andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2011 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 | |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 11 | #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |
| 12 | #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 13 | |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 14 | #include <vector> |
| 15 | |
| 16 | #include "webrtc/modules/audio_processing/common.h" |
| 17 | #include "webrtc/modules/audio_processing/include/audio_processing.h" |
pbos@webrtc.org | 382c8b3 | 2013-05-28 08:11:59 +0000 | [diff] [blame] | 18 | #include "webrtc/modules/interface/module_common_types.h" |
| 19 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 20 | #include "webrtc/system_wrappers/interface/scoped_vector.h" |
pbos@webrtc.org | 382c8b3 | 2013-05-28 08:11:59 +0000 | [diff] [blame] | 21 | #include "webrtc/typedefs.h" |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 22 | |
| 23 | namespace webrtc { |
| 24 | |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 25 | class PushSincResampler; |
| 26 | class SplitChannelBuffer; |
| 27 | |
| 28 | struct SplitFilterStates { |
| 29 | SplitFilterStates() { |
| 30 | memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); |
| 31 | memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); |
| 32 | memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); |
| 33 | memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); |
| 34 | } |
| 35 | |
| 36 | static const int kStateSize = 6; |
| 37 | int analysis_filter_state1[kStateSize]; |
| 38 | int analysis_filter_state2[kStateSize]; |
| 39 | int synthesis_filter_state1[kStateSize]; |
| 40 | int synthesis_filter_state2[kStateSize]; |
| 41 | }; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 42 | |
| 43 | class AudioBuffer { |
| 44 | public: |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 45 | // TODO(ajm): Switch to take ChannelLayouts. |
| 46 | AudioBuffer(int input_samples_per_channel, |
| 47 | int num_input_channels, |
| 48 | int process_samples_per_channel, |
| 49 | int num_process_channels, |
| 50 | int output_samples_per_channel); |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 51 | virtual ~AudioBuffer(); |
| 52 | |
| 53 | int num_channels() const; |
| 54 | int samples_per_channel() const; |
| 55 | int samples_per_split_channel() const; |
andrew@webrtc.org | fbf2568 | 2014-04-24 18:28:56 +0000 | [diff] [blame] | 56 | int samples_per_keyboard_channel() const; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 57 | |
andrew@webrtc.org | c2e6438 | 2014-04-30 16:44:13 +0000 | [diff] [blame] | 58 | int16_t* data(int channel); |
| 59 | const int16_t* data(int channel) const; |
| 60 | int16_t* low_pass_split_data(int channel); |
| 61 | const int16_t* low_pass_split_data(int channel) const; |
| 62 | int16_t* high_pass_split_data(int channel); |
| 63 | const int16_t* high_pass_split_data(int channel) const; |
| 64 | const int16_t* mixed_data(int channel) const; |
| 65 | const int16_t* mixed_low_pass_data(int channel) const; |
| 66 | const int16_t* low_pass_reference(int channel) const; |
andrew@webrtc.org | fbf2568 | 2014-04-24 18:28:56 +0000 | [diff] [blame] | 67 | const float* keyboard_data() const; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 68 | |
andrew@webrtc.org | c2e6438 | 2014-04-30 16:44:13 +0000 | [diff] [blame] | 69 | SplitFilterStates* filter_states(int channel); |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 70 | |
| 71 | void set_activity(AudioFrame::VADActivity activity); |
| 72 | AudioFrame::VADActivity activity() const; |
| 73 | |
andrew@webrtc.org | 3c5112c | 2014-03-04 20:58:13 +0000 | [diff] [blame] | 74 | // Use for int16 interleaved data. |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 75 | void DeinterleaveFrom(AudioFrame* audioFrame); |
| 76 | void InterleaveTo(AudioFrame* audioFrame) const; |
| 77 | // If |data_changed| is false, only the non-audio data members will be copied |
| 78 | // to |frame|. |
| 79 | void InterleaveTo(AudioFrame* frame, bool data_changed) const; |
andrew@webrtc.org | 3c5112c | 2014-03-04 20:58:13 +0000 | [diff] [blame] | 80 | |
| 81 | // Use for float deinterleaved data. |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 82 | void CopyFrom(const float* const* data, |
| 83 | int samples_per_channel, |
| 84 | AudioProcessing::ChannelLayout layout); |
| 85 | void CopyTo(int samples_per_channel, |
| 86 | AudioProcessing::ChannelLayout layout, |
| 87 | float* const* data); |
andrew@webrtc.org | 3c5112c | 2014-03-04 20:58:13 +0000 | [diff] [blame] | 88 | |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 89 | void CopyAndMix(int num_mixed_channels); |
| 90 | void CopyAndMixLowPass(int num_mixed_channels); |
| 91 | void CopyLowPassToReference(); |
| 92 | |
| 93 | private: |
andrew@webrtc.org | 3c5112c | 2014-03-04 20:58:13 +0000 | [diff] [blame] | 94 | // Called from DeinterleaveFrom() and CopyFrom(). |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 95 | void InitForNewData(); |
andrew@webrtc.org | 3c5112c | 2014-03-04 20:58:13 +0000 | [diff] [blame] | 96 | |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 97 | const int input_samples_per_channel_; |
| 98 | const int num_input_channels_; |
| 99 | const int proc_samples_per_channel_; |
| 100 | const int num_proc_channels_; |
| 101 | const int output_samples_per_channel_; |
| 102 | int samples_per_split_channel_; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 103 | int num_mixed_channels_; |
| 104 | int num_mixed_low_pass_channels_; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 105 | bool reference_copied_; |
| 106 | AudioFrame::VADActivity activity_; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 107 | |
kwiberg@webrtc.org | 8b4f539 | 2014-05-08 07:10:11 +0000 | [diff] [blame] | 108 | // If non-null, use this instead of channels_->channel(0). This is an |
| 109 | // optimization for the case num_proc_channels_ == 1 that allows us to point |
| 110 | // to the data instead of copying it. |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 111 | int16_t* data_; |
kwiberg@webrtc.org | 8b4f539 | 2014-05-08 07:10:11 +0000 | [diff] [blame] | 112 | |
andrew@webrtc.org | fbf2568 | 2014-04-24 18:28:56 +0000 | [diff] [blame] | 113 | const float* keyboard_data_; |
mflodman@webrtc.org | 409cf2a | 2014-05-14 09:39:56 +0000 | [diff] [blame] | 114 | scoped_ptr<ChannelBuffer<int16_t> > channels_; |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 115 | scoped_ptr<SplitChannelBuffer> split_channels_; |
| 116 | scoped_ptr<SplitFilterStates[]> filter_states_; |
| 117 | scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; |
| 118 | scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; |
| 119 | scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; |
| 120 | scoped_ptr<ChannelBuffer<float> > input_buffer_; |
| 121 | scoped_ptr<ChannelBuffer<float> > process_buffer_; |
| 122 | ScopedVector<PushSincResampler> input_resamplers_; |
| 123 | ScopedVector<PushSincResampler> output_resamplers_; |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 124 | }; |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 125 | |
andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 126 | } // namespace webrtc |
| 127 | |
andrew@webrtc.org | 2e24460 | 2014-04-22 21:00:04 +0000 | [diff] [blame] | 128 | #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |