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
15#include "webrtc/base/checks.h"
16#include "webrtc/base/trace_event.h"
17
18namespace webrtc {
19
20int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
21                         int sample_rate_hz, size_t max_decoded_bytes,
22                         int16_t* decoded, SpeechType* speech_type) {
23  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
24  int duration = PacketDuration(encoded, encoded_len);
25  if (duration >= 0 &&
26      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
27    return -1;
28  }
29  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
30                        speech_type);
31}
32
33int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
34                                  int sample_rate_hz, size_t max_decoded_bytes,
35                                  int16_t* decoded, SpeechType* speech_type) {
36  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
37  int duration = PacketDurationRedundant(encoded, encoded_len);
38  if (duration >= 0 &&
39      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
40    return -1;
41  }
42  return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
43                                 speech_type);
44}
45
46int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
47                                          size_t encoded_len,
48                                          int sample_rate_hz, int16_t* decoded,
49                                          SpeechType* speech_type) {
50  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
51                        speech_type);
52}
53
54bool AudioDecoder::HasDecodePlc() const { return false; }
55
56size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
57  return 0;
58}
59
60int AudioDecoder::IncomingPacket(const uint8_t* payload,
61                                 size_t payload_len,
62                                 uint16_t rtp_sequence_number,
63                                 uint32_t rtp_timestamp,
64                                 uint32_t arrival_timestamp) {
65  return 0;
66}
67
68int AudioDecoder::ErrorCode() { return 0; }
69
70int AudioDecoder::PacketDuration(const uint8_t* encoded,
71                                 size_t encoded_len) const {
72  return kNotImplemented;
73}
74
75int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
76                                          size_t encoded_len) const {
77  return kNotImplemented;
78}
79
80bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
81                                size_t encoded_len) const {
82  return false;
83}
84
85CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
86  FATAL() << "Not a CNG decoder";
87  return NULL;
88}
89
90AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
91  switch (type) {
92    case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
93    case 1:
94      return kSpeech;
95    case 2:
96      return kComfortNoise;
97    default:
98      assert(false);
99      return kSpeech;
100  }
101}
102
103}  // namespace webrtc
104