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