d35a5c3506
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer. This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample]. All the files using the ChannelBuffer needed to be re-factored. Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test. R=andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36999004 Cr-Commit-Position: refs/heads/master@{#8318} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
109 lines
4.2 KiB
C++
109 lines
4.2 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/audio_ring_buffer.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/channel_buffer.h"
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namespace webrtc {
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class AudioRingBufferTest :
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public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
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};
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void ReadAndWriteTest(const ChannelBuffer<float>& input,
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size_t num_write_chunk_frames,
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size_t num_read_chunk_frames,
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size_t buffer_frames,
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ChannelBuffer<float>* output) {
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const size_t num_channels = input.num_channels();
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const size_t total_frames = input.num_frames();
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AudioRingBuffer buf(num_channels, buffer_frames);
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scoped_ptr<float*[]> slice(new float*[num_channels]);
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size_t input_pos = 0;
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size_t output_pos = 0;
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while (input_pos + buf.WriteFramesAvailable() < total_frames) {
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// Write until the buffer is as full as possible.
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while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
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buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
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num_channels, num_write_chunk_frames);
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input_pos += num_write_chunk_frames;
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}
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// Read until the buffer is as empty as possible.
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while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
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EXPECT_LT(output_pos, total_frames);
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buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
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num_channels, num_read_chunk_frames);
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output_pos += num_read_chunk_frames;
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}
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}
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// Write and read the last bit.
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if (input_pos < total_frames)
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buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
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num_channels, total_frames - input_pos);
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if (buf.ReadFramesAvailable())
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buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
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num_channels, buf.ReadFramesAvailable());
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EXPECT_EQ(0u, buf.ReadFramesAvailable());
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}
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TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
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const size_t kFrames = 5000;
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const size_t num_channels = ::testing::get<3>(GetParam());
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// Initialize the input data to an increasing sequence.
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ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
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for (size_t i = 0; i < num_channels; ++i)
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for (size_t j = 0; j < kFrames; ++j)
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input.channels()[i][j] = (i + 1) * (j + 1);
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ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
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ReadAndWriteTest(input,
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::testing::get<0>(GetParam()),
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::testing::get<1>(GetParam()),
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::testing::get<2>(GetParam()),
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&output);
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// Verify the read data matches the input.
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for (size_t i = 0; i < num_channels; ++i)
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for (size_t j = 0; j < kFrames; ++j)
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EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
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}
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INSTANTIATE_TEST_CASE_P(
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AudioRingBufferTest, AudioRingBufferTest,
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::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames
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::testing::Values(1, 10, 17), // num_read_chunk_frames
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::testing::Values(100, 256), // buffer_frames
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::testing::Values(1, 4))); // num_channels
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TEST_F(AudioRingBufferTest, MoveReadPosition) {
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const size_t kNumChannels = 1;
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const float kInputArray[] = {1, 2, 3, 4};
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const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
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ChannelBuffer<float> input(kNumFrames, kNumChannels);
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input.SetDataForTesting(kInputArray, kNumFrames);
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AudioRingBuffer buf(kNumChannels, kNumFrames);
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buf.Write(input.channels(), kNumChannels, kNumFrames);
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buf.MoveReadPosition(3);
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ChannelBuffer<float> output(1, kNumChannels);
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buf.Read(output.channels(), kNumChannels, 1);
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EXPECT_EQ(4, output.channels()[0][0]);
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buf.MoveReadPosition(-3);
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buf.Read(output.channels(), kNumChannels, 1);
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EXPECT_EQ(2, output.channels()[0][0]);
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}
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} // namespace webrtc
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