third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.cc

   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 "modules/audio_mixer/audio_frame_manipulator.h"
  12
  13 #include "audio/utility/audio_frame_operations.h"
  14 #include "audio/utility/channel_mixer.h"
  15 #include "rtc_base/checks.h"
  16
  17 namespace webrtc {
  18
  19 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
  20   if (audio_frame.muted()) {
  21     return 0;
  22   }
  23
  24   uint32_t energy = 0;
  25   const int16_t* frame_data = audio_frame.data();
  26   for (size_t position = 0;
  27        position < audio_frame.samples_per_channel_ * audio_frame.num_channels_;
  28        position++) {
  29     // TODO(aleloi): This can overflow. Convert to floats.
  30     energy += frame_data[position] * frame_data[position];
  31   }
  32   return energy;
  33 }
  34
  35 void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
  36   RTC_DCHECK(audio_frame);
  37   RTC_DCHECK_GE(start_gain, 0.0f);
  38   RTC_DCHECK_GE(target_gain, 0.0f);
  39   if (start_gain == target_gain || audio_frame->muted()) {
  40     return;
  41   }
  42
  43   size_t samples = audio_frame->samples_per_channel_;
  44   RTC_DCHECK_LT(0, samples);
  45   float increment = (target_gain - start_gain) / samples;
  46   float gain = start_gain;
  47   int16_t* frame_data = audio_frame->mutable_data();
  48   for (size_t i = 0; i < samples; ++i) {
  49     // If the audio is interleaved of several channels, we want to
  50     // apply the same gain change to the ith sample of every channel.
  51     for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
  52       frame_data[audio_frame->num_channels_ * i + ch] *= gain;
  53     }
  54     gain += increment;
  55   }
  56 }
  57
  58 void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
  59   RTC_DCHECK_GE(target_number_of_channels, 1);
  60   // TODO(bugs.webrtc.org/10783): take channel layout into account as well.
  61   if (frame->num_channels() == target_number_of_channels) {
  62     return;
  63   }
  64
  65   // Use legacy components for the most simple cases (mono <-> stereo) to ensure
  66   // that native WebRTC clients are not affected when support for multi-channel
  67   // audio is added to Chrome.
  68   // TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well.
  69   if (target_number_of_channels < 3 && frame->num_channels() < 3) {
  70     if (frame->num_channels() > target_number_of_channels) {
  71       AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
  72     } else {
  73       AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
  74     }
  75   } else {
  76     // Use generic channel mixer when the number of channels for input our
  77     // output is larger than two. E.g. stereo -> 5.1 channel up-mixing.
  78     // TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used
  79     // instead of guessing based on number of channels.
  80     const ChannelLayout output_layout(
  81         GuessChannelLayout(target_number_of_channels));
  82     ChannelMixer mixer(GuessChannelLayout(frame->num_channels()),
  83                        output_layout);
  84     mixer.Transform(frame);
  85     RTC_DCHECK_EQ(frame->channel_layout(), output_layout);
  86   }
  87   RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
  88       << "Wrong number of channels, " << frame->num_channels() << " vs "
  89       << target_number_of_channels;
  90 }
  91
  92 }  // namespace webrtc