Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #include "webrtc/audio/utility/audio_frame_operations.h"
      12             : #include "webrtc/base/checks.h"
      13             : #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
      14             : #include "webrtc/modules/include/module_common_types.h"
      15             : 
      16             : namespace webrtc {
      17             : 
      18           0 : uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
      19           0 :   uint32_t energy = 0;
      20           0 :   for (size_t position = 0; position < audio_frame.samples_per_channel_;
      21             :        position++) {
      22             :     // TODO(aleloi): This can overflow. Convert to floats.
      23           0 :     energy += audio_frame.data_[position] * audio_frame.data_[position];
      24             :   }
      25           0 :   return energy;
      26             : }
      27             : 
      28           0 : void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
      29           0 :   RTC_DCHECK(audio_frame);
      30           0 :   RTC_DCHECK_GE(start_gain, 0.0f);
      31           0 :   RTC_DCHECK_GE(target_gain, 0.0f);
      32           0 :   if (start_gain == target_gain) {
      33           0 :     return;
      34             :   }
      35             : 
      36           0 :   size_t samples = audio_frame->samples_per_channel_;
      37           0 :   RTC_DCHECK_LT(0, samples);
      38           0 :   float increment = (target_gain - start_gain) / samples;
      39           0 :   float gain = start_gain;
      40           0 :   for (size_t i = 0; i < samples; ++i) {
      41             :     // If the audio is interleaved of several channels, we want to
      42             :     // apply the same gain change to the ith sample of every channel.
      43           0 :     for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
      44           0 :       audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
      45             :     }
      46           0 :     gain += increment;
      47             :   }
      48             : }
      49             : 
      50           0 : void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
      51           0 :   RTC_DCHECK_GE(target_number_of_channels, 1);
      52           0 :   RTC_DCHECK_LE(target_number_of_channels, 2);
      53           0 :   if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
      54           0 :     AudioFrameOperations::MonoToStereo(frame);
      55           0 :   } else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
      56           0 :     AudioFrameOperations::StereoToMono(frame);
      57             :   }
      58           0 : }
      59             : }  // namespace webrtc