diff options
Diffstat (limited to 'libs/audiographer/audiographer/general')
10 files changed, 131 insertions, 131 deletions
diff --git a/libs/audiographer/audiographer/general/analyser.h b/libs/audiographer/audiographer/general/analyser.h index 9bd49b33c8..c912664cf0 100644 --- a/libs/audiographer/audiographer/general/analyser.h +++ b/libs/audiographer/audiographer/general/analyser.h @@ -29,7 +29,7 @@ namespace AudioGrapher class LIBAUDIOGRAPHER_API Analyser : public LoudnessReader { public: - Analyser (float sample_rate, unsigned int channels, framecnt_t bufsize, framecnt_t n_samples); + Analyser (float sample_rate, unsigned int channels, samplecnt_t bufsize, samplecnt_t n_samples); ~Analyser (); void process (ProcessContext<float> const & c); ARDOUR::ExportAnalysisPtr result (); @@ -48,10 +48,10 @@ class LIBAUDIOGRAPHER_API Analyser : public LoudnessReader ARDOUR::ExportAnalysis _result; - framecnt_t _n_samples; - framecnt_t _pos; - framecnt_t _spp; - framecnt_t _fpp; + samplecnt_t _n_samples; + samplecnt_t _pos; + samplecnt_t _spp; + samplecnt_t _fpp; float* _hann_window; uint32_t _fft_data_size; diff --git a/libs/audiographer/audiographer/general/chunker.h b/libs/audiographer/audiographer/general/chunker.h index 466a333655..28ac79b603 100644 --- a/libs/audiographer/audiographer/general/chunker.h +++ b/libs/audiographer/audiographer/general/chunker.h @@ -10,7 +10,7 @@ namespace AudioGrapher { -/// A class that chunks process cycles into equal sized frames +/// A class that chunks process cycles into equal sized samples template<typename T = DefaultSampleType> class /*LIBAUDIOGRAPHER_API*/ Chunker : public ListedSource<T> @@ -21,7 +21,7 @@ class /*LIBAUDIOGRAPHER_API*/ Chunker /** Constructs a new Chunker with a constant chunk size. * \n NOT RT safe */ - Chunker (framecnt_t chunk_size) + Chunker (samplecnt_t chunk_size) : chunk_size (chunk_size) , position (0) { @@ -42,29 +42,29 @@ class /*LIBAUDIOGRAPHER_API*/ Chunker { check_flags (*this, context); - framecnt_t frames_left = context.frames(); - framecnt_t input_position = 0; + samplecnt_t samples_left = context.samples(); + samplecnt_t input_position = 0; - while (position + frames_left >= chunk_size) { + while (position + samples_left >= chunk_size) { // Copy from context to buffer - framecnt_t const frames_to_copy = chunk_size - position; - TypeUtils<T>::copy (&context.data()[input_position], &buffer[position], frames_to_copy); + samplecnt_t const samples_to_copy = chunk_size - position; + TypeUtils<T>::copy (&context.data()[input_position], &buffer[position], samples_to_copy); // Update counters position = 0; - input_position += frames_to_copy; - frames_left -= frames_to_copy; + input_position += samples_to_copy; + samples_left -= samples_to_copy; // Output whole buffer ProcessContext<T> c_out (context, buffer, chunk_size); - if (frames_left) { c_out.remove_flag(ProcessContext<T>::EndOfInput); } + if (samples_left) { c_out.remove_flag(ProcessContext<T>::EndOfInput); } ListedSource<T>::output (c_out); } - if (frames_left) { + if (samples_left) { // Copy the rest of the data - TypeUtils<T>::copy (&context.data()[input_position], &buffer[position], frames_left); - position += frames_left; + TypeUtils<T>::copy (&context.data()[input_position], &buffer[position], samples_left); + position += samples_left; } if (context.has_flag (ProcessContext<T>::EndOfInput) && position > 0) { @@ -75,8 +75,8 @@ class /*LIBAUDIOGRAPHER_API*/ Chunker using Sink<T>::process; private: - framecnt_t chunk_size; - framecnt_t position; + samplecnt_t chunk_size; + samplecnt_t position; T * buffer; }; diff --git a/libs/audiographer/audiographer/general/deinterleaver.h b/libs/audiographer/audiographer/general/deinterleaver.h index 63b6c95589..491c07fcb6 100644 --- a/libs/audiographer/audiographer/general/deinterleaver.h +++ b/libs/audiographer/audiographer/general/deinterleaver.h @@ -26,7 +26,7 @@ class /*LIBAUDIOGRAPHER_API*/ DeInterleaver /// Constructor. \n RT safe DeInterleaver() : channels (0) - , max_frames (0) + , max_samples (0) , buffer (0) {} @@ -35,12 +35,12 @@ class /*LIBAUDIOGRAPHER_API*/ DeInterleaver typedef boost::shared_ptr<Source<T> > SourcePtr; /// Inits the deinterleaver. Must be called before using. \n Not RT safe - void init (unsigned int num_channels, framecnt_t max_frames_per_channel) + void init (unsigned int num_channels, samplecnt_t max_samples_per_channel) { reset(); channels = num_channels; - max_frames = max_frames_per_channel; - buffer = new T[max_frames]; + max_samples = max_samples_per_channel; + buffer = new T[max_samples]; for (unsigned int i = 0; i < channels; ++i) { outputs.push_back (OutputPtr (new IdentityVertex<T>)); @@ -60,28 +60,28 @@ class /*LIBAUDIOGRAPHER_API*/ DeInterleaver /// Deinterleaves data and outputs it to the outputs. \n RT safe void process (ProcessContext<T> const & c) { - framecnt_t frames = c.frames(); + samplecnt_t samples = c.samples(); T const * data = c.data(); - framecnt_t const frames_per_channel = frames / channels; + samplecnt_t const samples_per_channel = samples / channels; if (throw_level (ThrowProcess) && c.channels() != channels) { throw Exception (*this, "wrong amount of channels given to process()"); } - if (throw_level (ThrowProcess) && frames_per_channel > max_frames) { - throw Exception (*this, "too many frames given to process()"); + if (throw_level (ThrowProcess) && samples_per_channel > max_samples) { + throw Exception (*this, "too many samples given to process()"); } unsigned int channel = 0; for (typename std::vector<OutputPtr>::iterator it = outputs.begin(); it != outputs.end(); ++it, ++channel) { if (!*it) { continue; } - for (unsigned int i = 0; i < frames_per_channel; ++i) { + for (unsigned int i = 0; i < samples_per_channel; ++i) { buffer[i] = data[channel + (channels * i)]; } - ProcessContext<T> c_out (c, buffer, frames_per_channel, 1); + ProcessContext<T> c_out (c, buffer, samples_per_channel, 1); (*it)->process (c_out); } } @@ -96,12 +96,12 @@ class /*LIBAUDIOGRAPHER_API*/ DeInterleaver delete [] buffer; buffer = 0; channels = 0; - max_frames = 0; + max_samples = 0; } std::vector<OutputPtr> outputs; unsigned int channels; - framecnt_t max_frames; + samplecnt_t max_samples; T * buffer; }; diff --git a/libs/audiographer/audiographer/general/interleaver.h b/libs/audiographer/audiographer/general/interleaver.h index c1b5e92cfe..7ea1be1ab8 100644 --- a/libs/audiographer/audiographer/general/interleaver.h +++ b/libs/audiographer/audiographer/general/interleaver.h @@ -25,20 +25,20 @@ class /*LIBAUDIOGRAPHER_API*/ Interleaver /// Constructs an interleaver \n RT safe Interleaver() : channels (0) - , max_frames (0) + , max_samples (0) , buffer (0) {} ~Interleaver() { reset(); } /// Inits the interleaver. Must be called before using. \n Not RT safe - void init (unsigned int num_channels, framecnt_t max_frames_per_channel) + void init (unsigned int num_channels, samplecnt_t max_samples_per_channel) { reset(); channels = num_channels; - max_frames = max_frames_per_channel; + max_samples = max_samples_per_channel; - buffer = new T[channels * max_frames]; + buffer = new T[channels * max_samples]; for (unsigned int i = 0; i < channels; ++i) { inputs.push_back (InputPtr (new Input (*this, i))); @@ -63,27 +63,27 @@ class /*LIBAUDIOGRAPHER_API*/ Interleaver { public: Input (Interleaver & parent, unsigned int channel) - : frames_written (0), parent (parent), channel (channel) {} + : samples_written (0), parent (parent), channel (channel) {} void process (ProcessContext<T> const & c) { if (parent.throw_level (ThrowProcess) && c.channels() > 1) { throw Exception (*this, "Data input has more than on channel"); } - if (parent.throw_level (ThrowStrict) && frames_written) { + if (parent.throw_level (ThrowStrict) && samples_written) { throw Exception (*this, "Input channels out of sync"); } - frames_written = c.frames(); + samples_written = c.samples(); parent.write_channel (c, channel); } using Sink<T>::process; - framecnt_t frames() { return frames_written; } - void reset() { frames_written = 0; } + samplecnt_t samples() { return samples_written; } + void reset() { samples_written = 0; } private: - framecnt_t frames_written; + samplecnt_t samples_written; Interleaver & parent; unsigned int channel; }; @@ -94,7 +94,7 @@ class /*LIBAUDIOGRAPHER_API*/ Interleaver delete [] buffer; buffer = 0; channels = 0; - max_frames = 0; + max_samples = 0; } void reset_channels () @@ -107,44 +107,44 @@ class /*LIBAUDIOGRAPHER_API*/ Interleaver void write_channel (ProcessContext<T> const & c, unsigned int channel) { - if (throw_level (ThrowProcess) && c.frames() > max_frames) { + if (throw_level (ThrowProcess) && c.samples() > max_samples) { reset_channels(); - throw Exception (*this, "Too many frames given to an input"); + throw Exception (*this, "Too many samples given to an input"); } - for (unsigned int i = 0; i < c.frames(); ++i) { + for (unsigned int i = 0; i < c.samples(); ++i) { buffer[channel + (channels * i)] = c.data()[i]; } - framecnt_t const ready_frames = ready_to_output(); - if (ready_frames) { - ProcessContext<T> c_out (c, buffer, ready_frames, channels); + samplecnt_t const ready_samples = ready_to_output(); + if (ready_samples) { + ProcessContext<T> c_out (c, buffer, ready_samples, channels); ListedSource<T>::output (c_out); reset_channels (); } } - framecnt_t ready_to_output() + samplecnt_t ready_to_output() { - framecnt_t ready_frames = inputs[0]->frames(); - if (!ready_frames) { return 0; } + samplecnt_t ready_samples = inputs[0]->samples(); + if (!ready_samples) { return 0; } for (unsigned int i = 1; i < channels; ++i) { - framecnt_t const frames = inputs[i]->frames(); - if (!frames) { return 0; } - if (throw_level (ThrowProcess) && frames != ready_frames) { - init (channels, max_frames); - throw Exception (*this, "Frames count out of sync"); + samplecnt_t const samples = inputs[i]->samples(); + if (!samples) { return 0; } + if (throw_level (ThrowProcess) && samples != ready_samples) { + init (channels, max_samples); + throw Exception (*this, "Samples count out of sync"); } } - return ready_frames * channels; + return ready_samples * channels; } typedef boost::shared_ptr<Input> InputPtr; std::vector<InputPtr> inputs; unsigned int channels; - framecnt_t max_frames; + samplecnt_t max_samples; T * buffer; }; diff --git a/libs/audiographer/audiographer/general/loudness_reader.h b/libs/audiographer/audiographer/general/loudness_reader.h index c86fd521da..8c9ccb4149 100644 --- a/libs/audiographer/audiographer/general/loudness_reader.h +++ b/libs/audiographer/audiographer/general/loudness_reader.h @@ -32,7 +32,7 @@ namespace AudioGrapher class LIBAUDIOGRAPHER_API LoudnessReader : public ListedSource<float>, public Sink<float> { public: - LoudnessReader (float sample_rate, unsigned int channels, framecnt_t bufsize); + LoudnessReader (float sample_rate, unsigned int channels, samplecnt_t bufsize); ~LoudnessReader (); void reset (); @@ -52,8 +52,8 @@ class LIBAUDIOGRAPHER_API LoudnessReader : public ListedSource<float>, public Si float _sample_rate; unsigned int _channels; - framecnt_t _bufsize; - framecnt_t _pos; + samplecnt_t _bufsize; + samplecnt_t _pos; float* _bufs[2]; }; diff --git a/libs/audiographer/audiographer/general/normalizer.h b/libs/audiographer/audiographer/general/normalizer.h index e5f73a0f08..cd4e375db2 100644 --- a/libs/audiographer/audiographer/general/normalizer.h +++ b/libs/audiographer/audiographer/general/normalizer.h @@ -28,7 +28,7 @@ public: * non-const ProcessContexts are given to \a process() . * \n Not RT safe */ - void alloc_buffer(framecnt_t frames); + void alloc_buffer(samplecnt_t samples); /// Process a const ProcessContext \see alloc_buffer() \n RT safe void process (ProcessContext<float> const & c); @@ -42,7 +42,7 @@ private: float gain; float * buffer; - framecnt_t buffer_size; + samplecnt_t buffer_size; }; diff --git a/libs/audiographer/audiographer/general/peak_reader.h b/libs/audiographer/audiographer/general/peak_reader.h index 8bf0faa792..bb04239544 100644 --- a/libs/audiographer/audiographer/general/peak_reader.h +++ b/libs/audiographer/audiographer/general/peak_reader.h @@ -25,7 +25,7 @@ class /*LIBAUDIOGRAPHER_API*/ PeakReader : public ListedSource<float>, public Si /// Finds peaks from the data \n RT safe void process (ProcessContext<float> const & c) { - peak = Routines::compute_peak (c.data(), c.frames(), peak); + peak = Routines::compute_peak (c.data(), c.samples(), peak); ListedSource<float>::output(c); } diff --git a/libs/audiographer/audiographer/general/sample_format_converter.h b/libs/audiographer/audiographer/general/sample_format_converter.h index 96dd6aa72a..9a79fc927c 100644 --- a/libs/audiographer/audiographer/general/sample_format_converter.h +++ b/libs/audiographer/audiographer/general/sample_format_converter.h @@ -35,13 +35,13 @@ class LIBAUDIOGRAPHER_API SampleFormatConverter ~SampleFormatConverter (); /** Initialize and allocate buffers for processing. - * \param max_frames maximum number of frames that is allowed to be used in calls to \a process() + * \param max_samples maximum number of samples that is allowed to be used in calls to \a process() * \param type dither type from \a DitherType * \param data_width data with in bits * \note If the non-const version of process() is used with floats, * there is no need to call this function. */ - void init (framecnt_t max_frames, int type, int data_width); + void init (samplecnt_t max_samples, int type, int data_width); /// Set whether or not clipping to [-1.0, 1.0] should occur when TOut = float. Clipping is off by default void set_clip_floats (bool yn) { clip_floats = yn; } @@ -54,12 +54,12 @@ class LIBAUDIOGRAPHER_API SampleFormatConverter private: void reset(); - void init_common (framecnt_t max_frames); // not-template-specialized part of init - void check_frame_and_channel_count (framecnt_t frames, ChannelCount channels_); + void init_common (samplecnt_t max_samples); // not-template-specialized part of init + void check_sample_and_channel_count (samplecnt_t samples, ChannelCount channels_); ChannelCount channels; GDither dither; - framecnt_t data_out_size; + samplecnt_t data_out_size; TOut * data_out; bool clip_floats; diff --git a/libs/audiographer/audiographer/general/silence_trimmer.h b/libs/audiographer/audiographer/general/silence_trimmer.h index 7cfa0658f1..e09c348c52 100644 --- a/libs/audiographer/audiographer/general/silence_trimmer.h +++ b/libs/audiographer/audiographer/general/silence_trimmer.h @@ -30,7 +30,7 @@ struct SilenceTester<float> { }; -/// Removes and adds silent frames to beginning and/or end of stream +/// Removes and adds silent samples to beginning and/or end of stream template<typename T = DefaultSampleType> class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer : public ListedSource<T> @@ -41,7 +41,7 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer public: /// Constructor, \see reset() \n Not RT safe - SilenceTrimmer(framecnt_t silence_buffer_size_ = 1024, float thresh_dB = -INFINITY) + SilenceTrimmer(samplecnt_t silence_buffer_size_ = 1024, float thresh_dB = -INFINITY) : silence_buffer_size (0) , silence_buffer (0) , tester (thresh_dB) @@ -60,7 +60,7 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer * This also defines the maximum length of output process context * which can be output during long intermediate silence. */ - void reset (framecnt_t silence_buffer_size_ = 1024) + void reset (samplecnt_t silence_buffer_size_ = 1024) { if (throw_level (ThrowObject) && silence_buffer_size_ == 0) { throw Exception (*this, @@ -78,34 +78,34 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer processing_finished = false; trim_beginning = false; trim_end = false; - silence_frames = 0; + silence_samples = 0; max_output_frames = 0; add_to_beginning = 0; add_to_end = 0; } - /** Tells that \a frames_per_channel frames of silence per channel should be added to beginning + /** Tells that \a samples_per_channel samples of silence per channel should be added to beginning * Needs to be called before starting processing. * \n RT safe */ - void add_silence_to_beginning (framecnt_t frames_per_channel) + void add_silence_to_beginning (samplecnt_t samples_per_channel) { if (throw_level (ThrowObject) && processed_data) { throw Exception(*this, "Tried to add silence to beginning after processing started"); } - add_to_beginning = frames_per_channel; + add_to_beginning = samples_per_channel; } - /** Tells that \a frames_per_channel frames of silence per channel should be added to end + /** Tells that \a samples_per_channel samples of silence per channel should be added to end * Needs to be called before end is reached. * \n RT safe */ - void add_silence_to_end (framecnt_t frames_per_channel) + void add_silence_to_end (samplecnt_t samples_per_channel) { if (throw_level (ThrowObject) && processed_data) { throw Exception(*this, "Tried to add silence to end after processing started"); } - add_to_end = frames_per_channel; + add_to_end = samples_per_channel; } /** Tells whether ot nor silence should be trimmed from the beginning @@ -166,17 +166,17 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer * may lend itself for some inspiration. */ - framecnt_t output_start_index = 0; - framecnt_t output_sample_count = c.frames(); + samplecnt_t output_start_index = 0; + samplecnt_t output_sample_count = c.samples(); if (!processed_data) { if (trim_beginning) { - framecnt_t first_non_silent_frame_index = 0; - if (find_first_non_silent_frame (c, first_non_silent_frame_index)) { + samplecnt_t first_non_silent_sample_index = 0; + if (find_first_non_silent_sample (c, first_non_silent_sample_index)) { // output from start of non-silent data until end of buffer // output_sample_count may also be altered in trim end - output_start_index = first_non_silent_frame_index; - output_sample_count = c.frames() - first_non_silent_frame_index; + output_start_index = first_non_silent_sample_index; + output_sample_count = c.samples() - first_non_silent_sample_index; processed_data = true; } else { // keep entering this block until non-silence is found to trim @@ -189,35 +189,35 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer // This block won't be called again so add silence to beginning if (processed_data && add_to_beginning) { add_to_beginning *= c.channels (); - output_silence_frames (c, add_to_beginning); + output_silence_samples (c, add_to_beginning); } } if (processed_data) { if (trim_end) { - framecnt_t first_non_silent_frame_index = 0; - if (find_first_non_silent_frame (c, first_non_silent_frame_index)) { + samplecnt_t first_non_silent_sample_index = 0; + if (find_first_non_silent_sample (c, first_non_silent_sample_index)) { // context buffer contains non-silent data, flush any intermediate silence - output_silence_frames (c, silence_frames); + output_silence_samples (c, silence_samples); - framecnt_t silent_frame_index = 0; - find_last_silent_frame_reverse (c, silent_frame_index); + samplecnt_t silent_sample_index = 0; + find_last_silent_sample_reverse (c, silent_sample_index); // Count of samples at end of block that are "silent", may be zero. - framecnt_t silent_end_samples = c.frames () - silent_frame_index; - framecnt_t samples_before_silence = c.frames() - silent_end_samples; + samplecnt_t silent_end_samples = c.samples () - silent_sample_index; + samplecnt_t samples_before_silence = c.samples() - silent_end_samples; - assert (samples_before_silence + silent_end_samples == c.frames ()); + assert (samples_before_silence + silent_end_samples == c.samples ()); // output_start_index may be non-zero if start trim occurred above output_sample_count = samples_before_silence - output_start_index; // keep track of any silent samples not output - silence_frames = silent_end_samples; + silence_samples = silent_end_samples; } else { // whole context buffer is silent output nothing - silence_frames += c.frames (); + silence_samples += c.samples (); output_sample_count = 0; } } @@ -230,7 +230,7 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer // Finally, if in last process call, add silence to end if (processing_finished && processed_data && add_to_end) { add_to_end *= c.channels(); - output_silence_frames (c, add_to_end); + output_silence_samples (c, add_to_end); } if (processing_finished) { @@ -250,13 +250,13 @@ class /*LIBAUDIOGRAPHER_API*/ SilenceTrimmer private: - bool find_first_non_silent_frame (ProcessContext<T> const & c, framecnt_t & result_frame) + bool find_first_non_silent_sample (ProcessContext<T> const & c, samplecnt_t & result_sample) { - for (framecnt_t i = 0; i < c.frames(); ++i) { + for (samplecnt_t i = 0; i < c.samples(); ++i) { if (!tester.is_silent (c.data()[i])) { - result_frame = i; + result_sample = i; // Round down to nearest interleaved "frame" beginning - result_frame -= result_frame % c.channels(); + result_sample -= result_sample % c.channels(); return true; } } @@ -264,43 +264,43 @@ private: } /** - * Reverse find the last silent frame index. If the last sample in the + * Reverse find the last silent sample index. If the last sample in the * buffer is non-silent the index will be one past the end of the buffer and - * equal to c.frames(). e.g silent_end_samples = c.frames() - result_frame + * equal to c.samples(). e.g silent_end_samples = c.samples() - result_sample * - * @return true if result_frame index is valid, false if there were only + * @return true if result_sample index is valid, false if there were only * silent samples in the context buffer */ - bool find_last_silent_frame_reverse (ProcessContext<T> const & c, framecnt_t & result_frame) + bool find_last_silent_sample_reverse (ProcessContext<T> const & c, samplecnt_t & result_sample) { - framecnt_t last_sample_index = c.frames() - 1; + samplecnt_t last_sample_index = c.samples() - 1; - for (framecnt_t i = last_sample_index; i >= 0; --i) { + for (samplecnt_t i = last_sample_index; i >= 0; --i) { if (!tester.is_silent (c.data()[i])) { - result_frame = i; + result_sample = i; // Round down to nearest interleaved "frame" beginning - result_frame -= result_frame % c.channels(); - // Round up to return the "last" silent interleaved frame - result_frame += c.channels(); + result_sample -= result_sample % c.channels(); + // Round up to return the "last" silent interleaved sample + result_sample += c.channels(); return true; } } return false; } - void output_silence_frames (ProcessContext<T> const & c, framecnt_t & total_frames) + void output_silence_samples (ProcessContext<T> const & c, samplecnt_t & total_samples) { assert (!c.has_flag (ProcessContext<T>::EndOfInput)); - while (total_frames > 0) { - framecnt_t frames = std::min (silence_buffer_size, total_frames); + while (total_samples > 0) { + samplecnt_t samples = std::min (silence_buffer_size, total_samples); if (max_output_frames) { - frames = std::min (frames, max_output_frames); + samples = std::min (samples, max_output_frames); } - frames -= frames % c.channels(); + samples -= samples % c.channels(); - total_frames -= frames; - ConstProcessContext<T> c_out (c, silence_buffer, frames); + total_samples -= samples; + ConstProcessContext<T> c_out (c, silence_buffer, samples); ListedSource<T>::output (c_out); } } @@ -311,13 +311,13 @@ private: bool trim_beginning; bool trim_end; - framecnt_t silence_frames; - framecnt_t max_output_frames; + samplecnt_t silence_samples; + samplecnt_t max_output_frames; - framecnt_t add_to_beginning; - framecnt_t add_to_end; + samplecnt_t add_to_beginning; + samplecnt_t add_to_end; - framecnt_t silence_buffer_size; + samplecnt_t silence_buffer_size; T * silence_buffer; SilenceTester<T> tester; diff --git a/libs/audiographer/audiographer/general/sr_converter.h b/libs/audiographer/audiographer/general/sr_converter.h index a2e94d9bc0..8edc4b8c99 100644 --- a/libs/audiographer/audiographer/general/sr_converter.h +++ b/libs/audiographer/audiographer/general/sr_converter.h @@ -26,10 +26,10 @@ class LIBAUDIOGRAPHER_API SampleRateConverter ~SampleRateConverter (); /// Init converter \n Not RT safe - void init (framecnt_t in_rate, framecnt_t out_rate, int quality = 0); + void init (samplecnt_t in_rate, samplecnt_t out_rate, int quality = 0); - /// Returns max amount of frames that will be output \n RT safe - framecnt_t allocate_buffers (framecnt_t max_frames); + /// Returns max amount of samples that will be output \n RT safe + samplecnt_t allocate_buffers (samplecnt_t max_samples); /** Does sample rate conversion. * Note that outpt size may vary a lot. @@ -47,14 +47,14 @@ class LIBAUDIOGRAPHER_API SampleRateConverter bool active; uint32_t channels; - framecnt_t max_frames_in; + samplecnt_t max_samples_in; float * leftover_data; - framecnt_t leftover_frames; - framecnt_t max_leftover_frames; + samplecnt_t leftover_samples; + samplecnt_t max_leftover_samples; float * data_out; - framecnt_t data_out_size; + samplecnt_t data_out_size; SRC_DATA src_data; SRC_STATE* src_state; |