| /* | 
 |  *  Copyright (c) 2014 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 COMMON_AUDIO_BLOCKER_H_ | 
 | #define COMMON_AUDIO_BLOCKER_H_ | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "common_audio/audio_ring_buffer.h" | 
 | #include "common_audio/channel_buffer.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | // The callback function to process audio in the time domain. Input has already | 
 | // been windowed, and output will be windowed. The number of input channels | 
 | // must be >= the number of output channels. | 
 | class BlockerCallback { | 
 |  public: | 
 |   virtual ~BlockerCallback() {} | 
 |  | 
 |   virtual void ProcessBlock(const float* const* input, | 
 |                             size_t num_frames, | 
 |                             size_t num_input_channels, | 
 |                             size_t num_output_channels, | 
 |                             float* const* output) = 0; | 
 | }; | 
 |  | 
 | // The main purpose of Blocker is to abstract away the fact that often we | 
 | // receive a different number of audio frames than our transform takes. For | 
 | // example, most FFTs work best when the fft-size is a power of 2, but suppose | 
 | // we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames | 
 | // of audio, which is not a power of 2. Blocker allows us to specify the | 
 | // transform and all other necessary processing via the Process() callback | 
 | // function without any constraints on the transform-size | 
 | // (read: |block_size_|) or received-audio-size (read: |chunk_size_|). | 
 | // We handle this for the multichannel audio case, allowing for different | 
 | // numbers of input and output channels (for example, beamforming takes 2 or | 
 | // more input channels and returns 1 output channel). Audio signals are | 
 | // represented as deinterleaved floats in the range [-1, 1]. | 
 | // | 
 | // Blocker is responsible for: | 
 | // - blocking audio while handling potential discontinuities on the edges | 
 | //   of chunks | 
 | // - windowing blocks before sending them to Process() | 
 | // - windowing processed blocks, and overlap-adding them together before | 
 | //   sending back a processed chunk | 
 | // | 
 | // To use blocker: | 
 | // 1. Impelment a BlockerCallback object |bc|. | 
 | // 2. Instantiate a Blocker object |b|, passing in |bc|. | 
 | // 3. As you receive audio, call b.ProcessChunk() to get processed audio. | 
 | // | 
 | // A small amount of delay is added to the first received chunk to deal with | 
 | // the difference in chunk/block sizes. This delay is <= chunk_size. | 
 | // | 
 | // Ownership of window is retained by the caller.  That is, Blocker makes a | 
 | // copy of window and does not attempt to delete it. | 
 | class Blocker { | 
 |  public: | 
 |   Blocker(size_t chunk_size, | 
 |           size_t block_size, | 
 |           size_t num_input_channels, | 
 |           size_t num_output_channels, | 
 |           const float* window, | 
 |           size_t shift_amount, | 
 |           BlockerCallback* callback); | 
 |   ~Blocker(); | 
 |  | 
 |   void ProcessChunk(const float* const* input, | 
 |                     size_t chunk_size, | 
 |                     size_t num_input_channels, | 
 |                     size_t num_output_channels, | 
 |                     float* const* output); | 
 |  | 
 |   size_t initial_delay() const { return initial_delay_; } | 
 |  | 
 |  private: | 
 |   const size_t chunk_size_; | 
 |   const size_t block_size_; | 
 |   const size_t num_input_channels_; | 
 |   const size_t num_output_channels_; | 
 |  | 
 |   // The number of frames of delay to add at the beginning of the first chunk. | 
 |   const size_t initial_delay_; | 
 |  | 
 |   // The frame index into the input buffer where the first block should be read | 
 |   // from. This is necessary because shift_amount_ is not necessarily a | 
 |   // multiple of chunk_size_, so blocks won't line up at the start of the | 
 |   // buffer. | 
 |   size_t frame_offset_; | 
 |  | 
 |   // Since blocks nearly always overlap, there are certain blocks that require | 
 |   // frames from the end of one chunk and the beginning of the next chunk. The | 
 |   // input and output buffers are responsible for saving those frames between | 
 |   // calls to ProcessChunk(). | 
 |   // | 
 |   // Both contain |initial delay| + |chunk_size| frames. The input is a fairly | 
 |   // standard FIFO, but due to the overlap-add it's harder to use an | 
 |   // AudioRingBuffer for the output. | 
 |   AudioRingBuffer input_buffer_; | 
 |   ChannelBuffer<float> output_buffer_; | 
 |  | 
 |   // Space for the input block (can't wrap because of windowing). | 
 |   ChannelBuffer<float> input_block_; | 
 |  | 
 |   // Space for the output block (can't wrap because of overlap/add). | 
 |   ChannelBuffer<float> output_block_; | 
 |  | 
 |   std::unique_ptr<float[]> window_; | 
 |  | 
 |   // The amount of frames between the start of contiguous blocks. For example, | 
 |   // |shift_amount_| = |block_size_| / 2 for a Hann window. | 
 |   size_t shift_amount_; | 
 |  | 
 |   BlockerCallback* callback_; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // COMMON_AUDIO_BLOCKER_H_ |