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audio_decoder_pcm16b.cc (2980B)


      1 /*
      2 *  Copyright (c) 2015 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/pcm16b/audio_decoder_pcm16b.h"
     12 
     13 #include <cstddef>
     14 #include <cstdint>
     15 #include <utility>
     16 #include <vector>
     17 
     18 #include "api/audio_codecs/audio_decoder.h"
     19 #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
     20 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
     21 #include "rtc_base/buffer.h"
     22 #include "rtc_base/checks.h"
     23 
     24 namespace webrtc {
     25 
     26 AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
     27    : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
     28  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
     29             sample_rate_hz == 32000 || sample_rate_hz == 48000)
     30      << "Unsupported sample rate " << sample_rate_hz;
     31  RTC_DCHECK_GE(num_channels, 1);
     32  RTC_DCHECK_LE(num_channels, AudioDecoder::kMaxNumberOfChannels);
     33 }
     34 
     35 void AudioDecoderPcm16B::Reset() {}
     36 
     37 int AudioDecoderPcm16B::SampleRateHz() const {
     38  return sample_rate_hz_;
     39 }
     40 
     41 size_t AudioDecoderPcm16B::Channels() const {
     42  return num_channels_;
     43 }
     44 
     45 int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
     46                                       size_t encoded_len,
     47                                       int sample_rate_hz,
     48                                       int16_t* decoded,
     49                                       SpeechType* speech_type) {
     50  RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
     51  // Adjust the encoded length down to ensure the same number of samples in each
     52  // channel.
     53  const size_t encoded_len_adjusted =
     54      PacketDuration(encoded, encoded_len) * 2 *
     55      Channels();  // 2 bytes per sample per channel
     56  size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
     57  *speech_type = ConvertSpeechType(1);
     58  return static_cast<int>(ret);
     59 }
     60 
     61 std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
     62    Buffer&& payload,
     63    uint32_t timestamp) {
     64  const int samples_per_ms = CheckedDivExact(sample_rate_hz_, 1000);
     65  return LegacyEncodedAudioFrame::SplitBySamples(
     66      this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
     67      samples_per_ms);
     68 }
     69 
     70 int AudioDecoderPcm16B::PacketDuration(const uint8_t* /* encoded */,
     71                                       size_t encoded_len) const {
     72  // Two encoded byte per sample per channel.
     73  return static_cast<int>(encoded_len / (2 * Channels()));
     74 }
     75 
     76 int AudioDecoderPcm16B::PacketDurationRedundant(const uint8_t* encoded,
     77                                                size_t encoded_len) const {
     78  return PacketDuration(encoded, encoded_len);
     79 }
     80 
     81 }  // namespace webrtc