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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 | 14 |
| 15 #include "webrtc/base/array_view.h" |
15 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
| 17 #include "webrtc/base/sanitizer.h" |
16 #include "webrtc/base/trace_event.h" | 18 #include "webrtc/base/trace_event.h" |
17 | 19 |
18 namespace webrtc { | 20 namespace webrtc { |
19 | 21 |
20 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, | 22 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, |
21 int sample_rate_hz, size_t max_decoded_bytes, | 23 int sample_rate_hz, size_t max_decoded_bytes, |
22 int16_t* decoded, SpeechType* speech_type) { | 24 int16_t* decoded, SpeechType* speech_type) { |
23 TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); | 25 TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); |
| 26 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); |
24 int duration = PacketDuration(encoded, encoded_len); | 27 int duration = PacketDuration(encoded, encoded_len); |
25 if (duration >= 0 && | 28 if (duration >= 0 && |
26 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { | 29 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
27 return -1; | 30 return -1; |
28 } | 31 } |
29 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, | 32 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
30 speech_type); | 33 speech_type); |
31 } | 34 } |
32 | 35 |
33 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, | 36 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, |
34 int sample_rate_hz, size_t max_decoded_bytes, | 37 int sample_rate_hz, size_t max_decoded_bytes, |
35 int16_t* decoded, SpeechType* speech_type) { | 38 int16_t* decoded, SpeechType* speech_type) { |
36 TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); | 39 TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); |
| 40 rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); |
37 int duration = PacketDurationRedundant(encoded, encoded_len); | 41 int duration = PacketDurationRedundant(encoded, encoded_len); |
38 if (duration >= 0 && | 42 if (duration >= 0 && |
39 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { | 43 duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
40 return -1; | 44 return -1; |
41 } | 45 } |
42 return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, | 46 return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, |
43 speech_type); | 47 speech_type); |
44 } | 48 } |
45 | 49 |
46 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, | 50 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, |
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89 return kSpeech; | 93 return kSpeech; |
90 case 2: | 94 case 2: |
91 return kComfortNoise; | 95 return kComfortNoise; |
92 default: | 96 default: |
93 assert(false); | 97 assert(false); |
94 return kSpeech; | 98 return kSpeech; |
95 } | 99 } |
96 } | 100 } |
97 | 101 |
98 } // namespace webrtc | 102 } // namespace webrtc |
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