a33f05e8d7
Changes differing from https://webrtc-codereview.appspot.com/37859004: * I put the include_tests==1 stuff of audio_coding.gypi in its own audio_coding_tests.gypi file, including the Android and isolate targets which were incorrectly located in the previous CL * I moved the bwe utilities in remote_bitrate_estimator.gypi into include_tests==1 since they depend on test.gyp after I cleaned up the duplicated inclusion of rtp_file_reader.cc R=stefan@webrtc.org TBR=tina.legrand@webrtc.org TESTED=Passing gyp and compile using: webrtc/build/gyp_webrtc -Dinclude_tests=1 webrtc/build/gyp_webrtc -Dinclude_tests=0 I also setup a Chromium checkout with my checkout mounted in third_party/webrtc and ran build/gyp_chromium successfully. BUG=4185 Review URL: https://webrtc-codereview.appspot.com/33159004 Cr-Commit-Position: refs/heads/master@{#8205} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
64 lines
2.1 KiB
C++
64 lines
2.1 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/tools/agc/test_utils.h"
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#include <cmath>
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#include <algorithm>
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#include "webrtc/modules/interface/module_common_types.h"
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namespace webrtc {
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float MicLevel2Gain(int gain_range_db, int level) {
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return (level - 127.0f) / 128.0f * gain_range_db / 2;
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}
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float Db2Linear(float db) {
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return powf(10.0f, db / 20.0f);
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}
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void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
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const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
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// Smooth the transition between gain levels across the frame.
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float smoothed_gain = last_gain;
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float gain_step = (gain - last_gain) / (frame_length - 1);
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for (int i = 0; i < frame_length; ++i) {
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smoothed_gain += gain_step;
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float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
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sample = std::max(std::min(32767.0f, sample), -32768.0f);
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frame->data_[i] = static_cast<int16_t>(sample);
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}
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}
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void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
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ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
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}
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void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
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AudioFrame* frame) {
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assert(mic_level >= 0 && mic_level <= 255);
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assert(last_mic_level >= 0 && last_mic_level <= 255);
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ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
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MicLevel2Gain(gain_range_db, last_mic_level),
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frame);
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}
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void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
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AudioFrame* frame) {
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assert(mic_level >= 0 && mic_level <= 255);
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assert(last_mic_level >= 0 && last_mic_level <= 255);
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ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
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}
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} // namespace webrtc
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