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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" | 11 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
12 | 12 |
13 #include <assert.h> | 13 #include <assert.h> |
14 #include <memory> | |
15 #include <utility> | |
14 | 16 |
15 #include "webrtc/base/array_view.h" | 17 #include "webrtc/base/array_view.h" |
16 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/sanitizer.h" | 19 #include "webrtc/base/sanitizer.h" |
18 #include "webrtc/base/trace_event.h" | 20 #include "webrtc/base/trace_event.h" |
21 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" | |
19 | 22 |
20 namespace webrtc { | 23 namespace webrtc { |
21 | 24 |
22 namespace { | |
23 class LegacyFrame : public AudioDecoder::EncodedAudioFrame { | |
24 public: | |
25 LegacyFrame(AudioDecoder* decoder, | |
26 rtc::Buffer* payload, | |
27 bool is_primary_payload) | |
28 : decoder_(decoder), | |
29 payload_(std::move(*payload)), | |
30 is_primary_payload_(is_primary_payload) {} | |
31 | |
32 size_t Duration() const override { | |
33 int ret; | |
34 if (is_primary_payload_) { | |
35 ret = decoder_->PacketDuration(payload_.data(), payload_.size()); | |
36 } else { | |
37 ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size()); | |
38 } | |
39 return (ret < 0) ? 0 : static_cast<size_t>(ret); | |
40 } | |
41 | |
42 rtc::Optional<DecodeResult> Decode( | |
43 rtc::ArrayView<int16_t> decoded) const override { | |
44 AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech; | |
45 int ret; | |
46 if (is_primary_payload_) { | |
47 ret = decoder_->Decode( | |
48 payload_.data(), payload_.size(), decoder_->SampleRateHz(), | |
49 decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); | |
50 } else { | |
51 ret = decoder_->DecodeRedundant( | |
52 payload_.data(), payload_.size(), decoder_->SampleRateHz(), | |
53 decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); | |
54 } | |
55 | |
56 if (ret < 0) | |
57 return rtc::Optional<DecodeResult>(); | |
58 | |
59 return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type}); | |
60 } | |
61 | |
62 private: | |
63 AudioDecoder* const decoder_; | |
64 const rtc::Buffer payload_; | |
65 const bool is_primary_payload_; | |
66 }; | |
67 } | |
68 | |
69 AudioDecoder::ParseResult::ParseResult() = default; | 25 AudioDecoder::ParseResult::ParseResult() = default; |
70 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; | 26 AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; |
71 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, | 27 AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, |
72 bool primary, | 28 bool primary, |
73 std::unique_ptr<EncodedAudioFrame> frame) | 29 std::unique_ptr<EncodedAudioFrame> frame) |
74 : timestamp(timestamp), primary(primary), frame(std::move(frame)) {} | 30 : timestamp(timestamp), primary(primary), frame(std::move(frame)) {} |
75 | 31 |
76 AudioDecoder::ParseResult::~ParseResult() = default; | 32 AudioDecoder::ParseResult::~ParseResult() = default; |
77 | 33 |
78 AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=( | 34 AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=( |
79 ParseResult&& b) = default; | 35 ParseResult&& b) = default; |
80 | 36 |
81 std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload( | 37 std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload( |
82 rtc::Buffer* payload, | 38 rtc::Buffer* payload, |
83 uint32_t timestamp, | 39 uint32_t timestamp, |
84 bool is_primary) { | 40 bool is_primary) { |
85 std::vector<ParseResult> results; | 41 std::vector<ParseResult> results; |
86 std::unique_ptr<EncodedAudioFrame> frame( | 42 std::unique_ptr<EncodedAudioFrame> frame( |
87 new LegacyFrame(this, payload, is_primary)); | 43 new LegacyEncodedAudioFrame(this, payload, is_primary)); |
88 results.emplace_back(timestamp, is_primary, std::move(frame)); | 44 results.emplace_back(timestamp, is_primary, std::move(frame)); |
89 return results; | 45 return results; |
kwiberg-webrtc
2016/09/15 13:01:24
Won't a construction along these lines work?
st
ossu
2016/09/15 14:41:05
Ideally that _should_ work but I can't get it to.
kwiberg-webrtc
2016/09/16 00:48:07
No, they're allowed now. Could it be that we're us
| |
90 } | 46 } |
91 | 47 |
92 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, | 48 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, |
93 int sample_rate_hz, size_t max_decoded_bytes, | 49 int sample_rate_hz, size_t max_decoded_bytes, |
94 int16_t* decoded, SpeechType* speech_type) { | 50 int16_t* decoded, SpeechType* speech_type) { |
95 TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); | 51 TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); |
96 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); | 52 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); |
97 int duration = PacketDuration(encoded, encoded_len); | 53 int duration = PacketDuration(encoded, encoded_len); |
98 if (duration >= 0 && | 54 if (duration >= 0 && |
99 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { | 55 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
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163 return kSpeech; | 119 return kSpeech; |
164 case 2: | 120 case 2: |
165 return kComfortNoise; | 121 return kComfortNoise; |
166 default: | 122 default: |
167 assert(false); | 123 assert(false); |
168 return kSpeech; | 124 return kSpeech; |
169 } | 125 } |
170 } | 126 } |
171 | 127 |
172 } // namespace webrtc | 128 } // namespace webrtc |
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