modules/audio_coding/acm2/acm_receiver.cc

/*
 *  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>
#include <string.h>
#include <cstdint>
#include <vector>

#include "absl/strings/match.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/strings/audio_format_to_string.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,
         sizeof(int16_t) * 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;
}

absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
  rtc::CritScope lock(&crit_sect_);
  if (!last_decoder_) {
    return absl::nullopt;
  }
  return last_decoder_->second.clockrate_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) {
  if (incoming_payload.empty()) {
    neteq_->InsertEmptyPacket(rtp_header.header);
    return 0;
  }

  const RTPHeader& header = rtp_header.header;  // Just a shorthand.
  int payload_type = header.payloadType;
  auto format = neteq_->GetDecoderFormat(payload_type);
  if (format && absl::EqualsIgnoreCase(format->name, "red")) {
    // This is a RED packet. Get the format of the audio codec.
    payload_type = incoming_payload[0] & 0x7f;
    format = neteq_->GetDecoderFormat(payload_type);
  }
  if (!format) {
    RTC_LOG_F(LS_ERROR) << "Payload-type "
                        << payload_type
                        << " is not registered.";
    return -1;
  }

  {
    rtc::CritScope lock(&crit_sect_);
    if (absl::EqualsIgnoreCase(format->name, "cn")) {
      if (last_decoder_ && last_decoder_->second.num_channels > 1) {
        // This is a CNG and the audio codec is not mono, so skip pushing in
        // packets into NetEq.
        return 0;
      }
    } else {
      RTC_DCHECK(format);
      last_decoder_ = std::make_pair(payload_type, *format);
    }
  }  // |crit_sect_| is released.

  uint32_t receive_timestamp = NowInTimestamp(format->clockrate_hz);
  if (neteq_->InsertPacket(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);
  // Accessing members, take the lock.
  rtc::CritScope lock(&crit_sect_);

  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);
}

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

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

absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
  return neteq_->GetPlayoutTimestamp();
}

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

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

absl::optional<std::pair<int, SdpAudioFormat>>
    AcmReceiver::LastDecoder() const {
  rtc::CritScope lock(&crit_sect_);
  if (!last_decoder_) {
    return absl::nullopt;
  }
  RTC_DCHECK_NE(-1, last_decoder_->first);  // Payload type should be valid.
  return last_decoder_;
}

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;
  acm_stat->delayedPacketOutageSamples =
      neteq_lifetime_stat.delayed_packet_outage_samples;

  NetEqOperationsAndState neteq_operations_and_state =
      neteq_->GetOperationsAndState();
  acm_stat->packetBufferFlushes =
      neteq_operations_and_state.packet_buffer_flushes;
}

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?
}

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