36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
66 lines
2.3 KiB
C++
66 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/audio/utility/audio_frame_operations.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
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if (audio_frame.muted()) {
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return 0;
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}
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uint32_t energy = 0;
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const int16_t* frame_data = audio_frame.data();
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for (size_t position = 0; position < audio_frame.samples_per_channel_;
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position++) {
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// TODO(aleloi): This can overflow. Convert to floats.
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energy += frame_data[position] * frame_data[position];
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}
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return energy;
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}
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void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
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RTC_DCHECK(audio_frame);
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RTC_DCHECK_GE(start_gain, 0.0f);
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RTC_DCHECK_GE(target_gain, 0.0f);
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if (start_gain == target_gain || audio_frame->muted()) {
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return;
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}
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size_t samples = audio_frame->samples_per_channel_;
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RTC_DCHECK_LT(0, samples);
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float increment = (target_gain - start_gain) / samples;
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float gain = start_gain;
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int16_t* frame_data = audio_frame->mutable_data();
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for (size_t i = 0; i < samples; ++i) {
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// If the audio is interleaved of several channels, we want to
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// apply the same gain change to the ith sample of every channel.
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for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
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frame_data[audio_frame->num_channels_ * i + ch] *= gain;
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}
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gain += increment;
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}
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}
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void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
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RTC_DCHECK_GE(target_number_of_channels, 1);
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RTC_DCHECK_LE(target_number_of_channels, 2);
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if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
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AudioFrameOperations::MonoToStereo(frame);
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} else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
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AudioFrameOperations::StereoToMono(frame);
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}
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}
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} // namespace webrtc
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