Files
platform-external-webrtc/webrtc/common_audio/audio_ring_buffer_unittest.cc
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

111 lines
4.2 KiB
C++

/*
* Copyright (c) 2015 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 "webrtc/common_audio/audio_ring_buffer.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/channel_buffer.h"
namespace webrtc {
class AudioRingBufferTest :
public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
};
void ReadAndWriteTest(const ChannelBuffer<float>& input,
size_t num_write_chunk_frames,
size_t num_read_chunk_frames,
size_t buffer_frames,
ChannelBuffer<float>* output) {
const size_t num_channels = input.num_channels();
const size_t total_frames = input.num_frames();
AudioRingBuffer buf(num_channels, buffer_frames);
rtc::scoped_ptr<float* []> slice(new float* [num_channels]);
size_t input_pos = 0;
size_t output_pos = 0;
while (input_pos + buf.WriteFramesAvailable() < total_frames) {
// Write until the buffer is as full as possible.
while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
buf.Write(input.Slice(slice.get(), input_pos), num_channels,
num_write_chunk_frames);
input_pos += num_write_chunk_frames;
}
// Read until the buffer is as empty as possible.
while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
EXPECT_LT(output_pos, total_frames);
buf.Read(output->Slice(slice.get(), output_pos), num_channels,
num_read_chunk_frames);
output_pos += num_read_chunk_frames;
}
}
// Write and read the last bit.
if (input_pos < total_frames) {
buf.Write(input.Slice(slice.get(), input_pos), num_channels,
total_frames - input_pos);
}
if (buf.ReadFramesAvailable()) {
buf.Read(output->Slice(slice.get(), output_pos), num_channels,
buf.ReadFramesAvailable());
}
EXPECT_EQ(0u, buf.ReadFramesAvailable());
}
TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
const size_t kFrames = 5000;
const size_t num_channels = ::testing::get<3>(GetParam());
// Initialize the input data to an increasing sequence.
ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
for (size_t i = 0; i < num_channels; ++i)
for (size_t j = 0; j < kFrames; ++j)
input.channels()[i][j] = (i + 1) * (j + 1);
ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
ReadAndWriteTest(input,
::testing::get<0>(GetParam()),
::testing::get<1>(GetParam()),
::testing::get<2>(GetParam()),
&output);
// Verify the read data matches the input.
for (size_t i = 0; i < num_channels; ++i)
for (size_t j = 0; j < kFrames; ++j)
EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
}
INSTANTIATE_TEST_CASE_P(
AudioRingBufferTest, AudioRingBufferTest,
::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames
::testing::Values(1, 10, 17), // num_read_chunk_frames
::testing::Values(100, 256), // buffer_frames
::testing::Values(1, 4))); // num_channels
TEST_F(AudioRingBufferTest, MoveReadPosition) {
const size_t kNumChannels = 1;
const float kInputArray[] = {1, 2, 3, 4};
const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
ChannelBuffer<float> input(kNumFrames, kNumChannels);
input.SetDataForTesting(kInputArray, kNumFrames);
AudioRingBuffer buf(kNumChannels, kNumFrames);
buf.Write(input.channels(), kNumChannels, kNumFrames);
buf.MoveReadPositionForward(3);
ChannelBuffer<float> output(1, kNumChannels);
buf.Read(output.channels(), kNumChannels, 1);
EXPECT_EQ(4, output.channels()[0][0]);
buf.MoveReadPositionBackward(3);
buf.Read(output.channels(), kNumChannels, 1);
EXPECT_EQ(2, output.channels()[0][0]);
}
} // namespace webrtc