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 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ 13 14 #include <vector> 15 16 #include "webrtc/modules/audio_processing/common.h" 17 #include "webrtc/modules/audio_processing/include/audio_processing.h" 18 #include "webrtc/modules/interface/module_common_types.h" 19 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 20 #include "webrtc/system_wrappers/interface/scoped_vector.h" 21 #include "webrtc/typedefs.h" 22 23 namespace webrtc { 24 25 class PushSincResampler; 26 class SplitChannelBuffer; 27 class IFChannelBuffer; 28 29 struct SplitFilterStates { SplitFilterStatesSplitFilterStates30 SplitFilterStates() { 31 memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); 32 memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); 33 memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); 34 memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); 35 } 36 37 static const int kStateSize = 6; 38 int analysis_filter_state1[kStateSize]; 39 int analysis_filter_state2[kStateSize]; 40 int synthesis_filter_state1[kStateSize]; 41 int synthesis_filter_state2[kStateSize]; 42 }; 43 44 class AudioBuffer { 45 public: 46 // TODO(ajm): Switch to take ChannelLayouts. 47 AudioBuffer(int input_samples_per_channel, 48 int num_input_channels, 49 int process_samples_per_channel, 50 int num_process_channels, 51 int output_samples_per_channel); 52 virtual ~AudioBuffer(); 53 54 int num_channels() const; 55 int samples_per_channel() const; 56 int samples_per_split_channel() const; 57 int samples_per_keyboard_channel() const; 58 59 int16_t* data(int channel); 60 const int16_t* data(int channel) const; 61 int16_t* low_pass_split_data(int channel); 62 const int16_t* low_pass_split_data(int channel) const; 63 int16_t* high_pass_split_data(int channel); 64 const int16_t* high_pass_split_data(int channel) const; 65 const int16_t* mixed_data(int channel) const; 66 const int16_t* mixed_low_pass_data(int channel) const; 67 const int16_t* low_pass_reference(int channel) const; 68 69 // Float versions of the accessors, with automatic conversion back and forth 70 // as necessary. The range of the numbers are the same as for int16_t. 71 float* data_f(int channel); 72 float* low_pass_split_data_f(int channel); 73 float* high_pass_split_data_f(int channel); 74 75 const float* keyboard_data() const; 76 77 SplitFilterStates* filter_states(int channel); 78 79 void set_activity(AudioFrame::VADActivity activity); 80 AudioFrame::VADActivity activity() const; 81 82 // Use for int16 interleaved data. 83 void DeinterleaveFrom(AudioFrame* audioFrame); 84 void InterleaveTo(AudioFrame* audioFrame) const; 85 // If |data_changed| is false, only the non-audio data members will be copied 86 // to |frame|. 87 void InterleaveTo(AudioFrame* frame, bool data_changed) const; 88 89 // Use for float deinterleaved data. 90 void CopyFrom(const float* const* data, 91 int samples_per_channel, 92 AudioProcessing::ChannelLayout layout); 93 void CopyTo(int samples_per_channel, 94 AudioProcessing::ChannelLayout layout, 95 float* const* data); 96 97 void CopyAndMix(int num_mixed_channels); 98 void CopyAndMixLowPass(int num_mixed_channels); 99 void CopyLowPassToReference(); 100 101 private: 102 // Called from DeinterleaveFrom() and CopyFrom(). 103 void InitForNewData(); 104 105 const int input_samples_per_channel_; 106 const int num_input_channels_; 107 const int proc_samples_per_channel_; 108 const int num_proc_channels_; 109 const int output_samples_per_channel_; 110 int samples_per_split_channel_; 111 int num_mixed_channels_; 112 int num_mixed_low_pass_channels_; 113 bool reference_copied_; 114 AudioFrame::VADActivity activity_; 115 116 const float* keyboard_data_; 117 scoped_ptr<IFChannelBuffer> channels_; 118 scoped_ptr<SplitChannelBuffer> split_channels_; 119 scoped_ptr<SplitFilterStates[]> filter_states_; 120 scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; 121 scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; 122 scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; 123 scoped_ptr<ChannelBuffer<float> > input_buffer_; 124 scoped_ptr<ChannelBuffer<float> > process_buffer_; 125 ScopedVector<PushSincResampler> input_resamplers_; 126 ScopedVector<PushSincResampler> output_resamplers_; 127 }; 128 129 } // namespace webrtc 130 131 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ 132