DXR is a code search and navigation tool aimed at making sense of large projects. It supports full-text and regex searches as well as structural queries.

Header

Mercurial (c68fe15a81fc)

VCS Links

Line Code
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418
/*
 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/audio_coding/acm2/acm_receiver.h"

#include <stdlib.h>  // malloc

#include <algorithm>  // sort
#include <vector>

#include "api/audio_codecs/audio_decoder.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "common_types.h"  // NOLINT(build/include)
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

namespace acm2 {

AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
    : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
      neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)),
      clock_(config.clock),
      resampled_last_output_frame_(true) {
  RTC_DCHECK(clock_);
  memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
}

AcmReceiver::~AcmReceiver() = default;

int AcmReceiver::SetMinimumDelay(int delay_ms) {
  if (neteq_->SetMinimumDelay(delay_ms))
    return 0;
  RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
  return -1;
}

int AcmReceiver::SetMaximumDelay(int delay_ms) {
  if (neteq_->SetMaximumDelay(delay_ms))
    return 0;
  RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
  return -1;
}

int AcmReceiver::LeastRequiredDelayMs() const {
  return neteq_->LeastRequiredDelayMs();
}

rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
  rtc::CritScope lock(&crit_sect_);
  return last_packet_sample_rate_hz_;
}

int AcmReceiver::last_output_sample_rate_hz() const {
  return neteq_->last_output_sample_rate_hz();
}

int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
                              rtc::ArrayView<const uint8_t> incoming_payload) {
  uint32_t receive_timestamp = 0;
  const RTPHeader* header = &rtp_header.header;  // Just a shorthand.

  if (incoming_payload.empty()) {
    neteq_->InsertEmptyPacket(rtp_header.header);
    return 0;
  }

  {
    rtc::CritScope lock(&crit_sect_);

    const rtc::Optional<CodecInst> ci =
        RtpHeaderToDecoder(*header, incoming_payload[0]);
    if (!ci) {
      RTC_LOG_F(LS_ERROR) << "Payload-type "
                          << static_cast<int>(header->payloadType)
                          << " is not registered.";
      return -1;
    }
    receive_timestamp = NowInTimestamp(ci->plfreq);

    if (STR_CASE_CMP(ci->plname, "cn") == 0) {
      if (last_audio_decoder_ && last_audio_decoder_->channels > 1) {
        // This is a CNG and the audio codec is not mono, so skip pushing in
        // packets into NetEq.
        return 0;
      }
    } else {
      last_audio_decoder_ = ci;
      last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
      last_audio_format_clockrate_hz_ = last_audio_format_->clockrate_hz;
      RTC_DCHECK(last_audio_format_);
      last_packet_sample_rate_hz_ = ci->plfreq;
    }
  }  // |crit_sect_| is released.

  if (neteq_->InsertPacket(rtp_header.header, incoming_payload,
                           receive_timestamp) < 0) {
    RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
                    << static_cast<int>(header->payloadType)
                    << " Failed to insert packet";
    return -1;
  }
  return 0;
}

int AcmReceiver::GetAudio(int desired_freq_hz,
                          AudioFrame* audio_frame,
                          bool* muted) {
  RTC_DCHECK(muted);

  if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
    RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
    return -1;
  }

  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();

  // Update if resampling is required.
  const bool need_resampling =
      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);

  if (need_resampling && !resampled_last_output_frame_) {
    // Prime the resampler with the last frame.
    int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
    int samples_per_channel_int = resampler_.Resample10Msec(
        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
        temp_output);
    if (samples_per_channel_int < 0) {
      RTC_LOG(LERROR) << "AcmReceiver::GetAudio - "
                         "Resampling last_audio_buffer_ failed.";
      return -1;
    }
  }

  // TODO(henrik.lundin) Glitches in the output may appear if the output rate
  // from NetEq changes. See WebRTC issue 3923.
  if (need_resampling) {
    // TODO(yujo): handle this more efficiently for muted frames.
    int samples_per_channel_int = resampler_.Resample10Msec(
        audio_frame->data(), current_sample_rate_hz, desired_freq_hz,
        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
        audio_frame->mutable_data());
    if (samples_per_channel_int < 0) {
      RTC_LOG(LERROR)
          << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
      return -1;
    }
    audio_frame->samples_per_channel_ =
        static_cast<size_t>(samples_per_channel_int);
    audio_frame->sample_rate_hz_ = desired_freq_hz;
    RTC_DCHECK_EQ(
        audio_frame->sample_rate_hz_,
        rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
    resampled_last_output_frame_ = true;
  } else {
    resampled_last_output_frame_ = false;
    // We might end up here ONLY if codec is changed.
  }

  // Store current audio in |last_audio_buffer_| for next time.
  memcpy(last_audio_buffer_.get(), audio_frame->data(),
         sizeof(int16_t) * audio_frame->samples_per_channel_ *
             audio_frame->num_channels_);

  call_stats_.DecodedByNetEq(audio_frame->speech_type_, *muted);
  return 0;
}

void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
  neteq_->SetCodecs(codecs);
}

int32_t AcmReceiver::AddCodec(int acm_codec_id,
                              uint8_t payload_type,
                              size_t channels,
                              int /*sample_rate_hz*/,
                              AudioDecoder* audio_decoder,
                              const std::string& name) {
  // TODO(kwiberg): This function has been ignoring the |sample_rate_hz|
  // argument for a long time. Arguably, it should simply be removed.

  const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
    if (acm_codec_id == -1)
      return NetEqDecoder::kDecoderArbitrary;  // External decoder.
    const rtc::Optional<RentACodec::CodecId> cid =
        RentACodec::CodecIdFromIndex(acm_codec_id);
    RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
    const rtc::Optional<NetEqDecoder> ned =
        RentACodec::NetEqDecoderFromCodecId(*cid, channels);
    RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
    return *ned;
  }();
  const rtc::Optional<SdpAudioFormat> new_format =
      NetEqDecoderToSdpAudioFormat(neteq_decoder);

  rtc::CritScope lock(&crit_sect_);

  const auto old_format = neteq_->GetDecoderFormat(payload_type);
  if (old_format && new_format && *old_format == *new_format) {
    // Re-registering the same codec. Do nothing and return.
    return 0;
  }

  if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
    RTC_LOG(LERROR) << "Cannot remove payload "
                    << static_cast<int>(payload_type);
    return -1;
  }

  int ret_val;
  if (!audio_decoder) {
    ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
  } else {
    ret_val = neteq_->RegisterExternalDecoder(
        audio_decoder, neteq_decoder, name, payload_type);
  }
  if (ret_val != NetEq::kOK) {
    RTC_LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
                    << static_cast<int>(payload_type)
                    << " channels: " << channels;
    return -1;
  }
  return 0;
}

bool AcmReceiver::AddCodec(int rtp_payload_type,
                           const SdpAudioFormat& audio_format) {
  const auto old_format = neteq_->GetDecoderFormat(rtp_payload_type);
  if (old_format && *old_format == audio_format) {
    // Re-registering the same codec. Do nothing and return.
    return true;
  }

  if (neteq_->RemovePayloadType(rtp_payload_type) != NetEq::kOK) {
    RTC_LOG(LERROR)
        << "AcmReceiver::AddCodec: Could not remove existing decoder"
           " for payload type "
        << rtp_payload_type;
    return false;
  }

  const bool success =
      neteq_->RegisterPayloadType(rtp_payload_type, audio_format);
  if (!success) {
    RTC_LOG(LERROR) << "AcmReceiver::AddCodec failed for payload type "
                    << rtp_payload_type << ", decoder format " << audio_format;
  }
  return success;
}

void AcmReceiver::FlushBuffers() {
  neteq_->FlushBuffers();
}

void AcmReceiver::RemoveAllCodecs() {
  rtc::CritScope lock(&crit_sect_);
  neteq_->RemoveAllPayloadTypes();
  last_audio_decoder_ = rtc::nullopt;
  last_audio_format_ = rtc::nullopt;
  last_packet_sample_rate_hz_ = rtc::nullopt;
}

int AcmReceiver::RemoveCodec(uint8_t payload_type) {
  rtc::CritScope lock(&crit_sect_);
  if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
    RTC_LOG(LERROR) << "AcmReceiver::RemoveCodec "
                    << static_cast<int>(payload_type);
    return -1;
  }
  if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
    last_audio_decoder_ = rtc::nullopt;
    last_audio_format_ = rtc::nullopt;
    last_packet_sample_rate_hz_ = rtc::nullopt;
  }
  return 0;
}

rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
  return neteq_->GetPlayoutTimestamp();
}

int AcmReceiver::FilteredCurrentDelayMs() const {
  return neteq_->FilteredCurrentDelayMs();
}

int AcmReceiver::TargetDelayMs() const {
  return neteq_->TargetDelayMs();
}

int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
  rtc::CritScope lock(&crit_sect_);
  if (!last_audio_decoder_) {
    return -1;
  }
  *codec = *last_audio_decoder_;
  return 0;
}

rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
  rtc::CritScope lock(&crit_sect_);
  return last_audio_format_;
}

int AcmReceiver::LastAudioSampleRate() const {
  return last_audio_format_clockrate_hz_;
}

void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
  NetEqNetworkStatistics neteq_stat;
  // NetEq function always returns zero, so we don't check the return value.
  neteq_->NetworkStatistics(&neteq_stat);

  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
  acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
  acm_stat->currentExpandRate = neteq_stat.expand_rate;
  acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
  acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
  acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
  acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
  acm_stat->currentSecondaryDiscardedRate = neteq_stat.secondary_discarded_rate;
  acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
  acm_stat->addedSamples = neteq_stat.added_zero_samples;
  acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
  acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
  acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
  acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;

  NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
  acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
  acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
  acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
  acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
}

int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
                                      CodecInst* codec) const {
  rtc::CritScope lock(&crit_sect_);
  const rtc::Optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
  if (ci) {
    *codec = *ci;
    return 0;
  } else {
    RTC_LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
                    << static_cast<int>(payload_type);
    return -1;
  }
}

int AcmReceiver::EnableNack(size_t max_nack_list_size) {
  neteq_->EnableNack(max_nack_list_size);
  return 0;
}

void AcmReceiver::DisableNack() {
  neteq_->DisableNack();
}

std::vector<uint16_t> AcmReceiver::GetNackList(
    int64_t round_trip_time_ms) const {
  return neteq_->GetNackList(round_trip_time_ms);
}

void AcmReceiver::ResetInitialDelay() {
  neteq_->SetMinimumDelay(0);
  // TODO(turajs): Should NetEq Buffer be flushed?
}

const rtc::Optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
    const RTPHeader& rtp_header,
    uint8_t first_payload_byte) const {
  const rtc::Optional<CodecInst> ci =
      neteq_->GetDecoder(rtp_header.payloadType);
  if (ci && STR_CASE_CMP(ci->plname, "red") == 0) {
    // This is a RED packet. Get the payload of the audio codec.
    return neteq_->GetDecoder(first_payload_byte & 0x7f);
  } else {
    return ci;
  }
}

uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
  // Down-cast the time to (32-6)-bit since we only care about
  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
  // We masked 6 most significant bits of 32-bit so there is no overflow in
  // the conversion from milliseconds to timestamp.
  const uint32_t now_in_ms = static_cast<uint32_t>(
      clock_->TimeInMilliseconds() & 0x03ffffff);
  return static_cast<uint32_t>(
      (decoder_sampling_rate / 1000) * now_in_ms);
}

void AcmReceiver::GetDecodingCallStatistics(
    AudioDecodingCallStats* stats) const {
  rtc::CritScope lock(&crit_sect_);
  *stats = call_stats_.GetDecodingStatistics();
}

}  // namespace acm2

}  // namespace webrtc