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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
| 11 #include <memory> |
| 12 |
11 #include "webrtc/common_audio/audio_ring_buffer.h" | 13 #include "webrtc/common_audio/audio_ring_buffer.h" |
12 | 14 |
13 #include "testing/gtest/include/gtest/gtest.h" | 15 #include "testing/gtest/include/gtest/gtest.h" |
14 #include "webrtc/common_audio/channel_buffer.h" | 16 #include "webrtc/common_audio/channel_buffer.h" |
15 | 17 |
16 namespace webrtc { | 18 namespace webrtc { |
17 | 19 |
18 class AudioRingBufferTest : | 20 class AudioRingBufferTest : |
19 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { | 21 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { |
20 }; | 22 }; |
21 | 23 |
22 void ReadAndWriteTest(const ChannelBuffer<float>& input, | 24 void ReadAndWriteTest(const ChannelBuffer<float>& input, |
23 size_t num_write_chunk_frames, | 25 size_t num_write_chunk_frames, |
24 size_t num_read_chunk_frames, | 26 size_t num_read_chunk_frames, |
25 size_t buffer_frames, | 27 size_t buffer_frames, |
26 ChannelBuffer<float>* output) { | 28 ChannelBuffer<float>* output) { |
27 const size_t num_channels = input.num_channels(); | 29 const size_t num_channels = input.num_channels(); |
28 const size_t total_frames = input.num_frames(); | 30 const size_t total_frames = input.num_frames(); |
29 AudioRingBuffer buf(num_channels, buffer_frames); | 31 AudioRingBuffer buf(num_channels, buffer_frames); |
30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]); | 32 std::unique_ptr<float* []> slice(new float*[num_channels]); |
31 | 33 |
32 size_t input_pos = 0; | 34 size_t input_pos = 0; |
33 size_t output_pos = 0; | 35 size_t output_pos = 0; |
34 while (input_pos + buf.WriteFramesAvailable() < total_frames) { | 36 while (input_pos + buf.WriteFramesAvailable() < total_frames) { |
35 // Write until the buffer is as full as possible. | 37 // Write until the buffer is as full as possible. |
36 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { | 38 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { |
37 buf.Write(input.Slice(slice.get(), input_pos), num_channels, | 39 buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
38 num_write_chunk_frames); | 40 num_write_chunk_frames); |
39 input_pos += num_write_chunk_frames; | 41 input_pos += num_write_chunk_frames; |
40 } | 42 } |
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101 buf.MoveReadPositionForward(3); | 103 buf.MoveReadPositionForward(3); |
102 ChannelBuffer<float> output(1, kNumChannels); | 104 ChannelBuffer<float> output(1, kNumChannels); |
103 buf.Read(output.channels(), kNumChannels, 1); | 105 buf.Read(output.channels(), kNumChannels, 1); |
104 EXPECT_EQ(4, output.channels()[0][0]); | 106 EXPECT_EQ(4, output.channels()[0][0]); |
105 buf.MoveReadPositionBackward(3); | 107 buf.MoveReadPositionBackward(3); |
106 buf.Read(output.channels(), kNumChannels, 1); | 108 buf.Read(output.channels(), kNumChannels, 1); |
107 EXPECT_EQ(2, output.channels()[0][0]); | 109 EXPECT_EQ(2, output.channels()[0][0]); |
108 } | 110 } |
109 | 111 |
110 } // namespace webrtc | 112 } // namespace webrtc |
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