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audio_encoder_pcm.cc (4206B)

      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 "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
     12 
     13 #include <cstddef>
     14 #include <cstdint>
     15 #include <optional>
     16 #include <utility>
     17 
     18 #include "api/array_view.h"
     19 #include "api/audio_codecs/audio_encoder.h"
     20 #include "api/units/time_delta.h"
     21 #include "modules/audio_coding/codecs/g711/g711_interface.h"
     22 #include "rtc_base/buffer.h"
     23 #include "rtc_base/checks.h"
     24 
     25 namespace webrtc {
     26 
     27 bool AudioEncoderPcm::Config::IsOk() const {
     28  return (frame_size_ms % 10 == 0) && (num_channels >= 1) &&
     29         (num_channels <= AudioEncoder::kMaxNumberOfChannels);
     30 }
     31 
     32 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     33    : sample_rate_hz_(sample_rate_hz),
     34      num_channels_(config.num_channels),
     35      payload_type_(config.payload_type),
     36      num_10ms_frames_per_packet_(
     37          static_cast<size_t>(config.frame_size_ms / 10)),
     38      full_frame_samples_(config.num_channels * config.frame_size_ms *
     39                          sample_rate_hz / 1000),
     40      first_timestamp_in_buffer_(0) {
     41  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
     42  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
     43      << "Frame size must be an integer multiple of 10 ms.";
     44  speech_buffer_.reserve(full_frame_samples_);
     45 }
     46 
     47 AudioEncoderPcm::~AudioEncoderPcm() = default;
     48 
     49 int AudioEncoderPcm::SampleRateHz() const {
     50  return sample_rate_hz_;
     51 }
     52 
     53 size_t AudioEncoderPcm::NumChannels() const {
     54  return num_channels_;
     55 }
     56 
     57 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
     58  return num_10ms_frames_per_packet_;
     59 }
     60 
     61 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
     62  return num_10ms_frames_per_packet_;
     63 }
     64 
     65 int AudioEncoderPcm::GetTargetBitrate() const {
     66  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
     67                          NumChannels());
     68 }
     69 
     70 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
     71    uint32_t rtp_timestamp,
     72    ArrayView<const int16_t> audio,
     73    Buffer* encoded) {
     74  if (speech_buffer_.empty()) {
     75    first_timestamp_in_buffer_ = rtp_timestamp;
     76  }
     77  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
     78  if (speech_buffer_.size() < full_frame_samples_) {
     79    return EncodedInfo();
     80  }
     81  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
     82  EncodedInfo info;
     83  info.encoded_timestamp = first_timestamp_in_buffer_;
     84  info.payload_type = payload_type_;
     85  info.encoded_bytes = encoded->AppendData(
     86      full_frame_samples_ * BytesPerSample(), [&](ArrayView<uint8_t> encoded) {
     87        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
     88                          encoded.data());
     89      });
     90  speech_buffer_.clear();
     91  info.encoder_type = GetCodecType();
     92  return info;
     93 }
     94 
     95 void AudioEncoderPcm::Reset() {
     96  speech_buffer_.clear();
     97 }
     98 
     99 std::optional<std::pair<TimeDelta, TimeDelta>>
    100 AudioEncoderPcm::GetFrameLengthRange() const {
    101  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
    102           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
    103 }
    104 
    105 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
    106                                    size_t input_len,
    107                                    uint8_t* encoded) {
    108  return WebRtcG711_EncodeA(audio, input_len, encoded);
    109 }
    110 
    111 size_t AudioEncoderPcmA::BytesPerSample() const {
    112  return 1;
    113 }
    114 
    115 AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
    116  return AudioEncoder::CodecType::kPcmA;
    117 }
    118 
    119 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
    120                                    size_t input_len,
    121                                    uint8_t* encoded) {
    122  return WebRtcG711_EncodeU(audio, input_len, encoded);
    123 }
    124 
    125 size_t AudioEncoderPcmU::BytesPerSample() const {
    126  return 1;
    127 }
    128 
    129 AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
    130  return AudioEncoder::CodecType::kPcmU;
    131 }
    132 
    133 }  // namespace webrtc