| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| // Modifications Copyright 2017 Google Inc. All Rights Reserved. |
| // |
| // Licensed under the Apache License, Version 2.0 (the "License"); |
| // you may not use this file except in compliance with the License. |
| // You may obtain a copy of the License at |
| // |
| // http://www.apache.org/licenses/LICENSE-2.0 |
| // |
| // Unless required by applicable law or agreed to in writing, software |
| // distributed under the License is distributed on an "AS IS" BASIS, |
| // WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| // See the License for the specific language governing permissions and |
| // limitations under the License. |
| |
| // AudioTimeStretcher buffers and transforms audio data. The owner of this |
| // object provides audio data to the object through EnqueueBuffer() and requests |
| // data from the buffer via FillBuffer(). |
| // |
| // This class is *not* thread-safe. Calls to enqueue and retrieve data must be |
| // locked if called from multiple threads. |
| // |
| // AudioTimeStretcher uses the Waveform Similarity Overlap and Add (WSOLA) |
| // algorithm to stretch or compress audio data to meet playback speeds less than |
| // or greater than the natural playback of the audio stream. The algorithm |
| // preserves local properties of the audio, therefore, pitch and harmonics are |
| // preserved. See audio_renderer_algorith.cc for a more elaborate description of |
| // the algorithm. |
| // |
| // Audio at very low or very high playback rates are muted to preserve quality. |
| |
| #ifndef STARBOARD_SHARED_STARBOARD_PLAYER_FILTER_AUDIO_TIME_STRETCHER_H_ |
| #define STARBOARD_SHARED_STARBOARD_PLAYER_FILTER_AUDIO_TIME_STRETCHER_H_ |
| |
| #include "starboard/common/ref_counted.h" |
| #include "starboard/common/scoped_ptr.h" |
| #include "starboard/shared/starboard/player/decoded_audio_internal.h" |
| #include "starboard/shared/starboard/player/filter/decoded_audio_queue.h" |
| #include "starboard/types.h" |
| |
| namespace starboard { |
| namespace shared { |
| namespace starboard { |
| namespace player { |
| namespace filter { |
| |
| class AudioTimeStretcher { |
| public: |
| AudioTimeStretcher(); |
| ~AudioTimeStretcher(); |
| |
| // Initializes this object with information about the audio stream. |
| void Initialize(int channels, int samples_per_second); |
| |
| // Tries to fill |requested_frames| frames into |dest| with possibly scaled |
| // data from our |audio_buffer_|. Data is scaled based on |playback_rate|, |
| // using a variation of the Overlap-Add method to combine sample windows. |
| // |
| // Data from |audio_buffer_| is consumed in proportion to the playback rate. |
| // |
| // |dest_offset| is the offset in frames for writing into |dest|. |
| // |
| // Returns the number of frames copied into |dest|. |
| scoped_refptr<DecodedAudio> Read(int requested_frames, double playback_rate); |
| |
| // Clears |audio_buffer_|. |
| void FlushBuffers(); |
| |
| // Enqueues a buffer. It is called from the owner of the algorithm after a |
| // read completes. |
| void EnqueueBuffer(const scoped_refptr<DecodedAudio>& audio_data); |
| |
| // Returns true if |audio_buffer_| is at or exceeds capacity. |
| bool IsQueueFull() const; |
| |
| // Returns the number of frames left in |audio_buffer_|, which may be larger |
| // than QueueCapacity() in the event that EnqueueBuffer() delivered more data |
| // than |audio_buffer_| was intending to hold. |
| int frames_buffered() { return audio_buffer_.frames(); } |
| |
| private: |
| // Within |search_block_|, find the block of data that is most similar to |
| // |target_block_|, and write it in |optimal_block_|. This method assumes that |
| // there is enough data to perform a search, i.e. |search_block_| and |
| // |target_block_| can be extracted from the available frames. |
| void GetOptimalBlock(); |
| |
| // Read a maximum of |requested_frames| frames from |wsola_output_|. Returns |
| // number of frames actually read. |
| int WriteCompletedFramesTo(int requested_frames, |
| int output_offset, |
| DecodedAudio* dest); |
| |
| // Fill |dest| with frames from |audio_buffer_| starting from frame |
| // |read_offset_frames|. |dest| is expected to have the same number of |
| // channels as |audio_buffer_|. A negative offset, i.e. |
| // |read_offset_frames| < 0, is accepted assuming that |audio_buffer| is zero |
| // for negative indices. This might happen for few first frames. This method |
| // assumes there is enough frames to fill |dest|, i.e. |read_offset_frames| + |
| // |dest->frames()| does not extend to future. |
| void PeekAudioWithZeroPrepend(int read_offset_frames, DecodedAudio* dest); |
| |
| // Run one iteration of WSOLA, if there are sufficient frames. This will |
| // overlap-and-add one block to |wsola_output_|, hence, |num_complete_frames_| |
| // is incremented by |ola_hop_size_|. |
| bool RunOneWsolaIteration(double playback_rate); |
| |
| // Seek |audio_buffer_| forward to remove frames from input that are not used |
| // any more. State of the WSOLA will be updated accordingly. |
| void RemoveOldInputFrames(double playback_rate); |
| |
| // Update |output_time_| by |time_change|. In turn |search_block_index_| is |
| // updated. |
| void UpdateOutputTime(double playback_rate, double time_change); |
| |
| // Is |target_block_| fully within |search_block_|? If so, we don't need to |
| // perform the search. |
| bool TargetIsWithinSearchRegion() const; |
| |
| // Do we have enough data to perform one round of WSOLA? |
| bool CanPerformWsola() const; |
| |
| // Converts a time in milliseconds to frames using |samples_per_second_|. |
| int ConvertMillisecondsToFrames(int ms) const; |
| |
| // Number of channels in audio stream. |
| int channels_; |
| |
| // Bytes per audio frame. |
| int bytes_per_frame_; |
| |
| // Sample rate of audio stream. |
| int samples_per_second_; |
| |
| // Buffered audio data. |
| DecodedAudioQueue audio_buffer_; |
| |
| // If muted, keep track of partial frames that should have been skipped over. |
| double muted_partial_frame_; |
| |
| // How many frames to have in the queue before we report the queue is full. |
| int capacity_; |
| |
| // Book keeping of the current time of generated audio, in frames. This |
| // should be appropriately updated when out samples are generated, regardless |
| // of whether we push samples out when FillBuffer() is called or we store |
| // audio in |wsola_output_| for the subsequent calls to FillBuffer(). |
| // Furthermore, if samples from |audio_buffer_| are evicted then this |
| // member variable should be updated based on |playback_rate_|. |
| // Note that this member should be updated ONLY by calling UpdateOutputTime(), |
| // so that |search_block_index_| is update accordingly. |
| double output_time_; |
| |
| // The offset of the center frame of |search_block_| w.r.t. its first frame. |
| int search_block_center_offset_; |
| |
| // Index of the beginning of the |search_block_|, in frames. |
| int search_block_index_; |
| |
| // Number of Blocks to search to find the most similar one to the target |
| // frame. |
| int num_candidate_blocks_; |
| |
| // Index of the beginning of the target block, counted in frames. |
| int target_block_index_; |
| |
| // Overlap-and-add window size in frames. |
| int ola_window_size_; |
| |
| // The hop size of overlap-and-add in frames. This implementation assumes 50% |
| // overlap-and-add. |
| int ola_hop_size_; |
| |
| // Number of frames in |wsola_output_| that overlap-and-add is completed for |
| // them and can be copied to output if FillBuffer() is called. It also |
| // specifies the index where the next WSOLA window has to overlap-and-add. |
| int num_complete_frames_; |
| |
| // This stores a part of the output that is created but couldn't be rendered. |
| // Output is generated frame-by-frame which at some point might exceed the |
| // number of requested samples. Furthermore, due to overlap-and-add, |
| // the last half-window of the output is incomplete, which is stored in this |
| // buffer. |
| scoped_refptr<DecodedAudio> wsola_output_; |
| |
| // Overlap-and-add window. |
| scoped_array<float> ola_window_; |
| |
| // Transition window, used to update |optimal_block_| by a weighted sum of |
| // |optimal_block_| and |target_block_|. |
| scoped_array<float> transition_window_; |
| |
| // Auxiliary variables to avoid allocation in every iteration. |
| |
| // Stores the optimal block in every iteration. This is the most |
| // similar block to |target_block_| within |search_block_| and it is |
| // overlap-and-added to |wsola_output_|. |
| scoped_refptr<DecodedAudio> optimal_block_; |
| |
| // A block of data that search is performed over to find the |optimal_block_|. |
| scoped_refptr<DecodedAudio> search_block_; |
| |
| // Stores the target block, denoted as |target| above. |search_block_| is |
| // searched for a block (|optimal_block_|) that is most similar to |
| // |target_block_|. |
| scoped_refptr<DecodedAudio> target_block_; |
| |
| // The initial and maximum capacity calculated by Initialize(). |
| int initial_capacity_; |
| int max_capacity_; |
| |
| SB_DISALLOW_COPY_AND_ASSIGN(AudioTimeStretcher); |
| }; |
| |
| } // namespace filter |
| } // namespace player |
| } // namespace starboard |
| } // namespace shared |
| } // namespace starboard |
| |
| #endif // STARBOARD_SHARED_STARBOARD_PLAYER_FILTER_AUDIO_TIME_STRETCHER_H_ |