/* * 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 "modules/audio_mixer/audio_frame_manipulator.h" #include "audio/utility/audio_frame_operations.h" #include "audio/utility/channel_mixer.h" #include "rtc_base/checks.h" namespace webrtc { uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) { if (audio_frame.muted()) { return 0; } uint32_t energy = 0; const int16_t* frame_data = audio_frame.data(); for (size_t position = 0; position < audio_frame.samples_per_channel_ * audio_frame.num_channels_; position++) { // TODO(aleloi): This can overflow. Convert to floats. energy += frame_data[position] * frame_data[position]; } return energy; } void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { RTC_DCHECK(audio_frame); RTC_DCHECK_GE(start_gain, 0.0f); RTC_DCHECK_GE(target_gain, 0.0f); if (start_gain == target_gain || audio_frame->muted()) { return; } size_t samples = audio_frame->samples_per_channel_; RTC_DCHECK_LT(0, samples); float increment = (target_gain - start_gain) / samples; float gain = start_gain; int16_t* frame_data = audio_frame->mutable_data(); for (size_t i = 0; i < samples; ++i) { // If the audio is interleaved of several channels, we want to // apply the same gain change to the ith sample of every channel. for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) { frame_data[audio_frame->num_channels_ * i + ch] *= gain; } gain += increment; } } void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) { RTC_DCHECK_GE(target_number_of_channels, 1); // TODO(bugs.webrtc.org/10783): take channel layout into account as well. if (frame->num_channels() == target_number_of_channels) { return; } // Use legacy components for the most simple cases (mono <-> stereo) to ensure // that native WebRTC clients are not affected when support for multi-channel // audio is added to Chrome. // TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well. if (target_number_of_channels < 3 && frame->num_channels() < 3) { if (frame->num_channels() > target_number_of_channels) { AudioFrameOperations::DownmixChannels(target_number_of_channels, frame); } else { AudioFrameOperations::UpmixChannels(target_number_of_channels, frame); } } else { // Use generic channel mixer when the number of channels for input our // output is larger than two. E.g. stereo -> 5.1 channel up-mixing. // TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used // instead of guessing based on number of channels. const ChannelLayout output_layout( GuessChannelLayout(target_number_of_channels)); ChannelMixer mixer(GuessChannelLayout(frame->num_channels()), output_layout); mixer.Transform(frame); RTC_DCHECK_EQ(frame->channel_layout(), output_layout); } RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels) << "Wrong number of channels, " << frame->num_channels() << " vs " << target_number_of_channels; } } // namespace webrtc