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// 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.
// AudioRendererAlgorithm 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.
// AudioRendererAlgorithm 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
// are preserved. See for a more elaborate
// description of the algorithm.
// Audio at very low or very high playback rates are muted to preserve quality.
#include <stdint.h>
#include <memory>
#include <vector>
#include "base/macros.h"
#include "base/memory/ref_counted.h"
#include "media/base/audio_buffer.h"
#include "media/base/audio_buffer_queue.h"
#include "media/base/audio_parameters.h"
#include "media/base/media_log.h"
#include "media/base/multi_channel_resampler.h"
#include "third_party/abseil-cpp/absl/types/optional.h"
namespace media {
class AudioBus;
class MEDIA_EXPORT AudioRendererAlgorithm {
AudioRendererAlgorithm(MediaLog* media_log);
AudioRendererAlgorithm(MediaLog* media_log,
AudioRendererAlgorithmParameters params);
AudioRendererAlgorithm(const AudioRendererAlgorithm&) = delete;
AudioRendererAlgorithm& operator=(const AudioRendererAlgorithm&) = delete;
// Initializes this object with information about the audio stream.
void Initialize(const AudioParameters& params, bool is_encrypted);
// Allows clients to specify which channels will be considered by the
// algorithm when adapting for playback rate, other channels will be muted.
// Useful to avoid performance overhead of the adapatation algorithm. Must
// only be called after Initialize(); may be called multiple times if the
// mask changes.
// E.g., If |channel_mask| is [true, false] only the first channel will be
// used to construct the playback rate adapated signal. This is useful if
// channel upmixing has been performed prior to this point.
void SetChannelMask(std::vector<bool> channel_mask);
// 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|.
int FillBuffer(AudioBus* dest,
int dest_offset,
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(scoped_refptr<AudioBuffer> buffer_in);
// Sets a target queue latency. This target will be clamped and stored in
// |playback_threshold_|. It may also cause an increase in |capacity_|. A
// value of nullopt indicates the algorithm should restore the default value.
void SetLatencyHint(absl::optional<base::TimeDelta> latency_hint);
// Sets a flag indicating whether apply pitch adjustments when playing back
// at rates other than 1.0. Concretely, we use WSOLA when this is true, and
// resampling when this is false.
void SetPreservesPitch(bool preserves_pitch);
// Returns true if the |audio_buffer_| is >= |playback_threshold_|.
bool IsQueueAdequateForPlayback();
// Returns the required size for |audio_buffer_| to be "adequate for
// playback". See IsQueueAdequateForPlayback().
int QueuePlaybackThreshold() const { return playback_threshold_; }
// Returns true if |audio_buffer_| is >= |capacity_|.
bool IsQueueFull();
// Returns the capacity of |audio_buffer_| in frames.
int QueueCapacity() const { return capacity_; }
// Increase the |playback_threshold_| and |capacity_| of |audio_buffer_| if
// possible. Should not be called if a custom |playback_threshold_| was
// specified.
void IncreasePlaybackThreshold();
// Sets a flag to bypass underflow detection, to read out all remaining data.
void MarkEndOfStream();
// Returns an estimate of the amount of memory (in bytes) used for frames.
int64_t GetMemoryUsage() const;
// Returns the total number of frames in |audio_buffer_| as well as
// unconsumed input frames in the |resampler_|. The returned value may be
// larger than QueueCapacity() in the event that EnqueueBuffer() delivered
// more data than |audio_buffer_| was intending to hold.
int BufferedFrames() const;
// Returns the effective delay in output frames at the given |playback rate|.
// Effectively this tells the caller, if new audio is enqueued via
// EnqueueBuffer(), how many frames must be read via FillBuffer() at the
// |playback_rate| before the new audio is read out. Note that this is
// approximate, since due to WSOLA the audio output doesn't always directly
// correspond to the audio input (some samples may be duplicated or skipped).
double DelayInFrames(double playback_rate) const;
// Returns the samples per second for this audio stream.
int samples_per_second() const { return samples_per_second_; }
std::vector<bool> channel_mask_for_testing() { return channel_mask_; }
enum class FillBufferMode {
// Remove buffered data that will be outdated if we switch fill mode.
void SetFillBufferMode(FillBufferMode mode);
// 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, AudioBus* 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, AudioBus* 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;
// Creates or recreates |target_block_wrapper_| and |search_block_wrapper_|
// after a |channel_mask_| change. May be called at anytime after a channel
// mask has been specified.
void CreateSearchWrappers();
// Uses |resampler_| to speed up or slowdown audio, by using a resampling
// ratio of |playback_rate|.
int ResampleAndFill(AudioBus* dest,
int dest_offset,
int requested_frames,
double playback_rate);
// Called by |resampler_| to get more audio data.
void OnResamplerRead(int frame_delay, AudioBus* audio_bus);
MediaLog* media_log_;
// Parameters.
AudioRendererAlgorithmParameters audio_renderer_algorithm_params_;
// Number of channels in audio stream.
int channels_;
// Sample rate of audio stream.
int samples_per_second_;
// Is compressed audio output
bool is_bitstream_format_;
// Buffered audio data.
AudioBufferQueue audio_buffer_;
// Hint to adjust |playback_threshold_| as a means of controlling playback
// start latency. See SetLatencyHint();
absl::optional<base::TimeDelta> latency_hint_;
// Whether to apply pitch adjusments or not when playing back at rates other
// than 1.0. In other words, we use WSOLA to preserve pitch when this is on,
// and resampling when this
bool preserves_pitch_ = true;
// How many frames to have in queue before beginning playback.
int64_t playback_threshold_;
// Minimum allowed value for |plabyack_threshold_| calculated by Initialize().
int64_t min_playback_threshold_;
// How many frames to have in the queue before we report the queue is full.
int64_t 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_;
bool reached_end_of_stream_ = false;
// Used to replace WSOLA algorithm at playback speeds close to 1.0. This is to
// prevent noticeable audio artifacts introduced by WSOLA, at the expense of
// changing the pitch of the audio.
std::unique_ptr<MultiChannelResampler> resampler_;
// True when the last call to OnResamplerRead() only gave silence to
// |resampler_|. Used to determine whether or not we have played out all the
// valid audio from |resampler.BufferedFrames()|.
bool resampler_only_has_silence_ = false;
// 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.
std::unique_ptr<AudioBus> wsola_output_;
// Overlap-and-add window.
std::unique_ptr<float[]> ola_window_;
// Transition window, used to update |optimal_block_| by a weighted sum of
// |optimal_block_| and |target_block_|.
std::unique_ptr<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_|.
std::unique_ptr<AudioBus> optimal_block_;
// A block of data that search is performed over to find the |optimal_block_|.
std::unique_ptr<AudioBus> 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_|.
std::unique_ptr<AudioBus> target_block_;
// Active channels to consider while searching. Used to speed up WSOLA
// processing by ignoring always muted channels. Wrappers are always
// constructed during Initialize() and have <= |channels_|.
std::vector<bool> channel_mask_;
std::unique_ptr<AudioBus> search_block_wrapper_;
std::unique_ptr<AudioBus> target_block_wrapper_;
// The initial and maximum capacity calculated by Initialize().
int64_t initial_capacity_;
int64_t max_capacity_;
FillBufferMode last_mode_ = FillBufferMode::kPassthrough;
} // namespace media