ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 04:47:31 | [diff] [blame] | 11 | #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 12 | |
| 13 | #include <algorithm> |
| 14 | #include <memory> |
| 15 | #include <utility> |
| 16 | |
Yves Gerey | 988cc08 | 2018-10-23 10:03:01 | [diff] [blame] | 17 | #include "rtc_base/checks.h" |
| 18 | |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 19 | namespace webrtc { |
| 20 | |
| 21 | LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder, |
ossu | a70695a | 2016-09-22 09:06:28 | [diff] [blame] | 22 | rtc::Buffer&& payload) |
| 23 | : decoder_(decoder), payload_(std::move(payload)) {} |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 24 | |
| 25 | LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default; |
| 26 | |
| 27 | size_t LegacyEncodedAudioFrame::Duration() const { |
ossu | a70695a | 2016-09-22 09:06:28 | [diff] [blame] | 28 | const int ret = decoder_->PacketDuration(payload_.data(), payload_.size()); |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 29 | return (ret < 0) ? 0 : static_cast<size_t>(ret); |
| 30 | } |
| 31 | |
Danil Chapovalov | b602123 | 2018-06-19 11:26:36 | [diff] [blame] | 32 | absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult> |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 33 | LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const { |
| 34 | AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech; |
ossu | a70695a | 2016-09-22 09:06:28 | [diff] [blame] | 35 | const int ret = decoder_->Decode( |
| 36 | payload_.data(), payload_.size(), decoder_->SampleRateHz(), |
| 37 | decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 38 | |
| 39 | if (ret < 0) |
Danil Chapovalov | b602123 | 2018-06-19 11:26:36 | [diff] [blame] | 40 | return absl::nullopt; |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 41 | |
Oskar Sundbom | 12ab00b | 2017-11-16 14:31:38 | [diff] [blame] | 42 | return DecodeResult{static_cast<size_t>(ret), speech_type}; |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 43 | } |
| 44 | |
| 45 | std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples( |
| 46 | AudioDecoder* decoder, |
| 47 | rtc::Buffer&& payload, |
| 48 | uint32_t timestamp, |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 49 | size_t bytes_per_ms, |
| 50 | uint32_t timestamps_per_ms) { |
| 51 | RTC_DCHECK(payload.data()); |
| 52 | std::vector<AudioDecoder::ParseResult> results; |
| 53 | size_t split_size_bytes = payload.size(); |
| 54 | |
| 55 | // Find a "chunk size" >= 20 ms and < 40 ms. |
| 56 | const size_t min_chunk_size = bytes_per_ms * 20; |
| 57 | if (min_chunk_size >= payload.size()) { |
| 58 | std::unique_ptr<LegacyEncodedAudioFrame> frame( |
ossu | a70695a | 2016-09-22 09:06:28 | [diff] [blame] | 59 | new LegacyEncodedAudioFrame(decoder, std::move(payload))); |
| 60 | results.emplace_back(timestamp, 0, std::move(frame)); |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 61 | } else { |
| 62 | // Reduce the split size by half as long as |split_size_bytes| is at least |
| 63 | // twice the minimum chunk size (so that the resulting size is at least as |
| 64 | // large as the minimum chunk size). |
| 65 | while (split_size_bytes >= 2 * min_chunk_size) { |
| 66 | split_size_bytes /= 2; |
| 67 | } |
| 68 | |
| 69 | const uint32_t timestamps_per_chunk = static_cast<uint32_t>( |
| 70 | split_size_bytes * timestamps_per_ms / bytes_per_ms); |
| 71 | size_t byte_offset; |
| 72 | uint32_t timestamp_offset; |
Yves Gerey | 665174f | 2018-06-19 13:03:05 | [diff] [blame] | 73 | for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size(); |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 74 | byte_offset += split_size_bytes, |
Yves Gerey | 665174f | 2018-06-19 13:03:05 | [diff] [blame] | 75 | timestamp_offset += timestamps_per_chunk) { |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 76 | split_size_bytes = |
| 77 | std::min(split_size_bytes, payload.size() - byte_offset); |
| 78 | rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes); |
| 79 | std::unique_ptr<LegacyEncodedAudioFrame> frame( |
ossu | a70695a | 2016-09-22 09:06:28 | [diff] [blame] | 80 | new LegacyEncodedAudioFrame(decoder, std::move(new_payload))); |
| 81 | results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame)); |
ossu | 0d526d5 | 2016-09-21 08:57:31 | [diff] [blame] | 82 | } |
| 83 | } |
| 84 | |
| 85 | return results; |
| 86 | } |
| 87 | |
| 88 | } // namespace webrtc |