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 "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
12
13 #include <algorithm>
14 #include <memory>
15 #include <utility>
16
17 namespace webrtc {
18
LegacyEncodedAudioFrame(AudioDecoder * decoder,rtc::Buffer && payload)19 LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
20 rtc::Buffer&& payload)
21 : decoder_(decoder), payload_(std::move(payload)) {}
22
23 LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
24
Duration() const25 size_t LegacyEncodedAudioFrame::Duration() const {
26 const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
27 return (ret < 0) ? 0 : static_cast<size_t>(ret);
28 }
29
30 rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
Decode(rtc::ArrayView<int16_t> decoded) const31 LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
32 AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
33 const int ret = decoder_->Decode(
34 payload_.data(), payload_.size(), decoder_->SampleRateHz(),
35 decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
36
37 if (ret < 0)
38 return rtc::nullopt;
39
40 return DecodeResult{static_cast<size_t>(ret), speech_type};
41 }
42
SplitBySamples(AudioDecoder * decoder,rtc::Buffer && payload,uint32_t timestamp,size_t bytes_per_ms,uint32_t timestamps_per_ms)43 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 RTC_DCHECK(payload.data());
50 std::vector<AudioDecoder::ParseResult> results;
51 size_t split_size_bytes = payload.size();
52
53 // Find a "chunk size" >= 20 ms and < 40 ms.
54 const size_t min_chunk_size = bytes_per_ms * 20;
55 if (min_chunk_size >= payload.size()) {
56 std::unique_ptr<LegacyEncodedAudioFrame> frame(
57 new LegacyEncodedAudioFrame(decoder, std::move(payload)));
58 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 while (split_size_bytes >= 2 * min_chunk_size) {
64 split_size_bytes /= 2;
65 }
66
67 const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
68 split_size_bytes * timestamps_per_ms / bytes_per_ms);
69 size_t byte_offset;
70 uint32_t timestamp_offset;
71 for (byte_offset = 0, timestamp_offset = 0;
72 byte_offset < payload.size();
73 byte_offset += split_size_bytes,
74 timestamp_offset += timestamps_per_chunk) {
75 split_size_bytes =
76 std::min(split_size_bytes, payload.size() - byte_offset);
77 rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
78 std::unique_ptr<LegacyEncodedAudioFrame> frame(
79 new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
80 results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
81 }
82 }
83
84 return results;
85 }
86
87 } // namespace webrtc
88