612 lines
23 KiB
C++
Executable File
612 lines
23 KiB
C++
Executable File
/*
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* Copyright © 2016 Mozilla Foundation
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*
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* This program is made available under an ISC-style license. See the
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* accompanying file LICENSE for details.
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*/
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#if !defined(CUBEB_RESAMPLER_INTERNAL)
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#define CUBEB_RESAMPLER_INTERNAL
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#include <cmath>
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#include <cassert>
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#include <algorithm>
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#include <memory>
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#ifdef CUBEB_GECKO_BUILD
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#include "mozilla/UniquePtr.h"
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// In libc++, symbols such as std::unique_ptr may be defined in std::__1.
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// The _LIBCPP_BEGIN_NAMESPACE_STD and _LIBCPP_END_NAMESPACE_STD macros
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// will expand to the correct namespace.
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#ifdef _LIBCPP_BEGIN_NAMESPACE_STD
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#define MOZ_BEGIN_STD_NAMESPACE _LIBCPP_BEGIN_NAMESPACE_STD
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#define MOZ_END_STD_NAMESPACE _LIBCPP_END_NAMESPACE_STD
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#else
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#define MOZ_BEGIN_STD_NAMESPACE namespace std {
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#define MOZ_END_STD_NAMESPACE }
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#endif
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MOZ_BEGIN_STD_NAMESPACE
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using mozilla::DefaultDelete;
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using mozilla::UniquePtr;
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#define default_delete DefaultDelete
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#define unique_ptr UniquePtr
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MOZ_END_STD_NAMESPACE
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#endif
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#include "cubeb/cubeb.h"
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#include "cubeb_utils.h"
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#include "cubeb-speex-resampler.h"
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#include "cubeb_resampler.h"
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#include "cubeb_log.h"
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#include <stdio.h>
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/* This header file contains the internal C++ API of the resamplers, for testing. */
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// When dropping audio input frames to prevent building
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// an input delay, this function returns the number of frames
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// to keep in the buffer.
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// @parameter sample_rate The sample rate of the stream.
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// @return A number of frames to keep.
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uint32_t min_buffered_audio_frame(uint32_t sample_rate);
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int to_speex_quality(cubeb_resampler_quality q);
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struct cubeb_resampler {
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virtual long fill(void * input_buffer, long * input_frames_count,
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void * output_buffer, long frames_needed) = 0;
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virtual long latency() = 0;
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virtual ~cubeb_resampler() {}
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};
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/** Base class for processors. This is just used to share methods for now. */
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class processor {
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public:
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explicit processor(uint32_t channels)
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: channels(channels)
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{}
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protected:
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size_t frames_to_samples(size_t frames) const
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{
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return frames * channels;
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}
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size_t samples_to_frames(size_t samples) const
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{
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assert(!(samples % channels));
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return samples / channels;
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}
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/** The number of channel of the audio buffers to be resampled. */
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const uint32_t channels;
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};
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template<typename T>
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class passthrough_resampler : public cubeb_resampler
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, public processor {
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public:
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passthrough_resampler(cubeb_stream * s,
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cubeb_data_callback cb,
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void * ptr,
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uint32_t input_channels,
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uint32_t sample_rate);
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virtual long fill(void * input_buffer, long * input_frames_count,
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void * output_buffer, long output_frames);
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virtual long latency()
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{
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return 0;
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}
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void drop_audio_if_needed()
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{
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uint32_t to_keep = min_buffered_audio_frame(sample_rate);
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uint32_t available = samples_to_frames(internal_input_buffer.length());
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if (available > to_keep) {
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internal_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
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}
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}
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private:
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cubeb_stream * const stream;
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const cubeb_data_callback data_callback;
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void * const user_ptr;
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/* This allows to buffer some input to account for the fact that we buffer
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* some inputs. */
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auto_array<T> internal_input_buffer;
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uint32_t sample_rate;
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};
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/** Bidirectional resampler, can resample an input and an output stream, or just
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* an input stream or output stream. In this case a delay is inserted in the
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* opposite direction to keep the streams synchronized. */
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template<typename T, typename InputProcessing, typename OutputProcessing>
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class cubeb_resampler_speex : public cubeb_resampler {
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public:
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cubeb_resampler_speex(InputProcessing * input_processor,
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OutputProcessing * output_processor,
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cubeb_stream * s,
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cubeb_data_callback cb,
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void * ptr);
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virtual ~cubeb_resampler_speex();
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virtual long fill(void * input_buffer, long * input_frames_count,
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void * output_buffer, long output_frames_needed);
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virtual long latency()
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{
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if (input_processor && output_processor) {
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assert(input_processor->latency() == output_processor->latency());
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return input_processor->latency();
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} else if (input_processor) {
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return input_processor->latency();
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} else {
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return output_processor->latency();
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}
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}
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private:
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typedef long(cubeb_resampler_speex::*processing_callback)(T * input_buffer, long * input_frames_count, T * output_buffer, long output_frames_needed);
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long fill_internal_duplex(T * input_buffer, long * input_frames_count,
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T * output_buffer, long output_frames_needed);
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long fill_internal_input(T * input_buffer, long * input_frames_count,
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T * output_buffer, long output_frames_needed);
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long fill_internal_output(T * input_buffer, long * input_frames_count,
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T * output_buffer, long output_frames_needed);
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std::unique_ptr<InputProcessing> input_processor;
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std::unique_ptr<OutputProcessing> output_processor;
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processing_callback fill_internal;
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cubeb_stream * const stream;
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const cubeb_data_callback data_callback;
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void * const user_ptr;
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bool draining = false;
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};
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/** Handles one way of a (possibly) duplex resampler, working on interleaved
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* audio buffers of type T. This class is designed so that the number of frames
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* coming out of the resampler can be precisely controled. It manages its own
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* input buffer, and can use the caller's output buffer, or allocate its own. */
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template<typename T>
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class cubeb_resampler_speex_one_way : public processor {
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public:
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/** The sample type of this resampler, either 16-bit integers or 32-bit
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* floats. */
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typedef T sample_type;
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/** Construct a resampler resampling from #source_rate to #target_rate, that
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* can be arbitrary, strictly positive number.
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* @parameter channels The number of channels this resampler will resample.
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* @parameter source_rate The sample-rate of the audio input.
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* @parameter target_rate The sample-rate of the audio output.
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* @parameter quality A number between 0 (fast, low quality) and 10 (slow,
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* high quality). */
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cubeb_resampler_speex_one_way(uint32_t channels,
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uint32_t source_rate,
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uint32_t target_rate,
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int quality)
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: processor(channels)
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, resampling_ratio(static_cast<float>(source_rate) / target_rate)
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, source_rate(source_rate)
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, additional_latency(0)
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, leftover_samples(0)
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{
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int r;
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speex_resampler = speex_resampler_init(channels, source_rate,
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target_rate, quality, &r);
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assert(r == RESAMPLER_ERR_SUCCESS && "resampler allocation failure");
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uint32_t input_latency = speex_resampler_get_input_latency(speex_resampler);
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const size_t LATENCY_SAMPLES = 8192;
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T input_buffer[LATENCY_SAMPLES] = {};
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T output_buffer[LATENCY_SAMPLES] = {};
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uint32_t input_frame_count = input_latency;
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uint32_t output_frame_count = LATENCY_SAMPLES;
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assert(input_latency * channels <= LATENCY_SAMPLES);
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speex_resample(
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input_buffer,
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&input_frame_count,
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output_buffer,
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&output_frame_count);
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}
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/** Destructor, deallocate the resampler */
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virtual ~cubeb_resampler_speex_one_way()
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{
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speex_resampler_destroy(speex_resampler);
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}
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/* Fill the resampler with `input_frame_count` frames. */
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void input(T * input_buffer, size_t input_frame_count)
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{
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resampling_in_buffer.push(input_buffer,
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frames_to_samples(input_frame_count));
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}
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/** Outputs exactly `output_frame_count` into `output_buffer`.
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* `output_buffer` has to be at least `output_frame_count` long. */
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size_t output(T * output_buffer, size_t output_frame_count)
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{
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uint32_t in_len = samples_to_frames(resampling_in_buffer.length());
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uint32_t out_len = output_frame_count;
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speex_resample(resampling_in_buffer.data(), &in_len,
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output_buffer, &out_len);
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/* This shifts back any unresampled samples to the beginning of the input
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buffer. */
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resampling_in_buffer.pop(nullptr, frames_to_samples(in_len));
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return out_len;
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}
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size_t output_for_input(uint32_t input_frames)
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{
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return (size_t)floorf((input_frames + samples_to_frames(resampling_in_buffer.length()))
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/ resampling_ratio);
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}
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/** Returns a buffer containing exactly `output_frame_count` resampled frames.
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* The consumer should not hold onto the pointer. */
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T * output(size_t output_frame_count, size_t * input_frames_used)
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{
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if (resampling_out_buffer.capacity() < frames_to_samples(output_frame_count)) {
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resampling_out_buffer.reserve(frames_to_samples(output_frame_count));
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}
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uint32_t in_len = samples_to_frames(resampling_in_buffer.length());
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uint32_t out_len = output_frame_count;
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speex_resample(resampling_in_buffer.data(), &in_len,
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resampling_out_buffer.data(), &out_len);
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if (out_len < output_frame_count) {
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LOGV("underrun during resampling: got %u frames, expected %zu", (unsigned)out_len, output_frame_count);
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// silence the rightmost part
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T* data = resampling_out_buffer.data();
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for (uint32_t i = frames_to_samples(out_len); i < frames_to_samples(output_frame_count); i++) {
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data[i] = 0;
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}
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}
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/* This shifts back any unresampled samples to the beginning of the input
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buffer. */
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resampling_in_buffer.pop(nullptr, frames_to_samples(in_len));
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*input_frames_used = in_len;
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return resampling_out_buffer.data();
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}
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/** Get the latency of the resampler, in output frames. */
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uint32_t latency() const
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{
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/* The documentation of the resampler talks about "samples" here, but it
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* only consider a single channel here so it's the same number of frames. */
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int latency = 0;
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latency =
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speex_resampler_get_output_latency(speex_resampler) + additional_latency;
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assert(latency >= 0);
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return latency;
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}
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/** Returns the number of frames to pass in the input of the resampler to have
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* exactly `output_frame_count` resampled frames. This can return a number
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* slightly bigger than what is strictly necessary, but it guaranteed that the
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* number of output frames will be exactly equal. */
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uint32_t input_needed_for_output(int32_t output_frame_count) const
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{
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assert(output_frame_count >= 0); // Check overflow
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int32_t unresampled_frames_left = samples_to_frames(resampling_in_buffer.length());
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int32_t resampled_frames_left = samples_to_frames(resampling_out_buffer.length());
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float input_frames_needed =
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(output_frame_count - unresampled_frames_left) * resampling_ratio
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- resampled_frames_left;
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if (input_frames_needed < 0) {
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return 0;
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}
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return (uint32_t)ceilf(input_frames_needed);
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}
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/** Returns a pointer to the input buffer, that contains empty space for at
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* least `frame_count` elements. This is useful so that consumer can directly
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* write into the input buffer of the resampler. The pointer returned is
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* adjusted so that leftover data are not overwritten.
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*/
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T * input_buffer(size_t frame_count)
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{
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leftover_samples = resampling_in_buffer.length();
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resampling_in_buffer.reserve(leftover_samples +
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frames_to_samples(frame_count));
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return resampling_in_buffer.data() + leftover_samples;
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}
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/** This method works with `input_buffer`, and allows to inform the processor
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how much frames have been written in the provided buffer. */
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void written(size_t written_frames)
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{
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resampling_in_buffer.set_length(leftover_samples +
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frames_to_samples(written_frames));
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}
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void drop_audio_if_needed()
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{
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// Keep at most 100ms buffered.
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uint32_t available = samples_to_frames(resampling_in_buffer.length());
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uint32_t to_keep = min_buffered_audio_frame(source_rate);
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if (available > to_keep) {
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resampling_in_buffer.pop(nullptr, frames_to_samples(available - to_keep));
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}
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}
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private:
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/** Wrapper for the speex resampling functions to have a typed
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* interface. */
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void speex_resample(float * input_buffer, uint32_t * input_frame_count,
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float * output_buffer, uint32_t * output_frame_count)
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{
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#ifndef NDEBUG
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int rv;
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rv =
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#endif
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speex_resampler_process_interleaved_float(speex_resampler,
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input_buffer,
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input_frame_count,
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output_buffer,
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output_frame_count);
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assert(rv == RESAMPLER_ERR_SUCCESS);
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}
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void speex_resample(short * input_buffer, uint32_t * input_frame_count,
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short * output_buffer, uint32_t * output_frame_count)
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{
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#ifndef NDEBUG
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int rv;
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rv =
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#endif
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speex_resampler_process_interleaved_int(speex_resampler,
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input_buffer,
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input_frame_count,
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output_buffer,
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output_frame_count);
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assert(rv == RESAMPLER_ERR_SUCCESS);
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}
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/** The state for the speex resampler used internaly. */
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SpeexResamplerState * speex_resampler;
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/** Source rate / target rate. */
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const float resampling_ratio;
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const uint32_t source_rate;
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/** Storage for the input frames, to be resampled. Also contains
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* any unresampled frames after resampling. */
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auto_array<T> resampling_in_buffer;
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/* Storage for the resampled frames, to be passed back to the caller. */
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auto_array<T> resampling_out_buffer;
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/** Additional latency inserted into the pipeline for synchronisation. */
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uint32_t additional_latency;
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/** When `input_buffer` is called, this allows tracking the number of samples
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that were in the buffer. */
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uint32_t leftover_samples;
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};
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/** This class allows delaying an audio stream by `frames` frames. */
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template<typename T>
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class delay_line : public processor {
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public:
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/** Constructor
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* @parameter frames the number of frames of delay.
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* @parameter channels the number of channels of this delay line.
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* @parameter sample_rate sample-rate of the audio going through this delay line */
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delay_line(uint32_t frames, uint32_t channels, uint32_t sample_rate)
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: processor(channels)
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, length(frames)
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, leftover_samples(0)
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, sample_rate(sample_rate)
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{
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/* Fill the delay line with some silent frames to add latency. */
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delay_input_buffer.push_silence(frames * channels);
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}
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/** Push some frames into the delay line.
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* @parameter buffer the frames to push.
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* @parameter frame_count the number of frames in #buffer. */
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void input(T * buffer, uint32_t frame_count)
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{
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delay_input_buffer.push(buffer, frames_to_samples(frame_count));
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}
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/** Pop some frames from the internal buffer, into a internal output buffer.
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* @parameter frames_needed the number of frames to be returned.
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* @return a buffer containing the delayed frames. The consumer should not
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* hold onto the pointer. */
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T * output(uint32_t frames_needed, size_t * input_frames_used)
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{
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if (delay_output_buffer.capacity() < frames_to_samples(frames_needed)) {
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delay_output_buffer.reserve(frames_to_samples(frames_needed));
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}
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delay_output_buffer.clear();
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delay_output_buffer.push(delay_input_buffer.data(),
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frames_to_samples(frames_needed));
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delay_input_buffer.pop(nullptr, frames_to_samples(frames_needed));
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*input_frames_used = frames_needed;
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return delay_output_buffer.data();
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}
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/** Get a pointer to the first writable location in the input buffer>
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* @parameter frames_needed the number of frames the user needs to write into
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* the buffer.
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* @returns a pointer to a location in the input buffer where #frames_needed
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* can be writen. */
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T * input_buffer(uint32_t frames_needed)
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{
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leftover_samples = delay_input_buffer.length();
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delay_input_buffer.reserve(leftover_samples + frames_to_samples(frames_needed));
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return delay_input_buffer.data() + leftover_samples;
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}
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/** This method works with `input_buffer`, and allows to inform the processor
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how much frames have been written in the provided buffer. */
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void written(size_t frames_written)
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{
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delay_input_buffer.set_length(leftover_samples +
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frames_to_samples(frames_written));
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}
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/** Drains the delay line, emptying the buffer.
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* @parameter output_buffer the buffer in which the frames are written.
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* @parameter frames_needed the maximum number of frames to write.
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* @return the actual number of frames written. */
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size_t output(T * output_buffer, uint32_t frames_needed)
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{
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uint32_t in_len = samples_to_frames(delay_input_buffer.length());
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uint32_t out_len = frames_needed;
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uint32_t to_pop = std::min(in_len, out_len);
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delay_input_buffer.pop(output_buffer, frames_to_samples(to_pop));
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return to_pop;
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}
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/** Returns the number of frames one needs to input into the delay line to get
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* #frames_needed frames back.
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* @parameter frames_needed the number of frames one want to write into the
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* delay_line
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* @returns the number of frames one will get. */
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uint32_t input_needed_for_output(int32_t frames_needed) const
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{
|
|
assert(frames_needed >= 0); // Check overflow
|
|
return frames_needed;
|
|
}
|
|
/** Returns the number of frames produces for `input_frames` frames in input */
|
|
size_t output_for_input(uint32_t input_frames)
|
|
{
|
|
return input_frames;
|
|
}
|
|
/** The number of frames this delay line delays the stream by.
|
|
* @returns The number of frames of delay. */
|
|
size_t latency()
|
|
{
|
|
return length;
|
|
}
|
|
|
|
void drop_audio_if_needed()
|
|
{
|
|
size_t available = samples_to_frames(delay_input_buffer.length());
|
|
uint32_t to_keep = min_buffered_audio_frame(sample_rate);
|
|
if (available > to_keep) {
|
|
delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
|
|
}
|
|
}
|
|
private:
|
|
/** The length, in frames, of this delay line */
|
|
uint32_t length;
|
|
/** When `input_buffer` is called, this allows tracking the number of samples
|
|
that where in the buffer. */
|
|
uint32_t leftover_samples;
|
|
/** The input buffer, where the delay is applied. */
|
|
auto_array<T> delay_input_buffer;
|
|
/** The output buffer. This is only ever used if using the ::output with a
|
|
* single argument. */
|
|
auto_array<T> delay_output_buffer;
|
|
uint32_t sample_rate;
|
|
};
|
|
|
|
/** This sits behind the C API and is more typed. */
|
|
template<typename T>
|
|
cubeb_resampler *
|
|
cubeb_resampler_create_internal(cubeb_stream * stream,
|
|
cubeb_stream_params * input_params,
|
|
cubeb_stream_params * output_params,
|
|
unsigned int target_rate,
|
|
cubeb_data_callback callback,
|
|
void * user_ptr,
|
|
cubeb_resampler_quality quality)
|
|
{
|
|
std::unique_ptr<cubeb_resampler_speex_one_way<T>> input_resampler = nullptr;
|
|
std::unique_ptr<cubeb_resampler_speex_one_way<T>> output_resampler = nullptr;
|
|
std::unique_ptr<delay_line<T>> input_delay = nullptr;
|
|
std::unique_ptr<delay_line<T>> output_delay = nullptr;
|
|
|
|
assert((input_params || output_params) &&
|
|
"need at least one valid parameter pointer.");
|
|
|
|
/* All the streams we have have a sample rate that matches the target
|
|
sample rate, use a no-op resampler, that simply forwards the buffers to the
|
|
callback. */
|
|
if (((input_params && input_params->rate == target_rate) &&
|
|
(output_params && output_params->rate == target_rate)) ||
|
|
(input_params && !output_params && (input_params->rate == target_rate)) ||
|
|
(output_params && !input_params && (output_params->rate == target_rate))) {
|
|
LOG("Input and output sample-rate match, target rate of %dHz", target_rate);
|
|
return new passthrough_resampler<T>(stream, callback,
|
|
user_ptr,
|
|
input_params ? input_params->channels : 0,
|
|
target_rate);
|
|
}
|
|
|
|
/* Determine if we need to resampler one or both directions, and create the
|
|
resamplers. */
|
|
if (output_params && (output_params->rate != target_rate)) {
|
|
output_resampler.reset(
|
|
new cubeb_resampler_speex_one_way<T>(output_params->channels,
|
|
target_rate,
|
|
output_params->rate,
|
|
to_speex_quality(quality)));
|
|
if (!output_resampler) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
if (input_params && (input_params->rate != target_rate)) {
|
|
input_resampler.reset(
|
|
new cubeb_resampler_speex_one_way<T>(input_params->channels,
|
|
input_params->rate,
|
|
target_rate,
|
|
to_speex_quality(quality)));
|
|
if (!input_resampler) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* If we resample only one direction but we have a duplex stream, insert a
|
|
* delay line with a length equal to the resampler latency of the
|
|
* other direction so that the streams are synchronized. */
|
|
if (input_resampler && !output_resampler && input_params && output_params) {
|
|
output_delay.reset(new delay_line<T>(input_resampler->latency(),
|
|
output_params->channels,
|
|
output_params->rate));
|
|
if (!output_delay) {
|
|
return NULL;
|
|
}
|
|
} else if (output_resampler && !input_resampler && input_params && output_params) {
|
|
input_delay.reset(new delay_line<T>(output_resampler->latency(),
|
|
input_params->channels,
|
|
output_params->rate));
|
|
if (!input_delay) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
if (input_resampler && output_resampler) {
|
|
LOG("Resampling input (%d) and output (%d) to target rate of %dHz", input_params->rate, output_params->rate, target_rate);
|
|
return new cubeb_resampler_speex<T,
|
|
cubeb_resampler_speex_one_way<T>,
|
|
cubeb_resampler_speex_one_way<T>>
|
|
(input_resampler.release(),
|
|
output_resampler.release(),
|
|
stream, callback, user_ptr);
|
|
} else if (input_resampler) {
|
|
LOG("Resampling input (%d) to target and output rate of %dHz", input_params->rate, target_rate);
|
|
return new cubeb_resampler_speex<T,
|
|
cubeb_resampler_speex_one_way<T>,
|
|
delay_line<T>>
|
|
(input_resampler.release(),
|
|
output_delay.release(),
|
|
stream, callback, user_ptr);
|
|
} else {
|
|
LOG("Resampling output (%dHz) to target and input rate of %dHz", output_params->rate, target_rate);
|
|
return new cubeb_resampler_speex<T,
|
|
delay_line<T>,
|
|
cubeb_resampler_speex_one_way<T>>
|
|
(input_delay.release(),
|
|
output_resampler.release(),
|
|
stream, callback, user_ptr);
|
|
}
|
|
}
|
|
|
|
#endif /* CUBEB_RESAMPLER_INTERNAL */
|