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| 1 /* | |
| 2 * Copyright (c) 2012 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/audio_decoder.h" | |
| 12 | |
| 13 #include <assert.h> | |
| 14 #include <memory> | |
| 15 #include <utility> | |
| 16 | |
| 17 #include "webrtc/base/array_view.h" | |
| 18 #include "webrtc/base/checks.h" | |
| 19 #include "webrtc/base/sanitizer.h" | |
| 20 #include "webrtc/base/trace_event.h" | |
| 21 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" | |
| 22 | |
| 23 namespace webrtc { | |
| 24 | |
| 25 AudioDecoder::ParseResult::ParseResult() = default; | |
| 26 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; | |
| 27 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, | |
| 28 int priority, | |
| 29 std::unique_ptr<EncodedAudioFrame> frame) | |
| 30 : timestamp(timestamp), priority(priority), frame(std::move(frame)) { | |
| 31 RTC_DCHECK_GE(priority, 0); | |
| 32 } | |
| 33 | |
| 34 AudioDecoder::ParseResult::~ParseResult() = default; | |
| 35 | |
| 36 AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=( | |
| 37 ParseResult&& b) = default; | |
| 38 | |
| 39 std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload( | |
| 40 rtc::Buffer&& payload, | |
| 41 uint32_t timestamp) { | |
| 42 std::vector<ParseResult> results; | |
| 43 std::unique_ptr<EncodedAudioFrame> frame( | |
| 44 new LegacyEncodedAudioFrame(this, std::move(payload))); | |
| 45 results.emplace_back(timestamp, 0, std::move(frame)); | |
| 46 return results; | |
| 47 } | |
| 48 | |
| 49 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, | |
| 50 int sample_rate_hz, size_t max_decoded_bytes, | |
| 51 int16_t* decoded, SpeechType* speech_type) { | |
| 52 TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); | |
| 53 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); | |
| 54 int duration = PacketDuration(encoded, encoded_len); | |
| 55 if (duration >= 0 && | |
| 56 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { | |
| 57 return -1; | |
| 58 } | |
| 59 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, | |
| 60 speech_type); | |
| 61 } | |
| 62 | |
| 63 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, | |
| 64 int sample_rate_hz, size_t max_decoded_bytes, | |
| 65 int16_t* decoded, SpeechType* speech_type) { | |
| 66 TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); | |
| 67 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); | |
| 68 int duration = PacketDurationRedundant(encoded, encoded_len); | |
| 69 if (duration >= 0 && | |
| 70 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { | |
| 71 return -1; | |
| 72 } | |
| 73 return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, | |
| 74 speech_type); | |
| 75 } | |
| 76 | |
| 77 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, | |
| 78 size_t encoded_len, | |
| 79 int sample_rate_hz, int16_t* decoded, | |
| 80 SpeechType* speech_type) { | |
| 81 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, | |
| 82 speech_type); | |
| 83 } | |
| 84 | |
| 85 bool AudioDecoder::HasDecodePlc() const { return false; } | |
| 86 | |
| 87 size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) { | |
| 88 return 0; | |
| 89 } | |
| 90 | |
| 91 int AudioDecoder::IncomingPacket(const uint8_t* payload, | |
| 92 size_t payload_len, | |
| 93 uint16_t rtp_sequence_number, | |
| 94 uint32_t rtp_timestamp, | |
| 95 uint32_t arrival_timestamp) { | |
| 96 return 0; | |
| 97 } | |
| 98 | |
| 99 int AudioDecoder::ErrorCode() { return 0; } | |
| 100 | |
| 101 int AudioDecoder::PacketDuration(const uint8_t* encoded, | |
| 102 size_t encoded_len) const { | |
| 103 return kNotImplemented; | |
| 104 } | |
| 105 | |
| 106 int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded, | |
| 107 size_t encoded_len) const { | |
| 108 return kNotImplemented; | |
| 109 } | |
| 110 | |
| 111 bool AudioDecoder::PacketHasFec(const uint8_t* encoded, | |
| 112 size_t encoded_len) const { | |
| 113 return false; | |
| 114 } | |
| 115 | |
| 116 AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) { | |
| 117 switch (type) { | |
| 118 case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech. | |
| 119 case 1: | |
| 120 return kSpeech; | |
| 121 case 2: | |
| 122 return kComfortNoise; | |
| 123 default: | |
| 124 assert(false); | |
| 125 return kSpeech; | |
| 126 } | |
| 127 } | |
| 128 | |
| 129 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2668523004:
Move AudioDecoder and related stuff to the api/ directory (Closed)
