(&acm_crit_sect_); | |
| 661 return InitializeReceiverSafe(); | |
| 662 } | |
| 663 | |
| 664 // Initialize receiver, resets codec database etc. | |
| 665 int AudioCodingModuleImpl::InitializeReceiverSafe() { | |
| 666 // If the receiver is already initialized then we want to destroy any | |
| 667 // existing decoders. After a call to this function, we should have a clean | |
| 668 // start-up. | |
| 669 if (receiver_initialized_) { | |
| 670 if (receiver_.RemoveAllCodecs() < 0) | |
| 671 return -1; | |
| 672 } | |
| 673 receiver_.ResetInitialDelay(); | |
| 674 receiver_.SetMinimumDelay(0); | |
| 675 receiver_.SetMaximumDelay(0); | |
| 676 receiver_.FlushBuffers(); | |
| 677 | |
| 678 // Register RED and CN. | |
| 679 auto db = RentACodec::Database(); | |
| 680 for (size_t i = 0; i < db.size(); i++) { | |
| 681 if (IsCodecRED(db[i]) || IsCodecCN(db[i])) { | |
| 682 if (receiver_.AddCodec(static_cast<int>(i), | |
| 683 static_cast<uint8_t>(db[i].pltype), 1, | |
| 684 db[i].plfreq, nullptr, db[i].plname) < 0) { | |
| 685 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 686 "Cannot register master codec."); | |
| 687 return -1; | |
| 688 } | |
| 689 } | |
| 690 } | |
| 691 receiver_initialized_ = true; | |
| 692 return 0; | |
| 693 } | |
| 694 | |
| 695 // Get current receive frequency. | |
| 696 int AudioCodingModuleImpl::ReceiveFrequency() const { | |
| 697 const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz(); | |
| 698 return last_packet_sample_rate ? *last_packet_sample_rate | |
| 699 : receiver_.last_output_sample_rate_hz(); | |
| 700 } | |
| 701 | |
| 702 // Get current playout frequency. | |
| 703 int AudioCodingModuleImpl::PlayoutFrequency() const { | |
| 704 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, | |
| 705 "PlayoutFrequency()"); | |
| 706 return receiver_.last_output_sample_rate_hz(); | |
| 707 } | |
| 708 | |
| 709 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) { | |
| 710 rtc::CritScope lock(&acm_crit_sect_); | |
| 711 auto* ef = encoder_factory_.get(); | |
| 712 return RegisterReceiveCodecUnlocked( | |
| 713 codec, [&] { return ef->rent_a_codec.RentIsacDecoder(codec.plfreq); }); | |
| 714 } | |
| 715 | |
| 716 int AudioCodingModuleImpl::RegisterReceiveCodec( | |
| 717 const CodecInst& codec, | |
| 718 FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) { | |
| 719 rtc::CritScope lock(&acm_crit_sect_); | |
| 720 return RegisterReceiveCodecUnlocked(codec, isac_factory); | |
| 721 } | |
| 722 | |
| 723 int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked( | |
| 724 const CodecInst& codec, | |
| 725 FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) { | |
| 726 RTC_DCHECK(receiver_initialized_); | |
| 727 if (codec.channels > 2) { | |
| 728 LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels; | |
| 729 return -1; | |
| 730 } | |
| 731 | |
| 732 auto codec_id = | |
| 733 RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels); | |
| 734 if (!codec_id) { | |
| 735 LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec"; | |
| 736 return -1; | |
| 737 } | |
| 738 auto codec_index = RentACodec::CodecIndexFromId(*codec_id); | |
| 739 RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id); | |
| 740 | |
| 741 // Check if the payload-type is valid. | |
| 742 if (!RentACodec::IsPayloadTypeValid(codec.pltype)) { | |
| 743 LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for " | |
| 744 << codec.plname; | |
| 745 return -1; | |
| 746 } | |
| 747 | |
| 748 AudioDecoder* isac_decoder = nullptr; | |
| 749 if (STR_CASE_CMP(codec.plname, "isac") == 0) { | |
| 750 std::unique_ptr<AudioDecoder>& saved_isac_decoder = | |
| 751 codec.plfreq == 16000 ? isac_decoder_16k_ : isac_decoder_32k_; | |
| 752 if (!saved_isac_decoder) { | |
| 753 saved_isac_decoder = isac_factory(); | |
| 754 } | |
| 755 isac_decoder = saved_isac_decoder.get(); | |
| 756 } | |
| 757 return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels, | |
| 758 codec.plfreq, isac_decoder, codec.plname); | |
| 759 } | |
| 760 | |
| 761 int AudioCodingModuleImpl::RegisterExternalReceiveCodec( | |
| 762 int rtp_payload_type, | |
| 763 AudioDecoder* external_decoder, | |
| 764 int sample_rate_hz, | |
| 765 int num_channels, | |
| 766 const std::string& name) { | |
| 767 rtc::CritScope lock(&acm_crit_sect_); | |
| 768 RTC_DCHECK(receiver_initialized_); | |
| 769 if (num_channels > 2 || num_channels < 0) { | |
| 770 LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels; | |
| 771 return -1; | |
| 772 } | |
| 773 | |
| 774 // Check if the payload-type is valid. | |
| 775 if (!RentACodec::IsPayloadTypeValid(rtp_payload_type)) { | |
| 776 LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type | |
| 777 << " for external decoder."; | |
| 778 return -1; | |
| 779 } | |
| 780 | |
| 781 return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels, | |
| 782 sample_rate_hz, external_decoder, name); | |
| 783 } | |
| 784 | |
| 785 // Get current received codec. | |
| 786 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const { | |
| 787 rtc::CritScope lock(&acm_crit_sect_); | |
| 788 return receiver_.LastAudioCodec(current_codec); | |
| 789 } | |
| 790 | |
| 791 // Incoming packet from network parsed and ready for decode. | |
| 792 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload, | |
| 793 const size_t payload_length, | |
| 794 const WebRtcRTPHeader& rtp_header) { | |
| 795 return receiver_.InsertPacket( | |
| 796 rtp_header, | |
| 797 rtc::ArrayView<const uint8_t>(incoming_payload, payload_length)); | |
| 798 } | |
| 799 | |
| 800 // Minimum playout delay (Used for lip-sync). | |
| 801 int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) { | |
| 802 if ((time_ms < 0) || (time_ms > 10000)) { | |
| 803 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 804 "Delay must be in the range of 0-1000 milliseconds."); | |
| 805 return -1; | |
| 806 } | |
| 807 return receiver_.SetMinimumDelay(time_ms); | |
| 808 } | |
| 809 | |
| 810 int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) { | |
| 811 if ((time_ms < 0) || (time_ms > 10000)) { | |
| 812 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 813 "Delay must be in the range of 0-1000 milliseconds."); | |
| 814 return -1; | |
| 815 } | |
| 816 return receiver_.SetMaximumDelay(time_ms); | |
| 817 } | |
| 818 | |
| 819 // Get 10 milliseconds of raw audio data to play out. | |
| 820 // Automatic resample to the requested frequency. | |
| 821 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, | |
| 822 AudioFrame* audio_frame, | |
| 823 bool* muted) { | |
| 824 // GetAudio always returns 10 ms, at the requested sample rate. | |
| 825 if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) { | |
| 826 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 827 "PlayoutData failed, RecOut Failed"); | |
| 828 return -1; | |
| 829 } | |
| 830 audio_frame->id_ = id_; | |
| 831 return 0; | |
| 832 } | |
| 833 | |
| 834 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, | |
| 835 AudioFrame* audio_frame) { | |
| 836 bool muted; | |
| 837 int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted); | |
| 838 RTC_DCHECK(!muted); | |
| 839 return ret; | |
| 840 } | |
| 841 | |
| 842 ///////////////////////////////////////// | |
| 843 // Statistics | |
| 844 // | |
| 845 | |
| 846 // TODO(turajs) change the return value to void. Also change the corresponding | |
| 847 // NetEq function. | |
| 848 int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) { | |
| 849 receiver_.GetNetworkStatistics(statistics); | |
| 850 return 0; | |
| 851 } | |
| 852 | |
| 853 int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) { | |
| 854 WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_, | |
| 855 "RegisterVADCallback()"); | |
| 856 rtc::CritScope lock(&callback_crit_sect_); | |
| 857 vad_callback_ = vad_callback; | |
| 858 return 0; | |
| 859 } | |
| 860 | |
| 861 // TODO(kwiberg): Remove this method, and have callers call IncomingPacket | |
| 862 // instead. The translation logic and state belong with them, not with | |
| 863 // AudioCodingModuleImpl. | |
| 864 int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload, | |
| 865 size_t payload_length, | |
| 866 uint8_t payload_type, | |
| 867 uint32_t timestamp) { | |
| 868 // We are not acquiring any lock when interacting with |aux_rtp_header_| no | |
| 869 // other method uses this member variable. | |
| 870 if (!aux_rtp_header_) { | |
| 871 // This is the first time that we are using |dummy_rtp_header_| | |
| 872 // so we have to create it. | |
| 873 aux_rtp_header_.reset(new WebRtcRTPHeader); | |
| 874 aux_rtp_header_->header.payloadType = payload_type; | |
| 875 // Don't matter in this case. | |
| 876 aux_rtp_header_->header.ssrc = 0; | |
| 877 aux_rtp_header_->header.markerBit = false; | |
| 878 // Start with random numbers. | |
| 879 aux_rtp_header_->header.sequenceNumber = 0x1234; // Arbitrary. | |
| 880 aux_rtp_header_->type.Audio.channel = 1; | |
| 881 } | |
| 882 | |
| 883 aux_rtp_header_->header.timestamp = timestamp; | |
| 884 IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_); | |
| 885 // Get ready for the next payload. | |
| 886 aux_rtp_header_->header.sequenceNumber++; | |
| 887 return 0; | |
| 888 } | |
| 889 | |
| 890 int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) { | |
| 891 rtc::CritScope lock(&acm_crit_sect_); | |
| 892 if (!HaveValidEncoder("SetOpusApplication")) { | |
| 893 return -1; | |
| 894 } | |
| 895 AudioEncoder::Application app; | |
| 896 switch (application) { | |
| 897 case kVoip: | |
| 898 app = AudioEncoder::Application::kSpeech; | |
| 899 break; | |
| 900 case kAudio: | |
| 901 app = AudioEncoder::Application::kAudio; | |
| 902 break; | |
| 903 default: | |
| 904 FATAL(); | |
| 905 return 0; | |
| 906 } | |
| 907 return encoder_stack_->SetApplication(app) ? 0 : -1; | |
| 908 } | |
| 909 | |
| 910 // Informs Opus encoder of the maximum playback rate the receiver will render. | |
| 911 int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) { | |
| 912 rtc::CritScope lock(&acm_crit_sect_); | |
| 913 if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) { | |
| 914 return -1; | |
| 915 } | |
| 916 encoder_stack_->SetMaxPlaybackRate(frequency_hz); | |
| 917 return 0; | |
| 918 } | |
| 919 | |
| 920 int AudioCodingModuleImpl::EnableOpusDtx() { | |
| 921 rtc::CritScope lock(&acm_crit_sect_); | |
| 922 if (!HaveValidEncoder("EnableOpusDtx")) { | |
| 923 return -1; | |
| 924 } | |
| 925 return encoder_stack_->SetDtx(true) ? 0 : -1; | |
| 926 } | |
| 927 | |
| 928 int AudioCodingModuleImpl::DisableOpusDtx() { | |
| 929 rtc::CritScope lock(&acm_crit_sect_); | |
| 930 if (!HaveValidEncoder("DisableOpusDtx")) { | |
| 931 return -1; | |
| 932 } | |
| 933 return encoder_stack_->SetDtx(false) ? 0 : -1; | |
| 934 } | |
| 935 | |
| 936 int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) { | |
| 937 rtc::Optional<uint32_t> ts = PlayoutTimestamp(); | |
| 938 if (!ts) | |
| 939 return -1; | |
| 940 *timestamp = *ts; | |
| 941 return 0; | |
| 942 } | |
| 943 | |
| 944 rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() { | |
| 945 return receiver_.GetPlayoutTimestamp(); | |
| 946 } | |
| 947 | |
| 948 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { | |
| 949 if (!encoder_stack_) { | |
| 950 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, | |
| 951 "%s failed: No send codec is registered.", caller_name); | |
| 952 return false; | |
| 953 } | |
| 954 return true; | |
| 955 } | |
| 956 | |
| 957 int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { | |
| 958 return receiver_.RemoveCodec(payload_type); | |
| 959 } | |
| 960 | |
| 961 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) { | |
| 962 return receiver_.EnableNack(max_nack_list_size); | |
| 963 } | |
| 964 | |
| 965 void AudioCodingModuleImpl::DisableNack() { | |
| 966 receiver_.DisableNack(); | |
| 967 } | |
| 968 | |
| 969 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList( | |
| 970 int64_t round_trip_time_ms) const { | |
| 971 return receiver_.GetNackList(round_trip_time_ms); | |
| 972 } | |
| 973 | |
| 974 int AudioCodingModuleImpl::LeastRequiredDelayMs() const { | |
| 975 return receiver_.LeastRequiredDelayMs(); | |
| 976 } | |
| 977 | |
| 978 void AudioCodingModuleImpl::GetDecodingCallStatistics( | |
| 979 AudioDecodingCallStats* call_stats) const { | |
| 980 receiver_.GetDecodingCallStatistics(call_stats); | |
| 981 } | |
| 982 | |
| 983 } // namespace acm2 | |
| 984 } // namespace webrtc | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2069723003:
Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master