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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 | |
| 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 |
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