24899e58ec
The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping). To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner. The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other. This also fixes a few minor GN issues so that warnings do not have to be suppressed. BUG=webrtc:7167 Review-Url: https://codereview.webrtc.org/2692333002 Cr-Commit-Position: refs/heads/master@{#16742}
133 lines
4.6 KiB
C++
133 lines
4.6 KiB
C++
/*
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* Copyright (c) 2017 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/modules/audio_mixer/frame_combiner.h"
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#include <numeric>
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#include <sstream>
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#include <string>
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#include "webrtc/base/checks.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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namespace {
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std::string ProduceDebugText(int sample_rate_hz,
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int number_of_channels,
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int number_of_sources) {
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std::ostringstream ss;
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ss << "Sample rate: " << sample_rate_hz << " ";
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ss << "Number of channels: " << number_of_channels << " ";
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ss << "Number of sources: " << number_of_sources;
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return ss.str();
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}
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AudioFrame frame1;
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AudioFrame frame2;
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AudioFrame audio_frame_for_mixing;
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void SetUpFrames(int sample_rate_hz, int number_of_channels) {
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for (auto* frame : {&frame1, &frame2}) {
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frame->UpdateFrame(-1, 0, nullptr,
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rtc::CheckedDivExact(sample_rate_hz, 100),
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sample_rate_hz, AudioFrame::kNormalSpeech,
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AudioFrame::kVadActive, number_of_channels);
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}
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}
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} // namespace
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TEST(FrameCombiner, BasicApiCallsLimiter) {
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FrameCombiner combiner(true);
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for (const int rate : {8000, 16000, 32000, 48000}) {
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for (const int number_of_channels : {1, 2}) {
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const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
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SetUpFrames(rate, number_of_channels);
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for (const int number_of_frames : {0, 1, 2}) {
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SCOPED_TRACE(
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ProduceDebugText(rate, number_of_channels, number_of_frames));
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const std::vector<AudioFrame*> frames_to_combine(
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all_frames.begin(), all_frames.begin() + number_of_frames);
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combiner.Combine(frames_to_combine, number_of_channels, rate,
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&audio_frame_for_mixing);
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}
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}
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}
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}
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// No APM limiter means no AudioProcessing::NativeRate restriction
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// on rate. The rate has to be divisible by 100 since we use
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// 10 ms frames, though.
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TEST(FrameCombiner, BasicApiCallsNoLimiter) {
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FrameCombiner combiner(false);
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for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
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for (const int number_of_channels : {1, 2}) {
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const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
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SetUpFrames(rate, number_of_channels);
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for (const int number_of_frames : {0, 1, 2}) {
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SCOPED_TRACE(
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ProduceDebugText(rate, number_of_channels, number_of_frames));
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const std::vector<AudioFrame*> frames_to_combine(
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all_frames.begin(), all_frames.begin() + number_of_frames);
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combiner.Combine(frames_to_combine, number_of_channels, rate,
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&audio_frame_for_mixing);
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}
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}
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}
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}
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TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
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FrameCombiner combiner(false);
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for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
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for (const int number_of_channels : {1, 2}) {
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SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0));
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const std::vector<AudioFrame*> frames_to_combine;
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combiner.Combine(frames_to_combine, number_of_channels, rate,
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&audio_frame_for_mixing);
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const std::vector<int16_t> mixed_data(
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audio_frame_for_mixing.data_,
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audio_frame_for_mixing.data_ + number_of_channels * rate / 100);
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const std::vector<int16_t> expected(number_of_channels * rate / 100, 0);
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EXPECT_EQ(mixed_data, expected);
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}
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}
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}
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TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
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FrameCombiner combiner(false);
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for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
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for (const int number_of_channels : {1, 2}) {
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SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
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SetUpFrames(rate, number_of_channels);
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std::iota(frame1.data_, frame1.data_ + number_of_channels * rate / 100,
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0);
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const std::vector<AudioFrame*> frames_to_combine = {&frame1};
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combiner.Combine(frames_to_combine, number_of_channels, rate,
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&audio_frame_for_mixing);
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const std::vector<int16_t> mixed_data(
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audio_frame_for_mixing.data_,
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audio_frame_for_mixing.data_ + number_of_channels * rate / 100);
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std::vector<int16_t> expected(number_of_channels * rate / 100);
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std::iota(expected.begin(), expected.end(), 0);
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EXPECT_EQ(mixed_data, expected);
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}
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}
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}
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} // namespace webrtc
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