Move functionality out from AudioFrame and into AudioFrameOperations.

webrtc/modules/utility/source/audio_frame_operations.cc

-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-namespace  {
-
-// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
-const size_t kMuteFadeFrames = 128;
-const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
-
-}  // namespace {
-
-void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
-                                        size_t samples_per_channel,
-                                        int16_t* dst_audio) {
-  for (size_t i = 0; i < samples_per_channel; i++) {
-    dst_audio[2 * i] = src_audio[i];
-    dst_audio[2 * i + 1] = src_audio[i];
-  }
-}
-
-int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
-  if (frame->num_channels_ != 1) {
-    return -1;
-  }
-  if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) {
-    // Not enough memory to expand from mono to stereo.
-    return -1;
-  }
-
-  int16_t data_copy[AudioFrame::kMaxDataSizeSamples];
-  memcpy(data_copy, frame->data_,
-         sizeof(int16_t) * frame->samples_per_channel_);
-  MonoToStereo(data_copy, frame->samples_per_channel_, frame->data_);
-  frame->num_channels_ = 2;
-
-  return 0;
-}
-
-void AudioFrameOperations::StereoToMono(const int16_t* src_audio,
-                                        size_t samples_per_channel,
-                                        int16_t* dst_audio) {
-  for (size_t i = 0; i < samples_per_channel; i++) {
-    dst_audio[i] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1;
-  }
-}
-
-int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
-  if (frame->num_channels_ != 2) {
-    return -1;
-  }
-
-  StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_);
-  frame->num_channels_ = 1;
-
-  return 0;
-}
-
-void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
-  if (frame->num_channels_ != 2) return;
-
-  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
-    int16_t temp_data = frame->data_[i];
-    frame->data_[i] = frame->data_[i + 1];
-    frame->data_[i + 1] = temp_data;
-  }
-}
-
-void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
-                                bool current_frame_muted) {
-  RTC_DCHECK(frame);
-  if (!previous_frame_muted && !current_frame_muted) {
-    // Not muted, don't touch.
-  } else if (previous_frame_muted && current_frame_muted) {
-    // Frame fully muted.
-    size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
-    RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
-    memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples);
-  } else {
-    // Limit number of samples to fade, if frame isn't long enough.
-    size_t count = kMuteFadeFrames;
-    float inc = kMuteFadeInc;
-    if (frame->samples_per_channel_ < kMuteFadeFrames) {
-      count = frame->samples_per_channel_;
-      if (count > 0) {
-        inc = 1.0f / count;
-      }
-    }
-
-    size_t start = 0;
-    size_t end = count;
-    float start_g = 0.0f;
-    if (current_frame_muted) {
-      // Fade out the last |count| samples of frame.
-      RTC_DCHECK(!previous_frame_muted);
-      start = frame->samples_per_channel_ - count;
-      end = frame->samples_per_channel_;
-      start_g = 1.0f;
-      inc = -inc;
-    } else {
-      // Fade in the first |count| samples of frame.
-      RTC_DCHECK(previous_frame_muted);
-    }
-
-    // Perform fade.
-    size_t channels = frame->num_channels_;
-    for (size_t j = 0; j < channels; ++j) {
-      float g = start_g;
-      for (size_t i = start * channels; i < end * channels; i += channels) {
-        g += inc;
-        frame->data_[i + j] *= g;
-      }
-    }
-  }
-}
-
-int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
-  if (frame.num_channels_ != 2) {
-    return -1;
-  }
-
-  for (size_t i = 0; i < frame.samples_per_channel_; i++) {
-    frame.data_[2 * i] =
-        static_cast<int16_t>(left * frame.data_[2 * i]);
-    frame.data_[2 * i + 1] =
-        static_cast<int16_t>(right * frame.data_[2 * i + 1]);
-  }
-  return 0;
-}
-
-int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
-  int32_t temp_data = 0;
-
-  // Ensure that the output result is saturated [-32768, +32767].
-  for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
-       i++) {
-    temp_data = static_cast<int32_t>(scale * frame.data_[i]);
-    if (temp_data < -32768) {
-      frame.data_[i] = -32768;
-    } else if (temp_data > 32767) {
-      frame.data_[i] = 32767;
-    } else {
-      frame.data_[i] = static_cast<int16_t>(temp_data);
-    }
-  }
-  return 0;
-}
-
-}  // namespace webrtc