LCOV - code coverage report
Current view: top level - media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng - audio_encoder_cng.cc (source / functions) Hit Total Coverage
Test: output.info Lines: 0 138 0.0 %
Date: 2017-07-14 16:53:18 Functions: 0 21 0.0 %
Legend: Lines: hit not hit

          Line data    Source code
       1             : /*
       2             :  *  Copyright (c) 2014 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/codecs/cng/audio_encoder_cng.h"
      12             : 
      13             : #include <algorithm>
      14             : #include <memory>
      15             : #include <limits>
      16             : #include <utility>
      17             : 
      18             : namespace webrtc {
      19             : 
      20             : namespace {
      21             : 
      22             : const int kMaxFrameSizeMs = 60;
      23             : 
      24             : }  // namespace
      25             : 
      26             : AudioEncoderCng::Config::Config() = default;
      27             : AudioEncoderCng::Config::Config(Config&&) = default;
      28             : AudioEncoderCng::Config::~Config() = default;
      29             : 
      30           0 : bool AudioEncoderCng::Config::IsOk() const {
      31           0 :   if (num_channels != 1)
      32           0 :     return false;
      33           0 :   if (!speech_encoder)
      34           0 :     return false;
      35           0 :   if (num_channels != speech_encoder->NumChannels())
      36           0 :     return false;
      37           0 :   if (sid_frame_interval_ms <
      38           0 :       static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
      39           0 :     return false;
      40           0 :   if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
      41           0 :       num_cng_coefficients <= 0)
      42           0 :     return false;
      43           0 :   return true;
      44             : }
      45             : 
      46           0 : AudioEncoderCng::AudioEncoderCng(Config&& config)
      47             :     : speech_encoder_(
      48           0 :           ([&] { RTC_CHECK(config.IsOk()) << "Invalid configuration."; }(),
      49           0 :            std::move(config.speech_encoder))),
      50           0 :       cng_payload_type_(config.payload_type),
      51           0 :       num_cng_coefficients_(config.num_cng_coefficients),
      52           0 :       sid_frame_interval_ms_(config.sid_frame_interval_ms),
      53             :       last_frame_active_(true),
      54           0 :       vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
      55             :            : CreateVad(config.vad_mode)),
      56           0 :       cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
      57           0 :                                            sid_frame_interval_ms_,
      58           0 :                                            num_cng_coefficients_)) {
      59           0 : }
      60             : 
      61             : AudioEncoderCng::~AudioEncoderCng() = default;
      62             : 
      63           0 : int AudioEncoderCng::SampleRateHz() const {
      64           0 :   return speech_encoder_->SampleRateHz();
      65             : }
      66             : 
      67           0 : size_t AudioEncoderCng::NumChannels() const {
      68           0 :   return 1;
      69             : }
      70             : 
      71           0 : int AudioEncoderCng::RtpTimestampRateHz() const {
      72           0 :   return speech_encoder_->RtpTimestampRateHz();
      73             : }
      74             : 
      75           0 : size_t AudioEncoderCng::Num10MsFramesInNextPacket() const {
      76           0 :   return speech_encoder_->Num10MsFramesInNextPacket();
      77             : }
      78             : 
      79           0 : size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
      80           0 :   return speech_encoder_->Max10MsFramesInAPacket();
      81             : }
      82             : 
      83           0 : int AudioEncoderCng::GetTargetBitrate() const {
      84           0 :   return speech_encoder_->GetTargetBitrate();
      85             : }
      86             : 
      87           0 : AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
      88             :     uint32_t rtp_timestamp,
      89             :     rtc::ArrayView<const int16_t> audio,
      90             :     rtc::Buffer* encoded) {
      91           0 :   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
      92           0 :   RTC_CHECK_EQ(speech_buffer_.size(),
      93           0 :                rtp_timestamps_.size() * samples_per_10ms_frame);
      94           0 :   rtp_timestamps_.push_back(rtp_timestamp);
      95           0 :   RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
      96           0 :   speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
      97           0 :   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
      98           0 :   if (rtp_timestamps_.size() < frames_to_encode) {
      99           0 :     return EncodedInfo();
     100             :   }
     101           0 :   RTC_CHECK_LE(frames_to_encode * 10, kMaxFrameSizeMs)
     102           0 :       << "Frame size cannot be larger than " << kMaxFrameSizeMs
     103           0 :       << " ms when using VAD/CNG.";
     104             : 
     105             :   // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
     106             :   // following split sizes:
     107             :   // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
     108             :   // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
     109             :   size_t blocks_in_first_vad_call =
     110           0 :       (frames_to_encode > 3 ? 3 : frames_to_encode);
     111           0 :   if (frames_to_encode == 4)
     112           0 :     blocks_in_first_vad_call = 2;
     113           0 :   RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
     114             :   const size_t blocks_in_second_vad_call =
     115           0 :       frames_to_encode - blocks_in_first_vad_call;
     116             : 
     117             :   // Check if all of the buffer is passive speech. Start with checking the first
     118             :   // block.
     119           0 :   Vad::Activity activity = vad_->VoiceActivity(
     120           0 :       &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call,
     121           0 :       SampleRateHz());
     122           0 :   if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) {
     123             :     // Only check the second block if the first was passive.
     124           0 :     activity = vad_->VoiceActivity(
     125           0 :         &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call],
     126           0 :         samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz());
     127             :   }
     128             : 
     129           0 :   EncodedInfo info;
     130           0 :   switch (activity) {
     131             :     case Vad::kPassive: {
     132           0 :       info = EncodePassive(frames_to_encode, encoded);
     133           0 :       last_frame_active_ = false;
     134           0 :       break;
     135             :     }
     136             :     case Vad::kActive: {
     137           0 :       info = EncodeActive(frames_to_encode, encoded);
     138           0 :       last_frame_active_ = true;
     139           0 :       break;
     140             :     }
     141             :     case Vad::kError: {
     142           0 :       FATAL();  // Fails only if fed invalid data.
     143             :       break;
     144             :     }
     145             :   }
     146             : 
     147             :   speech_buffer_.erase(
     148           0 :       speech_buffer_.begin(),
     149           0 :       speech_buffer_.begin() + frames_to_encode * samples_per_10ms_frame);
     150           0 :   rtp_timestamps_.erase(rtp_timestamps_.begin(),
     151           0 :                         rtp_timestamps_.begin() + frames_to_encode);
     152           0 :   return info;
     153             : }
     154             : 
     155           0 : void AudioEncoderCng::Reset() {
     156           0 :   speech_encoder_->Reset();
     157           0 :   speech_buffer_.clear();
     158           0 :   rtp_timestamps_.clear();
     159           0 :   last_frame_active_ = true;
     160           0 :   vad_->Reset();
     161           0 :   cng_encoder_.reset(
     162           0 :       new ComfortNoiseEncoder(SampleRateHz(), sid_frame_interval_ms_,
     163           0 :                               num_cng_coefficients_));
     164           0 : }
     165             : 
     166           0 : bool AudioEncoderCng::SetFec(bool enable) {
     167           0 :   return speech_encoder_->SetFec(enable);
     168             : }
     169             : 
     170           0 : bool AudioEncoderCng::SetDtx(bool enable) {
     171           0 :   return speech_encoder_->SetDtx(enable);
     172             : }
     173             : 
     174           0 : bool AudioEncoderCng::SetApplication(Application application) {
     175           0 :   return speech_encoder_->SetApplication(application);
     176             : }
     177             : 
     178           0 : void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
     179           0 :   speech_encoder_->SetMaxPlaybackRate(frequency_hz);
     180           0 : }
     181             : 
     182             : rtc::ArrayView<std::unique_ptr<AudioEncoder>>
     183           0 : AudioEncoderCng::ReclaimContainedEncoders() {
     184           0 :   return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
     185             : }
     186             : 
     187           0 : void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
     188             :     float uplink_packet_loss_fraction) {
     189           0 :   speech_encoder_->OnReceivedUplinkPacketLossFraction(
     190           0 :       uplink_packet_loss_fraction);
     191           0 : }
     192             : 
     193           0 : void AudioEncoderCng::OnReceivedUplinkBandwidth(
     194             :     int target_audio_bitrate_bps,
     195             :     rtc::Optional<int64_t> probing_interval_ms) {
     196           0 :   speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
     197           0 :                                              probing_interval_ms);
     198           0 : }
     199             : 
     200           0 : AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     201             :     size_t frames_to_encode,
     202             :     rtc::Buffer* encoded) {
     203           0 :   bool force_sid = last_frame_active_;
     204           0 :   bool output_produced = false;
     205           0 :   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
     206           0 :   AudioEncoder::EncodedInfo info;
     207             : 
     208           0 :   for (size_t i = 0; i < frames_to_encode; ++i) {
     209             :     // It's important not to pass &info.encoded_bytes directly to
     210             :     // WebRtcCng_Encode(), since later loop iterations may return zero in
     211             :     // that value, in which case we don't want to overwrite any value from
     212             :     // an earlier iteration.
     213             :     size_t encoded_bytes_tmp =
     214           0 :         cng_encoder_->Encode(
     215             :             rtc::ArrayView<const int16_t>(
     216           0 :                 &speech_buffer_[i * samples_per_10ms_frame],
     217             :                 samples_per_10ms_frame),
     218           0 :             force_sid, encoded);
     219             : 
     220           0 :     if (encoded_bytes_tmp > 0) {
     221           0 :       RTC_CHECK(!output_produced);
     222           0 :       info.encoded_bytes = encoded_bytes_tmp;
     223           0 :       output_produced = true;
     224           0 :       force_sid = false;
     225             :     }
     226             :   }
     227             : 
     228           0 :   info.encoded_timestamp = rtp_timestamps_.front();
     229           0 :   info.payload_type = cng_payload_type_;
     230           0 :   info.send_even_if_empty = true;
     231           0 :   info.speech = false;
     232           0 :   return info;
     233             : }
     234             : 
     235           0 : AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
     236             :     size_t frames_to_encode,
     237             :     rtc::Buffer* encoded) {
     238           0 :   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
     239           0 :   AudioEncoder::EncodedInfo info;
     240           0 :   for (size_t i = 0; i < frames_to_encode; ++i) {
     241             :     info =
     242           0 :         speech_encoder_->Encode(rtp_timestamps_.front(),
     243             :                                 rtc::ArrayView<const int16_t>(
     244           0 :                                     &speech_buffer_[i * samples_per_10ms_frame],
     245             :                                     samples_per_10ms_frame),
     246           0 :                                 encoded);
     247           0 :     if (i + 1 == frames_to_encode) {
     248           0 :       RTC_CHECK_GT(info.encoded_bytes, 0) << "Encoder didn't deliver data.";
     249             :     } else {
     250           0 :       RTC_CHECK_EQ(info.encoded_bytes, 0)
     251           0 :           << "Encoder delivered data too early.";
     252             :     }
     253             :   }
     254           0 :   return info;
     255             : }
     256             : 
     257           0 : size_t AudioEncoderCng::SamplesPer10msFrame() const {
     258           0 :   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
     259             : }
     260             : 
     261             : }  // namespace webrtc

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