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andrew@webrtc.org041035b2015-01-26 21:23:53 +00001/*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
kwibergc2b785d2016-02-24 05:22:32 -080011#include <memory>
12
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020013#include "common_audio/audio_ring_buffer.h"
andrew@webrtc.org041035b2015-01-26 21:23:53 +000014
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "common_audio/channel_buffer.h"
16#include "test/gtest.h"
andrew@webrtc.org041035b2015-01-26 21:23:53 +000017
18namespace webrtc {
19
20class AudioRingBufferTest :
21 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
22};
23
24void ReadAndWriteTest(const ChannelBuffer<float>& input,
25 size_t num_write_chunk_frames,
26 size_t num_read_chunk_frames,
27 size_t buffer_frames,
28 ChannelBuffer<float>* output) {
29 const size_t num_channels = input.num_channels();
aluebs@webrtc.orgd35a5c32015-02-10 22:52:15 +000030 const size_t total_frames = input.num_frames();
andrew@webrtc.org041035b2015-01-26 21:23:53 +000031 AudioRingBuffer buf(num_channels, buffer_frames);
kwibergc2b785d2016-02-24 05:22:32 -080032 std::unique_ptr<float* []> slice(new float*[num_channels]);
andrew@webrtc.org041035b2015-01-26 21:23:53 +000033
34 size_t input_pos = 0;
35 size_t output_pos = 0;
36 while (input_pos + buf.WriteFramesAvailable() < total_frames) {
37 // Write until the buffer is as full as possible.
38 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
Peter Kastingdce40cf2015-08-24 14:52:23 -070039 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
40 num_write_chunk_frames);
andrew@webrtc.org041035b2015-01-26 21:23:53 +000041 input_pos += num_write_chunk_frames;
42 }
43 // Read until the buffer is as empty as possible.
44 while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
45 EXPECT_LT(output_pos, total_frames);
Peter Kastingdce40cf2015-08-24 14:52:23 -070046 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
47 num_read_chunk_frames);
andrew@webrtc.org041035b2015-01-26 21:23:53 +000048 output_pos += num_read_chunk_frames;
49 }
50 }
51
52 // Write and read the last bit.
Peter Kasting728d9032015-06-11 14:31:38 -070053 if (input_pos < total_frames) {
Peter Kastingdce40cf2015-08-24 14:52:23 -070054 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
55 total_frames - input_pos);
Peter Kasting728d9032015-06-11 14:31:38 -070056 }
57 if (buf.ReadFramesAvailable()) {
Peter Kastingdce40cf2015-08-24 14:52:23 -070058 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
59 buf.ReadFramesAvailable());
Peter Kasting728d9032015-06-11 14:31:38 -070060 }
andrew@webrtc.org041035b2015-01-26 21:23:53 +000061 EXPECT_EQ(0u, buf.ReadFramesAvailable());
62}
63
64TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
65 const size_t kFrames = 5000;
66 const size_t num_channels = ::testing::get<3>(GetParam());
67
68 // Initialize the input data to an increasing sequence.
69 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
70 for (size_t i = 0; i < num_channels; ++i)
71 for (size_t j = 0; j < kFrames; ++j)
andrew@webrtc.org922cfcd2015-01-27 21:59:33 +000072 input.channels()[i][j] = (i + 1) * (j + 1);
andrew@webrtc.org041035b2015-01-26 21:23:53 +000073
74 ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
75 ReadAndWriteTest(input,
76 ::testing::get<0>(GetParam()),
77 ::testing::get<1>(GetParam()),
78 ::testing::get<2>(GetParam()),
79 &output);
80
81 // Verify the read data matches the input.
82 for (size_t i = 0; i < num_channels; ++i)
83 for (size_t j = 0; j < kFrames; ++j)
84 EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
85}
86
87INSTANTIATE_TEST_CASE_P(
88 AudioRingBufferTest, AudioRingBufferTest,
89 ::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames
90 ::testing::Values(1, 10, 17), // num_read_chunk_frames
91 ::testing::Values(100, 256), // buffer_frames
92 ::testing::Values(1, 4))); // num_channels
93
94TEST_F(AudioRingBufferTest, MoveReadPosition) {
95 const size_t kNumChannels = 1;
96 const float kInputArray[] = {1, 2, 3, 4};
97 const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
aluebs@webrtc.orgd35a5c32015-02-10 22:52:15 +000098 ChannelBuffer<float> input(kNumFrames, kNumChannels);
99 input.SetDataForTesting(kInputArray, kNumFrames);
andrew@webrtc.org041035b2015-01-26 21:23:53 +0000100 AudioRingBuffer buf(kNumChannels, kNumFrames);
101 buf.Write(input.channels(), kNumChannels, kNumFrames);
102
andrewd40af692015-07-28 00:52:59 -0700103 buf.MoveReadPositionForward(3);
andrew@webrtc.org041035b2015-01-26 21:23:53 +0000104 ChannelBuffer<float> output(1, kNumChannels);
105 buf.Read(output.channels(), kNumChannels, 1);
aluebs@webrtc.orgd35a5c32015-02-10 22:52:15 +0000106 EXPECT_EQ(4, output.channels()[0][0]);
andrewd40af692015-07-28 00:52:59 -0700107 buf.MoveReadPositionBackward(3);
andrew@webrtc.org041035b2015-01-26 21:23:53 +0000108 buf.Read(output.channels(), kNumChannels, 1);
aluebs@webrtc.orgd35a5c32015-02-10 22:52:15 +0000109 EXPECT_EQ(2, output.channels()[0][0]);
andrew@webrtc.org041035b2015-01-26 21:23:53 +0000110}
111
112} // namespace webrtc