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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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/ilbc/audio_decoder_ilbc.h" | 11 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" |
12 | 12 |
13 #include <utility> | |
14 | |
13 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
14 #include "webrtc/base/logging.h" | 16 #include "webrtc/base/logging.h" |
15 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" | 17 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" |
16 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" | 18 #include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
17 | 19 |
18 namespace webrtc { | 20 namespace webrtc { |
19 | 21 |
20 AudioDecoderIlbc::AudioDecoderIlbc() { | 22 AudioDecoderIlbc::AudioDecoderIlbc() { |
21 WebRtcIlbcfix_DecoderCreate(&dec_state_); | 23 WebRtcIlbcfix_DecoderCreate(&dec_state_); |
22 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); | 24 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); |
(...skipping 23 matching lines...) Expand all Loading... | |
46 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) { | 48 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) { |
47 return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); | 49 return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); |
48 } | 50 } |
49 | 51 |
50 void AudioDecoderIlbc::Reset() { | 52 void AudioDecoderIlbc::Reset() { |
51 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); | 53 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); |
52 } | 54 } |
53 | 55 |
54 std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload( | 56 std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload( |
55 rtc::Buffer&& payload, | 57 rtc::Buffer&& payload, |
56 uint32_t timestamp, | 58 uint32_t timestamp) { |
57 bool is_primary) { | |
58 std::vector<ParseResult> results; | 59 std::vector<ParseResult> results; |
59 size_t bytes_per_frame; | 60 size_t bytes_per_frame; |
60 int timestamps_per_frame; | 61 int timestamps_per_frame; |
61 if (payload.size() >= 950) { | 62 if (payload.size() >= 950) { |
62 LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Payload too large"; | 63 LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Payload too large"; |
63 return results; | 64 return results; |
64 } | 65 } |
65 if (payload.size() % 38 == 0) { | 66 if (payload.size() % 38 == 0) { |
66 // 20 ms frames. | 67 // 20 ms frames. |
67 bytes_per_frame = 38; | 68 bytes_per_frame = 38; |
68 timestamps_per_frame = 160; | 69 timestamps_per_frame = 160; |
69 } else if (payload.size() % 50 == 0) { | 70 } else if (payload.size() % 50 == 0) { |
70 // 30 ms frames. | 71 // 30 ms frames. |
71 bytes_per_frame = 50; | 72 bytes_per_frame = 50; |
72 timestamps_per_frame = 240; | 73 timestamps_per_frame = 240; |
73 } else { | 74 } else { |
74 LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload"; | 75 LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload"; |
75 return results; | 76 return results; |
76 } | 77 } |
77 | 78 |
78 RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame); | 79 RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame); |
79 if (payload.size() == bytes_per_frame) { | 80 if (payload.size() == bytes_per_frame) { |
80 std::unique_ptr<EncodedAudioFrame> frame( | 81 std::unique_ptr<EncodedAudioFrame> frame( |
81 new LegacyEncodedAudioFrame(this, std::move(payload), is_primary)); | 82 new LegacyEncodedAudioFrame(this, std::move(payload))); |
82 results.emplace_back(timestamp, is_primary, std::move(frame)); | 83 results.emplace_back(timestamp, 0, std::move(frame)); |
83 } else { | 84 } else { |
84 for (size_t byte_offset = 0, timestamp_offset = 0; | 85 for (size_t byte_offset = 0, timestamp_offset = 0; |
85 byte_offset < payload.size(); | 86 byte_offset < payload.size(); |
86 byte_offset += bytes_per_frame, | 87 byte_offset += bytes_per_frame, |
87 timestamp_offset += timestamps_per_frame) { | 88 timestamp_offset += timestamps_per_frame) { |
88 rtc::Buffer new_payload(payload.data() + byte_offset, bytes_per_frame); | 89 rtc::Buffer new_payload(payload.data() + byte_offset, bytes_per_frame); |
89 std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame( | 90 std::unique_ptr<EncodedAudioFrame> frame( |
90 this, std::move(new_payload), is_primary)); | 91 new LegacyEncodedAudioFrame(this, std::move(new_payload))); |
kwiberg-webrtc
2016/09/20 09:14:45
Arguably, new_payload doesn't have to be a named v
ossu
2016/09/20 13:51:56
Acknowledged.
| |
91 results.emplace_back(timestamp + timestamp_offset, is_primary, | 92 results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame)); |
92 std::move(frame)); | |
93 } | 93 } |
94 } | 94 } |
95 | 95 |
96 return results; | 96 return results; |
97 } | 97 } |
98 | 98 |
99 int AudioDecoderIlbc::SampleRateHz() const { | 99 int AudioDecoderIlbc::SampleRateHz() const { |
100 return 8000; | 100 return 8000; |
101 } | 101 } |
102 | 102 |
103 size_t AudioDecoderIlbc::Channels() const { | 103 size_t AudioDecoderIlbc::Channels() const { |
104 return 1; | 104 return 1; |
105 } | 105 } |
106 | 106 |
107 } // namespace webrtc | 107 } // namespace webrtc |
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