dep: Add cubeb

dep/cubeb/src/cubeb_resampler_internal.h

@@ -0,0 +1,602 @@
+/*
+ * 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 <cmath>
+#include <cassert>
+#include <algorithm>
+#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/cubeb.h"
+#include "cubeb_utils.h"
+#include "cubeb-speex-resampler.h"
+#include "cubeb_resampler.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) {
+      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");
+  }
+
+  /** Destructor, deallocate the resampler */
+  virtual ~cubeb_resampler_speex_one_way()
+  {
+    speex_resampler_destroy(speex_resampler);
+  }
+
+  /** Sometimes, it is necessary to add latency on one way of a two-way
+   * resampler so that the stream are synchronized. This must be called only on
+   * a fresh resampler, otherwise, silent samples will be inserted in the
+   * stream.
+   * @param frames the number of frames of latency to add. */
+  void add_latency(size_t frames)
+  {
+    additional_latency += frames;
+    resampling_in_buffer.push_silence(frames_to_samples(frames));
+  }
+
+  /* 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);
+
+    assert(out_len == output_frame_count);
+
+    /* 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(uint32_t output_frame_count) const
+  {
+    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) {
+      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);
+  }
+  /* Add some latency to the delay line.
+   * @param frames the number of frames of latency to add. */
+  void add_latency(size_t frames)
+  {
+    length += frames;
+    delay_input_buffer.push_silence(frames_to_samples(frames));
+  }
+  /** 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. */
+  size_t input_needed_for_output(uint32_t frames_needed) const
+  {
+    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))) {
+    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) {
+    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) {
+    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 {
+    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 */