OLD | NEW |
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 | |
13 #include "webrtc/common_audio/audio_ring_buffer.h" | 11 #include "webrtc/common_audio/audio_ring_buffer.h" |
14 | 12 |
15 #include "testing/gtest/include/gtest/gtest.h" | 13 #include "testing/gtest/include/gtest/gtest.h" |
16 #include "webrtc/common_audio/channel_buffer.h" | 14 #include "webrtc/common_audio/channel_buffer.h" |
17 | 15 |
18 namespace webrtc { | 16 namespace webrtc { |
19 | 17 |
20 class AudioRingBufferTest : | 18 class AudioRingBufferTest : |
21 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { | 19 public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { |
22 }; | 20 }; |
23 | 21 |
24 void ReadAndWriteTest(const ChannelBuffer<float>& input, | 22 void ReadAndWriteTest(const ChannelBuffer<float>& input, |
25 size_t num_write_chunk_frames, | 23 size_t num_write_chunk_frames, |
26 size_t num_read_chunk_frames, | 24 size_t num_read_chunk_frames, |
27 size_t buffer_frames, | 25 size_t buffer_frames, |
28 ChannelBuffer<float>* output) { | 26 ChannelBuffer<float>* output) { |
29 const size_t num_channels = input.num_channels(); | 27 const size_t num_channels = input.num_channels(); |
30 const size_t total_frames = input.num_frames(); | 28 const size_t total_frames = input.num_frames(); |
31 AudioRingBuffer buf(num_channels, buffer_frames); | 29 AudioRingBuffer buf(num_channels, buffer_frames); |
32 std::unique_ptr<float* []> slice(new float*[num_channels]); | 30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]); |
33 | 31 |
34 size_t input_pos = 0; | 32 size_t input_pos = 0; |
35 size_t output_pos = 0; | 33 size_t output_pos = 0; |
36 while (input_pos + buf.WriteFramesAvailable() < total_frames) { | 34 while (input_pos + buf.WriteFramesAvailable() < total_frames) { |
37 // Write until the buffer is as full as possible. | 35 // Write until the buffer is as full as possible. |
38 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { | 36 while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { |
39 buf.Write(input.Slice(slice.get(), input_pos), num_channels, | 37 buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
40 num_write_chunk_frames); | 38 num_write_chunk_frames); |
41 input_pos += num_write_chunk_frames; | 39 input_pos += num_write_chunk_frames; |
42 } | 40 } |
(...skipping 60 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
103 buf.MoveReadPositionForward(3); | 101 buf.MoveReadPositionForward(3); |
104 ChannelBuffer<float> output(1, kNumChannels); | 102 ChannelBuffer<float> output(1, kNumChannels); |
105 buf.Read(output.channels(), kNumChannels, 1); | 103 buf.Read(output.channels(), kNumChannels, 1); |
106 EXPECT_EQ(4, output.channels()[0][0]); | 104 EXPECT_EQ(4, output.channels()[0][0]); |
107 buf.MoveReadPositionBackward(3); | 105 buf.MoveReadPositionBackward(3); |
108 buf.Read(output.channels(), kNumChannels, 1); | 106 buf.Read(output.channels(), kNumChannels, 1); |
109 EXPECT_EQ(2, output.channels()[0][0]); | 107 EXPECT_EQ(2, output.channels()[0][0]); |
110 } | 108 } |
111 | 109 |
112 } // namespace webrtc | 110 } // namespace webrtc |
OLD | NEW |