webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

/*
 *  Copyright (c) 2014 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 "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"

#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"

namespace webrtc {

namespace {

const int kMinBitrateBps = 500;
const int kMaxBitrateBps = 512000;

// TODO(tlegrand): Remove this code when we have proper APIs to set the
// complexity at a higher level.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
// default, to save encoder complexity.
const int kDefaultComplexity = 5;
#else
const int kDefaultComplexity = 9;
#endif

// We always encode at 48 kHz.
const int kSampleRateHz = 48000;

int16_t ClampInt16(size_t x) {
  return static_cast<int16_t>(
      std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
}

int16_t CastInt16(size_t x) {
  DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
  return static_cast<int16_t>(x);
}

}  // namespace

AudioEncoderOpus::Config::Config()
    : frame_size_ms(20),
      num_channels(1),
      payload_type(120),
      application(kVoip),
      bitrate_bps(64000),
      fec_enabled(false),
      max_playback_rate_hz(48000),
      complexity(kDefaultComplexity) {
}

bool AudioEncoderOpus::Config::IsOk() const {
  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
    return false;
  if (num_channels != 1 && num_channels != 2)
    return false;
  if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
    return false;
  if (complexity < 0 || complexity > 10)
    return false;
  return true;
}

AudioEncoderOpus::AudioEncoderOpus(const Config& config)
    : num_10ms_frames_per_packet_(
          rtc::CheckedDivExact(config.frame_size_ms, 10)),
      num_channels_(config.num_channels),
      payload_type_(config.payload_type),
      application_(config.application),
      samples_per_10ms_frame_(rtc::CheckedDivExact(kSampleRateHz, 100) *
                              num_channels_),
      packet_loss_rate_(0.0) {
  CHECK(config.IsOk());
  input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
  SetTargetBitrate(config.bitrate_bps);
  if (config.fec_enabled) {
    CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
  } else {
    CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
  }
  CHECK_EQ(0,
           WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
  CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));

}

AudioEncoderOpus::~AudioEncoderOpus() {
  CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
}

int AudioEncoderOpus::sample_rate_hz() const {
  return kSampleRateHz;
}

int AudioEncoderOpus::num_channels() const {
  return num_channels_;
}

int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
  return num_10ms_frames_per_packet_;
}

int AudioEncoderOpus::Max10MsFramesInAPacket() const {
  return num_10ms_frames_per_packet_;
}

void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
  CHECK_EQ(WebRtcOpus_SetBitRate(
               inst_, std::max(std::min(bits_per_second, kMaxBitrateBps),
                               kMinBitrateBps)),
           0);
}

void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
  DCHECK_GE(fraction, 0.0);
  DCHECK_LE(fraction, 1.0);
  // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
  // the input loss rate rounded down to various levels, because a robustly good
  // audio quality is achieved by lowering the packet loss down.
  // Additionally, to prevent toggling, margins are used, i.e., when jumping to
  // a loss rate from below, a higher threshold is used than jumping to the same
  // level from above.
  const double kPacketLossRate20 = 0.20;
  const double kPacketLossRate10 = 0.10;
  const double kPacketLossRate5 = 0.05;
  const double kPacketLossRate1 = 0.01;
  const double kLossRate20Margin = 0.02;
  const double kLossRate10Margin = 0.01;
  const double kLossRate5Margin = 0.01;
  double opt_loss_rate;
  if (fraction >=
      kPacketLossRate20 +
          kLossRate20Margin *
              (kPacketLossRate20 - packet_loss_rate_ > 0 ? 1 : -1)) {
    opt_loss_rate = kPacketLossRate20;
  } else if (fraction >=
             kPacketLossRate10 +
                 kLossRate10Margin *
                     (kPacketLossRate10 - packet_loss_rate_ > 0 ? 1 : -1)) {
    opt_loss_rate = kPacketLossRate10;
  } else if (fraction >=
             kPacketLossRate5 +
                 kLossRate5Margin *
                     (kPacketLossRate5 - packet_loss_rate_ > 0 ? 1 : -1)) {
    opt_loss_rate = kPacketLossRate5;
  } else if (fraction >= kPacketLossRate1) {
    opt_loss_rate = kPacketLossRate1;
  } else {
    opt_loss_rate = 0;
  }

  if (packet_loss_rate_ != opt_loss_rate) {
    // Ask the encoder to change the target packet loss rate.
    CHECK_EQ(WebRtcOpus_SetPacketLossRate(
                 inst_, static_cast<int32_t>(opt_loss_rate * 100 + .5)),
             0);
    packet_loss_rate_ = opt_loss_rate;
  }
}

bool AudioEncoderOpus::EncodeInternal(uint32_t rtp_timestamp,
                                      const int16_t* audio,
                                      size_t max_encoded_bytes,
                                      uint8_t* encoded,
                                      EncodedInfo* info) {
  if (input_buffer_.empty())
    first_timestamp_in_buffer_ = rtp_timestamp;
  input_buffer_.insert(input_buffer_.end(), audio,
                       audio + samples_per_10ms_frame_);
  if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
                              samples_per_10ms_frame_)) {
    info->encoded_bytes = 0;
    return true;
  }
  CHECK_EQ(input_buffer_.size(),
           static_cast<size_t>(num_10ms_frames_per_packet_) *
           samples_per_10ms_frame_);
  int16_t r = WebRtcOpus_Encode(
      inst_, &input_buffer_[0],
      rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
                           static_cast<int16_t>(num_channels_)),
      ClampInt16(max_encoded_bytes), encoded);
  input_buffer_.clear();
  if (r < 0)
    return false;
  info->encoded_bytes = r;
  info->encoded_timestamp = first_timestamp_in_buffer_;
  info->payload_type = payload_type_;
  return true;
}

}  // namespace webrtc