Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2016 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/legacy_encoded_audio_frame.h"
      12             : 
      13             : #include <algorithm>
      14             : #include <memory>
      15             : #include <utility>
      16             : 
      17             : namespace webrtc {
      18             : 
      19           0 : LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
      20           0 :                                                  rtc::Buffer&& payload)
      21           0 :     : decoder_(decoder), payload_(std::move(payload)) {}
      22             : 
      23             : LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
      24             : 
      25           0 : size_t LegacyEncodedAudioFrame::Duration() const {
      26           0 :   const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
      27           0 :   return (ret < 0) ? 0 : static_cast<size_t>(ret);
      28             : }
      29             : 
      30             : rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
      31           0 : LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
      32           0 :   AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
      33           0 :   const int ret = decoder_->Decode(
      34           0 :       payload_.data(), payload_.size(), decoder_->SampleRateHz(),
      35           0 :       decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
      36             : 
      37           0 :   if (ret < 0)
      38           0 :     return rtc::Optional<DecodeResult>();
      39             : 
      40           0 :   return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
      41             : }
      42             : 
      43           0 : std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
      44             :     AudioDecoder* decoder,
      45             :     rtc::Buffer&& payload,
      46             :     uint32_t timestamp,
      47             :     size_t bytes_per_ms,
      48             :     uint32_t timestamps_per_ms) {
      49           0 :   RTC_DCHECK(payload.data());
      50           0 :   std::vector<AudioDecoder::ParseResult> results;
      51           0 :   size_t split_size_bytes = payload.size();
      52             : 
      53             :   // Find a "chunk size" >= 20 ms and < 40 ms.
      54           0 :   const size_t min_chunk_size = bytes_per_ms * 20;
      55           0 :   if (min_chunk_size >= payload.size()) {
      56             :     std::unique_ptr<LegacyEncodedAudioFrame> frame(
      57           0 :         new LegacyEncodedAudioFrame(decoder, std::move(payload)));
      58           0 :     results.emplace_back(timestamp, 0, std::move(frame));
      59             :   } else {
      60             :     // Reduce the split size by half as long as |split_size_bytes| is at least
      61             :     // twice the minimum chunk size (so that the resulting size is at least as
      62             :     // large as the minimum chunk size).
      63           0 :     while (split_size_bytes >= 2 * min_chunk_size) {
      64           0 :       split_size_bytes /= 2;
      65             :     }
      66             : 
      67             :     const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
      68           0 :         split_size_bytes * timestamps_per_ms / bytes_per_ms);
      69             :     size_t byte_offset;
      70             :     uint32_t timestamp_offset;
      71           0 :     for (byte_offset = 0, timestamp_offset = 0;
      72           0 :          byte_offset < payload.size();
      73           0 :          byte_offset += split_size_bytes,
      74           0 :              timestamp_offset += timestamps_per_chunk) {
      75           0 :       split_size_bytes =
      76           0 :           std::min(split_size_bytes, payload.size() - byte_offset);
      77           0 :       rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
      78             :       std::unique_ptr<LegacyEncodedAudioFrame> frame(
      79           0 :           new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
      80           0 :       results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
      81             :     }
      82             :   }
      83             : 
      84           0 :   return results;
      85             : }
      86             : 
      87             : }  // namespace webrtc