| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |
| |
| #include <vector> |
| |
| #include "webrtc/modules/audio_processing/common.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/modules/interface/module_common_types.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/system_wrappers/interface/scoped_vector.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class PushSincResampler; |
| class SplitChannelBuffer; |
| |
| struct SplitFilterStates { |
| SplitFilterStates() { |
| memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1)); |
| memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2)); |
| memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1)); |
| memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2)); |
| } |
| |
| static const int kStateSize = 6; |
| int analysis_filter_state1[kStateSize]; |
| int analysis_filter_state2[kStateSize]; |
| int synthesis_filter_state1[kStateSize]; |
| int synthesis_filter_state2[kStateSize]; |
| }; |
| |
| class AudioBuffer { |
| public: |
| // TODO(ajm): Switch to take ChannelLayouts. |
| AudioBuffer(int input_samples_per_channel, |
| int num_input_channels, |
| int process_samples_per_channel, |
| int num_process_channels, |
| int output_samples_per_channel); |
| virtual ~AudioBuffer(); |
| |
| int num_channels() const; |
| int samples_per_channel() const; |
| int samples_per_split_channel() const; |
| int samples_per_keyboard_channel() const; |
| |
| int16_t* data(int channel); |
| const int16_t* data(int channel) const; |
| int16_t* low_pass_split_data(int channel); |
| const int16_t* low_pass_split_data(int channel) const; |
| int16_t* high_pass_split_data(int channel); |
| const int16_t* high_pass_split_data(int channel) const; |
| const int16_t* mixed_data(int channel) const; |
| const int16_t* mixed_low_pass_data(int channel) const; |
| const int16_t* low_pass_reference(int channel) const; |
| const float* keyboard_data() const; |
| |
| SplitFilterStates* filter_states(int channel); |
| |
| void set_activity(AudioFrame::VADActivity activity); |
| AudioFrame::VADActivity activity() const; |
| |
| bool is_muted() const; |
| |
| // Use for int16 interleaved data. |
| void DeinterleaveFrom(AudioFrame* audioFrame); |
| void InterleaveTo(AudioFrame* audioFrame) const; |
| // If |data_changed| is false, only the non-audio data members will be copied |
| // to |frame|. |
| void InterleaveTo(AudioFrame* frame, bool data_changed) const; |
| |
| // Use for float deinterleaved data. |
| void CopyFrom(const float* const* data, |
| int samples_per_channel, |
| AudioProcessing::ChannelLayout layout); |
| void CopyTo(int samples_per_channel, |
| AudioProcessing::ChannelLayout layout, |
| float* const* data); |
| |
| void CopyAndMix(int num_mixed_channels); |
| void CopyAndMixLowPass(int num_mixed_channels); |
| void CopyLowPassToReference(); |
| |
| private: |
| // Called from DeinterleaveFrom() and CopyFrom(). |
| void InitForNewData(); |
| |
| const int input_samples_per_channel_; |
| const int num_input_channels_; |
| const int proc_samples_per_channel_; |
| const int num_proc_channels_; |
| const int output_samples_per_channel_; |
| int samples_per_split_channel_; |
| int num_mixed_channels_; |
| int num_mixed_low_pass_channels_; |
| bool reference_copied_; |
| AudioFrame::VADActivity activity_; |
| bool is_muted_; |
| |
| // If non-null, use this instead of channels_->channel(0). This is an |
| // optimization for the case num_proc_channels_ == 1 that allows us to point |
| // to the data instead of copying it. |
| int16_t* data_; |
| |
| const float* keyboard_data_; |
| scoped_ptr<ChannelBuffer<int16_t> > channels_; |
| scoped_ptr<SplitChannelBuffer> split_channels_; |
| scoped_ptr<SplitFilterStates[]> filter_states_; |
| scoped_ptr<ChannelBuffer<int16_t> > mixed_channels_; |
| scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_; |
| scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; |
| scoped_ptr<ChannelBuffer<float> > input_buffer_; |
| scoped_ptr<ChannelBuffer<float> > process_buffer_; |
| ScopedVector<PushSincResampler> input_resamplers_; |
| ScopedVector<PushSincResampler> output_resamplers_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ |