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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 |
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30 // end of the expanded data (thanks to how the Expand class operates). However, | 30 // end of the expanded data (thanks to how the Expand class operates). However, |
31 // if a later packet arrives instead, the loss is a fact, and the new data must | 31 // if a later packet arrives instead, the loss is a fact, and the new data must |
32 // be stitched together with the end of the expanded data. This stitching is | 32 // be stitched together with the end of the expanded data. This stitching is |
33 // what the Merge class does. | 33 // what the Merge class does. |
34 class Merge { | 34 class Merge { |
35 public: | 35 public: |
36 Merge(int fs_hz, | 36 Merge(int fs_hz, |
37 size_t num_channels, | 37 size_t num_channels, |
38 Expand* expand, | 38 Expand* expand, |
39 SyncBuffer* sync_buffer); | 39 SyncBuffer* sync_buffer); |
40 virtual ~Merge() {} | 40 virtual ~Merge(); |
41 | 41 |
42 // The main method to produce the audio data. The decoded data is supplied in | 42 // The main method to produce the audio data. The decoded data is supplied in |
43 // |input|, having |input_length| samples in total for all channels | 43 // |input|, having |input_length| samples in total for all channels |
44 // (interleaved). The result is written to |output|. The number of channels | 44 // (interleaved). The result is written to |output|. The number of channels |
45 // allocated in |output| defines the number of channels that will be used when | 45 // allocated in |output| defines the number of channels that will be used when |
46 // de-interleaving |input|. The values in |external_mute_factor_array| (Q14) | 46 // de-interleaving |input|. The values in |external_mute_factor_array| (Q14) |
47 // will be used to scale the audio, and is updated in the process. The array | 47 // will be used to scale the audio, and is updated in the process. The array |
48 // must have |num_channels_| elements. | 48 // must have |num_channels_| elements. |
49 virtual size_t Process(int16_t* input, size_t input_length, | 49 virtual size_t Process(int16_t* input, size_t input_length, |
50 int16_t* external_mute_factor_array, | 50 int16_t* external_mute_factor_array, |
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86 size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length, | 86 size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length, |
87 size_t expand_period) const; | 87 size_t expand_period) const; |
88 | 88 |
89 const int fs_mult_; // fs_hz_ / 8000. | 89 const int fs_mult_; // fs_hz_ / 8000. |
90 const size_t timestamps_per_call_; | 90 const size_t timestamps_per_call_; |
91 Expand* expand_; | 91 Expand* expand_; |
92 SyncBuffer* sync_buffer_; | 92 SyncBuffer* sync_buffer_; |
93 int16_t expanded_downsampled_[kExpandDownsampLength]; | 93 int16_t expanded_downsampled_[kExpandDownsampLength]; |
94 int16_t input_downsampled_[kInputDownsampLength]; | 94 int16_t input_downsampled_[kInputDownsampLength]; |
95 AudioMultiVector expanded_; | 95 AudioMultiVector expanded_; |
| 96 std::vector<int16_t> temp_data_; |
96 | 97 |
97 RTC_DISALLOW_COPY_AND_ASSIGN(Merge); | 98 RTC_DISALLOW_COPY_AND_ASSIGN(Merge); |
98 }; | 99 }; |
99 | 100 |
100 } // namespace webrtc | 101 } // namespace webrtc |
101 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_ | 102 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_ |
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