diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/test')
10 files changed, 1412 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn new file mode 100644 index 0000000000..8bc0bf5e0e --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn @@ -0,0 +1,55 @@ +# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("../../../../../webrtc.gni") + +visibility = [ + ":*", + "../../../:*", +] + +if (rtc_include_tests) { + rtc_library("test") { + testonly = true + + sources = [ + "audio_ring_buffer.cc", + "audio_ring_buffer.h", + "blocker.cc", + "blocker.h", + "lapped_transform.cc", + "lapped_transform.h", + ] + + deps = [ + "../../../../../common_audio", + "../../../../../common_audio:common_audio_c", + "../../../../../rtc_base:checks", + "../../../../../rtc_base/memory:aligned_malloc", + ] + } + + rtc_library("test_unittest") { + testonly = true + + sources = [ + "audio_ring_buffer_unittest.cc", + "blocker_unittest.cc", + "lapped_transform_unittest.cc", + ] + + deps = [ + ":test", + "../../../../../common_audio", + "../../../../../common_audio:common_audio_c", + "../../../../../rtc_base:macromagic", + "../../../../../test:test_support", + "//testing/gtest", + ] + } +} diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc new file mode 100644 index 0000000000..2a71b43d2c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc @@ -0,0 +1,76 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" + +#include "common_audio/ring_buffer.h" +#include "rtc_base/checks.h" + +// This is a simple multi-channel wrapper over the ring_buffer.h C interface. + +namespace webrtc { + +AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) { + buffers_.reserve(channels); + for (size_t i = 0; i < channels; ++i) + buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float))); +} + +AudioRingBuffer::~AudioRingBuffer() { + for (auto* buf : buffers_) + WebRtc_FreeBuffer(buf); +} + +void AudioRingBuffer::Write(const float* const* data, + size_t channels, + size_t frames) { + RTC_DCHECK_EQ(buffers_.size(), channels); + for (size_t i = 0; i < channels; ++i) { + const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames); + RTC_CHECK_EQ(written, frames); + } +} + +void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) { + RTC_DCHECK_EQ(buffers_.size(), channels); + for (size_t i = 0; i < channels; ++i) { + const size_t read = + WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames); + RTC_CHECK_EQ(read, frames); + } +} + +size_t AudioRingBuffer::ReadFramesAvailable() const { + // All buffers have the same amount available. + return WebRtc_available_read(buffers_[0]); +} + +size_t AudioRingBuffer::WriteFramesAvailable() const { + // All buffers have the same amount available. + return WebRtc_available_write(buffers_[0]); +} + +void AudioRingBuffer::MoveReadPositionForward(size_t frames) { + for (auto* buf : buffers_) { + const size_t moved = + static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames))); + RTC_CHECK_EQ(moved, frames); + } +} + +void AudioRingBuffer::MoveReadPositionBackward(size_t frames) { + for (auto* buf : buffers_) { + const size_t moved = static_cast<size_t>( + -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames))); + RTC_CHECK_EQ(moved, frames); + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h new file mode 100644 index 0000000000..a280ca2410 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h @@ -0,0 +1,57 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_ + +#include <stddef.h> + +#include <memory> +#include <vector> + +struct RingBuffer; + +namespace webrtc { + +// A ring buffer tailored for float deinterleaved audio. Any operation that +// cannot be performed as requested will cause a crash (e.g. insufficient data +// in the buffer to fulfill a read request.) +class AudioRingBuffer final { + public: + // Specify the number of channels and maximum number of frames the buffer will + // contain. + AudioRingBuffer(size_t channels, size_t max_frames); + ~AudioRingBuffer(); + + // Copies `data` to the buffer and advances the write pointer. `channels` must + // be the same as at creation time. + void Write(const float* const* data, size_t channels, size_t frames); + + // Copies from the buffer to `data` and advances the read pointer. `channels` + // must be the same as at creation time. + void Read(float* const* data, size_t channels, size_t frames); + + size_t ReadFramesAvailable() const; + size_t WriteFramesAvailable() const; + + // Moves the read position. The forward version advances the read pointer + // towards the write pointer and the backward verison withdraws the read + // pointer away from the write pointer (i.e. flushing and stuffing the buffer + // respectively.) + void MoveReadPositionForward(size_t frames); + void MoveReadPositionBackward(size_t frames); + + private: + // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead. + std::vector<RingBuffer*> buffers_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc new file mode 100644 index 0000000000..6dbc8ee9fe --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc @@ -0,0 +1,111 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" + +#include <memory> + +#include "common_audio/channel_buffer.h" +#include "test/gtest.h" + +namespace webrtc { + +class AudioRingBufferTest + : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > { +}; + +void ReadAndWriteTest(const ChannelBuffer<float>& input, + size_t num_write_chunk_frames, + size_t num_read_chunk_frames, + size_t buffer_frames, + ChannelBuffer<float>* output) { + const size_t num_channels = input.num_channels(); + const size_t total_frames = input.num_frames(); + AudioRingBuffer buf(num_channels, buffer_frames); + std::unique_ptr<float*[]> slice(new float*[num_channels]); + + size_t input_pos = 0; + size_t output_pos = 0; + while (input_pos + buf.WriteFramesAvailable() < total_frames) { + // Write until the buffer is as full as possible. + while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { + buf.Write(input.Slice(slice.get(), input_pos), num_channels, + num_write_chunk_frames); + input_pos += num_write_chunk_frames; + } + // Read until the buffer is as empty as possible. + while (buf.ReadFramesAvailable() >= num_read_chunk_frames) { + EXPECT_LT(output_pos, total_frames); + buf.Read(output->Slice(slice.get(), output_pos), num_channels, + num_read_chunk_frames); + output_pos += num_read_chunk_frames; + } + } + + // Write and read the last bit. + if (input_pos < total_frames) { + buf.Write(input.Slice(slice.get(), input_pos), num_channels, + total_frames - input_pos); + } + if (buf.ReadFramesAvailable()) { + buf.Read(output->Slice(slice.get(), output_pos), num_channels, + buf.ReadFramesAvailable()); + } + EXPECT_EQ(0u, buf.ReadFramesAvailable()); +} + +TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) { + const size_t kFrames = 5000; + const size_t num_channels = ::testing::get<3>(GetParam()); + + // Initialize the input data to an increasing sequence. + ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); + for (size_t i = 0; i < num_channels; ++i) + for (size_t j = 0; j < kFrames; ++j) + input.channels()[i][j] = (i + 1) * (j + 1); + + ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels)); + ReadAndWriteTest(input, ::testing::get<0>(GetParam()), + ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()), + &output); + + // Verify the read data matches the input. + for (size_t i = 0; i < num_channels; ++i) + for (size_t j = 0; j < kFrames; ++j) + EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]); +} + +INSTANTIATE_TEST_SUITE_P( + AudioRingBufferTest, + AudioRingBufferTest, + ::testing::Combine(::testing::Values(10, 20, 42), // num_write_chunk_frames + ::testing::Values(1, 10, 17), // num_read_chunk_frames + ::testing::Values(100, 256), // buffer_frames + ::testing::Values(1, 4))); // num_channels + +TEST_F(AudioRingBufferTest, MoveReadPosition) { + const size_t kNumChannels = 1; + const float kInputArray[] = {1, 2, 3, 4}; + const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray); + ChannelBuffer<float> input(kNumFrames, kNumChannels); + input.SetDataForTesting(kInputArray, kNumFrames); + AudioRingBuffer buf(kNumChannels, kNumFrames); + buf.Write(input.channels(), kNumChannels, kNumFrames); + + buf.MoveReadPositionForward(3); + ChannelBuffer<float> output(1, kNumChannels); + buf.Read(output.channels(), kNumChannels, 1); + EXPECT_EQ(4, output.channels()[0][0]); + buf.MoveReadPositionBackward(3); + buf.Read(output.channels(), kNumChannels, 1); + EXPECT_EQ(2, output.channels()[0][0]); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc new file mode 100644 index 0000000000..33406cead9 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc @@ -0,0 +1,215 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/blocker.h" + +#include <string.h> + +#include "rtc_base/checks.h" + +namespace { + +// Adds `a` and `b` frame by frame into `result` (basically matrix addition). +void AddFrames(const float* const* a, + size_t a_start_index, + const float* const* b, + int b_start_index, + size_t num_frames, + size_t num_channels, + float* const* result, + size_t result_start_index) { + for (size_t i = 0; i < num_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + result[i][j + result_start_index] = + a[i][j + a_start_index] + b[i][j + b_start_index]; + } + } +} + +// Copies `src` into `dst` channel by channel. +void CopyFrames(const float* const* src, + size_t src_start_index, + size_t num_frames, + size_t num_channels, + float* const* dst, + size_t dst_start_index) { + for (size_t i = 0; i < num_channels; ++i) { + memcpy(&dst[i][dst_start_index], &src[i][src_start_index], + num_frames * sizeof(dst[i][dst_start_index])); + } +} + +// Moves `src` into `dst` channel by channel. +void MoveFrames(const float* const* src, + size_t src_start_index, + size_t num_frames, + size_t num_channels, + float* const* dst, + size_t dst_start_index) { + for (size_t i = 0; i < num_channels; ++i) { + memmove(&dst[i][dst_start_index], &src[i][src_start_index], + num_frames * sizeof(dst[i][dst_start_index])); + } +} + +void ZeroOut(float* const* buffer, + size_t starting_idx, + size_t num_frames, + size_t num_channels) { + for (size_t i = 0; i < num_channels; ++i) { + memset(&buffer[i][starting_idx], 0, + num_frames * sizeof(buffer[i][starting_idx])); + } +} + +// Pointwise multiplies each channel of `frames` with `window`. Results are +// stored in `frames`. +void ApplyWindow(const float* window, + size_t num_frames, + size_t num_channels, + float* const* frames) { + for (size_t i = 0; i < num_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + frames[i][j] = frames[i][j] * window[j]; + } + } +} + +size_t gcd(size_t a, size_t b) { + size_t tmp; + while (b) { + tmp = a; + a = b; + b = tmp % b; + } + return a; +} + +} // namespace + +namespace webrtc { + +Blocker::Blocker(size_t chunk_size, + size_t block_size, + size_t num_input_channels, + size_t num_output_channels, + const float* window, + size_t shift_amount, + BlockerCallback* callback) + : chunk_size_(chunk_size), + block_size_(block_size), + num_input_channels_(num_input_channels), + num_output_channels_(num_output_channels), + initial_delay_(block_size_ - gcd(chunk_size, shift_amount)), + frame_offset_(0), + input_buffer_(num_input_channels_, chunk_size_ + initial_delay_), + output_buffer_(chunk_size_ + initial_delay_, num_output_channels_), + input_block_(block_size_, num_input_channels_), + output_block_(block_size_, num_output_channels_), + window_(new float[block_size_]), + shift_amount_(shift_amount), + callback_(callback) { + RTC_CHECK_LE(num_output_channels_, num_input_channels_); + RTC_CHECK_LE(shift_amount_, block_size_); + + memcpy(window_.get(), window, block_size_ * sizeof(*window_.get())); + input_buffer_.MoveReadPositionBackward(initial_delay_); +} + +Blocker::~Blocker() = default; + +// When block_size < chunk_size the input and output buffers look like this: +// +// delay* chunk_size chunk_size + delay* +// buffer: <-------------|---------------------|---------------|> +// _a_ _b_ _c_ +// +// On each call to ProcessChunk(): +// 1. New input gets read into sections _b_ and _c_ of the input buffer. +// 2. We block starting from frame_offset. +// 3. We block until we reach a block `bl` that doesn't contain any frames +// from sections _a_ or _b_ of the input buffer. +// 4. We window the current block, fire the callback for processing, window +// again, and overlap/add to the output buffer. +// 5. We copy sections _a_ and _b_ of the output buffer into output. +// 6. For both the input and the output buffers, we copy section _c_ into +// section _a_. +// 7. We set the new frame_offset to be the difference between the first frame +// of `bl` and the border between sections _b_ and _c_. +// +// When block_size > chunk_size the input and output buffers look like this: +// +// chunk_size delay* chunk_size + delay* +// buffer: <-------------|---------------------|---------------|> +// _a_ _b_ _c_ +// +// On each call to ProcessChunk(): +// The procedure is the same as above, except for: +// 1. New input gets read into section _c_ of the input buffer. +// 3. We block until we reach a block `bl` that doesn't contain any frames +// from section _a_ of the input buffer. +// 5. We copy section _a_ of the output buffer into output. +// 6. For both the input and the output buffers, we copy sections _b_ and _c_ +// into section _a_ and _b_. +// 7. We set the new frame_offset to be the difference between the first frame +// of `bl` and the border between sections _a_ and _b_. +// +// * delay here refers to inintial_delay_ +// +// TODO(claguna): Look at using ring buffers to eliminate some copies. +void Blocker::ProcessChunk(const float* const* input, + size_t chunk_size, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) { + RTC_CHECK_EQ(chunk_size, chunk_size_); + RTC_CHECK_EQ(num_input_channels, num_input_channels_); + RTC_CHECK_EQ(num_output_channels, num_output_channels_); + + input_buffer_.Write(input, num_input_channels, chunk_size_); + size_t first_frame_in_block = frame_offset_; + + // Loop through blocks. + while (first_frame_in_block < chunk_size_) { + input_buffer_.Read(input_block_.channels(), num_input_channels, + block_size_); + input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_); + + ApplyWindow(window_.get(), block_size_, num_input_channels_, + input_block_.channels()); + callback_->ProcessBlock(input_block_.channels(), block_size_, + num_input_channels_, num_output_channels_, + output_block_.channels()); + ApplyWindow(window_.get(), block_size_, num_output_channels_, + output_block_.channels()); + + AddFrames(output_buffer_.channels(), first_frame_in_block, + output_block_.channels(), 0, block_size_, num_output_channels_, + output_buffer_.channels(), first_frame_in_block); + + first_frame_in_block += shift_amount_; + } + + // Copy output buffer to output + CopyFrames(output_buffer_.channels(), 0, chunk_size_, num_output_channels_, + output, 0); + + // Copy output buffer [chunk_size_, chunk_size_ + initial_delay] + // to output buffer [0, initial_delay], zero the rest. + MoveFrames(output_buffer_.channels(), chunk_size, initial_delay_, + num_output_channels_, output_buffer_.channels(), 0); + ZeroOut(output_buffer_.channels(), initial_delay_, chunk_size_, + num_output_channels_); + + // Calculate new starting frames. + frame_offset_ = first_frame_in_block - chunk_size_; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h new file mode 100644 index 0000000000..59b7e29621 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h @@ -0,0 +1,127 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_ + +#include <memory> + +#include "common_audio/channel_buffer.h" +#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" + +namespace webrtc { + +// The callback function to process audio in the time domain. Input has already +// been windowed, and output will be windowed. The number of input channels +// must be >= the number of output channels. +class BlockerCallback { + public: + virtual ~BlockerCallback() {} + + virtual void ProcessBlock(const float* const* input, + size_t num_frames, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) = 0; +}; + +// The main purpose of Blocker is to abstract away the fact that often we +// receive a different number of audio frames than our transform takes. For +// example, most FFTs work best when the fft-size is a power of 2, but suppose +// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames +// of audio, which is not a power of 2. Blocker allows us to specify the +// transform and all other necessary processing via the Process() callback +// function without any constraints on the transform-size +// (read: `block_size_`) or received-audio-size (read: `chunk_size_`). +// We handle this for the multichannel audio case, allowing for different +// numbers of input and output channels (for example, beamforming takes 2 or +// more input channels and returns 1 output channel). Audio signals are +// represented as deinterleaved floats in the range [-1, 1]. +// +// Blocker is responsible for: +// - blocking audio while handling potential discontinuities on the edges +// of chunks +// - windowing blocks before sending them to Process() +// - windowing processed blocks, and overlap-adding them together before +// sending back a processed chunk +// +// To use blocker: +// 1. Impelment a BlockerCallback object `bc`. +// 2. Instantiate a Blocker object `b`, passing in `bc`. +// 3. As you receive audio, call b.ProcessChunk() to get processed audio. +// +// A small amount of delay is added to the first received chunk to deal with +// the difference in chunk/block sizes. This delay is <= chunk_size. +// +// Ownership of window is retained by the caller. That is, Blocker makes a +// copy of window and does not attempt to delete it. +class Blocker { + public: + Blocker(size_t chunk_size, + size_t block_size, + size_t num_input_channels, + size_t num_output_channels, + const float* window, + size_t shift_amount, + BlockerCallback* callback); + ~Blocker(); + + void ProcessChunk(const float* const* input, + size_t chunk_size, + size_t num_input_channels, + size_t num_output_channels, + float* const* output); + + size_t initial_delay() const { return initial_delay_; } + + private: + const size_t chunk_size_; + const size_t block_size_; + const size_t num_input_channels_; + const size_t num_output_channels_; + + // The number of frames of delay to add at the beginning of the first chunk. + const size_t initial_delay_; + + // The frame index into the input buffer where the first block should be read + // from. This is necessary because shift_amount_ is not necessarily a + // multiple of chunk_size_, so blocks won't line up at the start of the + // buffer. + size_t frame_offset_; + + // Since blocks nearly always overlap, there are certain blocks that require + // frames from the end of one chunk and the beginning of the next chunk. The + // input and output buffers are responsible for saving those frames between + // calls to ProcessChunk(). + // + // Both contain |initial delay| + `chunk_size` frames. The input is a fairly + // standard FIFO, but due to the overlap-add it's harder to use an + // AudioRingBuffer for the output. + AudioRingBuffer input_buffer_; + ChannelBuffer<float> output_buffer_; + + // Space for the input block (can't wrap because of windowing). + ChannelBuffer<float> input_block_; + + // Space for the output block (can't wrap because of overlap/add). + ChannelBuffer<float> output_block_; + + std::unique_ptr<float[]> window_; + + // The amount of frames between the start of contiguous blocks. For example, + // `shift_amount_` = `block_size_` / 2 for a Hann window. + size_t shift_amount_; + + BlockerCallback* callback_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc new file mode 100644 index 0000000000..9c8e789ba9 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc @@ -0,0 +1,293 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/blocker.h" + +#include <memory> + +#include "rtc_base/arraysize.h" +#include "test/gtest.h" + +namespace { + +// Callback Function to add 3 to every sample in the signal. +class PlusThreeBlockerCallback : public webrtc::BlockerCallback { + public: + void ProcessBlock(const float* const* input, + size_t num_frames, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) override { + for (size_t i = 0; i < num_output_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + output[i][j] = input[i][j] + 3; + } + } + } +}; + +// No-op Callback Function. +class CopyBlockerCallback : public webrtc::BlockerCallback { + public: + void ProcessBlock(const float* const* input, + size_t num_frames, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) override { + for (size_t i = 0; i < num_output_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + output[i][j] = input[i][j]; + } + } + } +}; + +} // namespace + +namespace webrtc { + +// Tests blocking with a window that multiplies the signal by 2, a callback +// that adds 3 to each sample in the signal, and different combinations of chunk +// size, block size, and shift amount. +class BlockerTest : public ::testing::Test { + protected: + void RunTest(Blocker* blocker, + size_t chunk_size, + size_t num_frames, + const float* const* input, + float* const* input_chunk, + float* const* output, + float* const* output_chunk, + size_t num_input_channels, + size_t num_output_channels) { + size_t start = 0; + size_t end = chunk_size - 1; + while (end < num_frames) { + CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input); + blocker->ProcessChunk(input_chunk, chunk_size, num_input_channels, + num_output_channels, output_chunk); + CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk); + + start += chunk_size; + end += chunk_size; + } + } + + void ValidateSignalEquality(const float* const* expected, + const float* const* actual, + size_t num_channels, + size_t num_frames) { + for (size_t i = 0; i < num_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]); + } + } + } + + void ValidateInitialDelay(const float* const* output, + size_t num_channels, + size_t num_frames, + size_t initial_delay) { + for (size_t i = 0; i < num_channels; ++i) { + for (size_t j = 0; j < num_frames; ++j) { + if (j < initial_delay) { + EXPECT_FLOAT_EQ(output[i][j], 0.f); + } else { + EXPECT_GT(output[i][j], 0.f); + } + } + } + } + + static void CopyTo(float* const* dst, + size_t start_index_dst, + size_t start_index_src, + size_t num_channels, + size_t num_frames, + const float* const* src) { + for (size_t i = 0; i < num_channels; ++i) { + memcpy(&dst[i][start_index_dst], &src[i][start_index_src], + num_frames * sizeof(float)); + } + } +}; + +TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) { + const size_t kNumInputChannels = 3; + const size_t kNumOutputChannels = 2; + const size_t kNumFrames = 10; + const size_t kBlockSize = 4; + const size_t kChunkSize = 5; + const size_t kShiftAmount = 2; + + const float kInput[kNumInputChannels][kNumFrames] = { + {1, 1, 1, 1, 1, 1, 1, 1, 1, 1}, + {2, 2, 2, 2, 2, 2, 2, 2, 2, 2}, + {3, 3, 3, 3, 3, 3, 3, 3, 3, 3}}; + ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels); + input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput)); + + const float kExpectedOutput[kNumInputChannels][kNumFrames] = { + {6, 6, 12, 20, 20, 20, 20, 20, 20, 20}, + {6, 6, 12, 28, 28, 28, 28, 28, 28, 28}}; + ChannelBuffer<float> expected_output_cb(kNumFrames, kNumInputChannels); + expected_output_cb.SetDataForTesting( + kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput)); + + const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f}; + + ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels); + ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels); + ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels); + + PlusThreeBlockerCallback callback; + Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels, + kWindow, kShiftAmount, &callback); + + RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(), + input_chunk_cb.channels(), actual_output_cb.channels(), + output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels); + + ValidateSignalEquality(expected_output_cb.channels(), + actual_output_cb.channels(), kNumOutputChannels, + kNumFrames); +} + +TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) { + const size_t kNumInputChannels = 3; + const size_t kNumOutputChannels = 2; + const size_t kNumFrames = 12; + const size_t kBlockSize = 4; + const size_t kChunkSize = 6; + const size_t kShiftAmount = 3; + + const float kInput[kNumInputChannels][kNumFrames] = { + {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}, + {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2}, + {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}}; + ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels); + input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput)); + + const float kExpectedOutput[kNumOutputChannels][kNumFrames] = { + {6, 10, 10, 20, 10, 10, 20, 10, 10, 20, 10, 10}, + {6, 14, 14, 28, 14, 14, 28, 14, 14, 28, 14, 14}}; + ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels); + expected_output_cb.SetDataForTesting( + kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput)); + + const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f}; + + ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels); + ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels); + ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels); + + PlusThreeBlockerCallback callback; + Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels, + kWindow, kShiftAmount, &callback); + + RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(), + input_chunk_cb.channels(), actual_output_cb.channels(), + output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels); + + ValidateSignalEquality(expected_output_cb.channels(), + actual_output_cb.channels(), kNumOutputChannels, + kNumFrames); +} + +TEST_F(BlockerTest, TestBlockerNoOverlap) { + const size_t kNumInputChannels = 3; + const size_t kNumOutputChannels = 2; + const size_t kNumFrames = 12; + const size_t kBlockSize = 4; + const size_t kChunkSize = 4; + const size_t kShiftAmount = 4; + + const float kInput[kNumInputChannels][kNumFrames] = { + {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}, + {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2}, + {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}}; + ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels); + input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput)); + + const float kExpectedOutput[kNumOutputChannels][kNumFrames] = { + {10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10}, + {14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}}; + ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels); + expected_output_cb.SetDataForTesting( + kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput)); + + const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f}; + + ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels); + ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels); + ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels); + + PlusThreeBlockerCallback callback; + Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels, + kWindow, kShiftAmount, &callback); + + RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(), + input_chunk_cb.channels(), actual_output_cb.channels(), + output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels); + + ValidateSignalEquality(expected_output_cb.channels(), + actual_output_cb.channels(), kNumOutputChannels, + kNumFrames); +} + +TEST_F(BlockerTest, InitialDelaysAreMinimum) { + const size_t kNumInputChannels = 3; + const size_t kNumOutputChannels = 2; + const size_t kNumFrames = 1280; + const size_t kChunkSize[] = {80, 80, 80, 80, 80, 80, + 160, 160, 160, 160, 160, 160}; + const size_t kBlockSize[] = {64, 64, 64, 128, 128, 128, + 128, 128, 128, 256, 256, 256}; + const size_t kShiftAmount[] = {16, 32, 64, 32, 64, 128, + 32, 64, 128, 64, 128, 256}; + const size_t kInitialDelay[] = {48, 48, 48, 112, 112, 112, + 96, 96, 96, 224, 224, 224}; + + float input[kNumInputChannels][kNumFrames]; + for (size_t i = 0; i < kNumInputChannels; ++i) { + for (size_t j = 0; j < kNumFrames; ++j) { + input[i][j] = i + 1; + } + } + ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels); + input_cb.SetDataForTesting(input[0], sizeof(input) / sizeof(**input)); + + ChannelBuffer<float> output_cb(kNumFrames, kNumOutputChannels); + + CopyBlockerCallback callback; + + for (size_t i = 0; i < arraysize(kChunkSize); ++i) { + std::unique_ptr<float[]> window(new float[kBlockSize[i]]); + for (size_t j = 0; j < kBlockSize[i]; ++j) { + window[j] = 1.f; + } + + ChannelBuffer<float> input_chunk_cb(kChunkSize[i], kNumInputChannels); + ChannelBuffer<float> output_chunk_cb(kChunkSize[i], kNumOutputChannels); + + Blocker blocker(kChunkSize[i], kBlockSize[i], kNumInputChannels, + kNumOutputChannels, window.get(), kShiftAmount[i], + &callback); + + RunTest(&blocker, kChunkSize[i], kNumFrames, input_cb.channels(), + input_chunk_cb.channels(), output_cb.channels(), + output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels); + + ValidateInitialDelay(output_cb.channels(), kNumOutputChannels, kNumFrames, + kInitialDelay[i]); + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc new file mode 100644 index 0000000000..b1a6526bba --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc @@ -0,0 +1,100 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/lapped_transform.h" + +#include <algorithm> +#include <cstdlib> +#include <cstring> + +#include "common_audio/real_fourier.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +void LappedTransform::BlockThunk::ProcessBlock(const float* const* input, + size_t num_frames, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) { + RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_); + RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_); + RTC_CHECK_EQ(parent_->block_length_, num_frames); + + for (size_t i = 0; i < num_input_channels; ++i) { + memcpy(parent_->real_buf_.Row(i), input[i], num_frames * sizeof(*input[0])); + parent_->fft_->Forward(parent_->real_buf_.Row(i), + parent_->cplx_pre_.Row(i)); + } + + size_t block_length = + RealFourier::ComplexLength(RealFourier::FftOrder(num_frames)); + RTC_CHECK_EQ(parent_->cplx_length_, block_length); + parent_->block_processor_->ProcessAudioBlock( + parent_->cplx_pre_.Array(), num_input_channels, parent_->cplx_length_, + num_output_channels, parent_->cplx_post_.Array()); + + for (size_t i = 0; i < num_output_channels; ++i) { + parent_->fft_->Inverse(parent_->cplx_post_.Row(i), + parent_->real_buf_.Row(i)); + memcpy(output[i], parent_->real_buf_.Row(i), + num_frames * sizeof(*input[0])); + } +} + +LappedTransform::LappedTransform(size_t num_in_channels, + size_t num_out_channels, + size_t chunk_length, + const float* window, + size_t block_length, + size_t shift_amount, + Callback* callback) + : blocker_callback_(this), + num_in_channels_(num_in_channels), + num_out_channels_(num_out_channels), + block_length_(block_length), + chunk_length_(chunk_length), + block_processor_(callback), + blocker_(chunk_length_, + block_length_, + num_in_channels_, + num_out_channels_, + window, + shift_amount, + &blocker_callback_), + fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))), + cplx_length_(RealFourier::ComplexLength(fft_->order())), + real_buf_(num_in_channels, + block_length_, + RealFourier::kFftBufferAlignment), + cplx_pre_(num_in_channels, + cplx_length_, + RealFourier::kFftBufferAlignment), + cplx_post_(num_out_channels, + cplx_length_, + RealFourier::kFftBufferAlignment) { + RTC_CHECK(num_in_channels_ > 0); + RTC_CHECK_GT(block_length_, 0); + RTC_CHECK_GT(chunk_length_, 0); + RTC_CHECK(block_processor_); + + // block_length_ power of 2? + RTC_CHECK_EQ(0, block_length_ & (block_length_ - 1)); +} + +LappedTransform::~LappedTransform() = default; + +void LappedTransform::ProcessChunk(const float* const* in_chunk, + float* const* out_chunk) { + blocker_.ProcessChunk(in_chunk, chunk_length_, num_in_channels_, + num_out_channels_, out_chunk); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h new file mode 100644 index 0000000000..bb25c34a9e --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h @@ -0,0 +1,175 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_ + +#include <complex> +#include <memory> + +#include "common_audio/real_fourier.h" +#include "modules/audio_coding/codecs/opus/test/blocker.h" +#include "rtc_base/memory/aligned_malloc.h" + +namespace webrtc { + +// Wrapper class for aligned arrays. Every row (and the first dimension) are +// aligned to the given byte alignment. +template <typename T> +class AlignedArray { + public: + AlignedArray(size_t rows, size_t cols, size_t alignment) + : rows_(rows), cols_(cols) { + RTC_CHECK_GT(alignment, 0); + head_row_ = + static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_), alignment)); + for (size_t i = 0; i < rows_; ++i) { + head_row_[i] = static_cast<T*>( + AlignedMalloc(cols_ * sizeof(**head_row_), alignment)); + } + } + + ~AlignedArray() { + for (size_t i = 0; i < rows_; ++i) { + AlignedFree(head_row_[i]); + } + AlignedFree(head_row_); + } + + T* const* Array() { return head_row_; } + + const T* const* Array() const { return head_row_; } + + T* Row(size_t row) { + RTC_CHECK_LE(row, rows_); + return head_row_[row]; + } + + const T* Row(size_t row) const { + RTC_CHECK_LE(row, rows_); + return head_row_[row]; + } + + private: + size_t rows_; + size_t cols_; + T** head_row_; +}; + +// Helper class for audio processing modules which operate on frequency domain +// input derived from the windowed time domain audio stream. +// +// The input audio chunk is sliced into possibly overlapping blocks, multiplied +// by a window and transformed with an FFT implementation. The transformed data +// is supplied to the given callback for processing. The processed output is +// then inverse transformed into the time domain and spliced back into a chunk +// which constitutes the final output of this processing module. +class LappedTransform { + public: + class Callback { + public: + virtual ~Callback() {} + + virtual void ProcessAudioBlock(const std::complex<float>* const* in_block, + size_t num_in_channels, + size_t frames, + size_t num_out_channels, + std::complex<float>* const* out_block) = 0; + }; + + // Construct a transform instance. `chunk_length` is the number of samples in + // each channel. `window` defines the window, owned by the caller (a copy is + // made internally); `window` should have length equal to `block_length`. + // `block_length` defines the length of a block, in samples. + // `shift_amount` is in samples. `callback` is the caller-owned audio + // processing function called for each block of the input chunk. + LappedTransform(size_t num_in_channels, + size_t num_out_channels, + size_t chunk_length, + const float* window, + size_t block_length, + size_t shift_amount, + Callback* callback); + ~LappedTransform(); + + // Main audio processing helper method. Internally slices `in_chunk` into + // blocks, transforms them to frequency domain, calls the callback for each + // block and returns a de-blocked time domain chunk of audio through + // `out_chunk`. Both buffers are caller-owned. + void ProcessChunk(const float* const* in_chunk, float* const* out_chunk); + + // Get the chunk length. + // + // The chunk length is the number of samples per channel that must be passed + // to ProcessChunk via the parameter in_chunk. + // + // Returns the same chunk_length passed to the LappedTransform constructor. + size_t chunk_length() const { return chunk_length_; } + + // Get the number of input channels. + // + // This is the number of arrays that must be passed to ProcessChunk via + // in_chunk. + // + // Returns the same num_in_channels passed to the LappedTransform constructor. + size_t num_in_channels() const { return num_in_channels_; } + + // Get the number of output channels. + // + // This is the number of arrays that must be passed to ProcessChunk via + // out_chunk. + // + // Returns the same num_out_channels passed to the LappedTransform + // constructor. + size_t num_out_channels() const { return num_out_channels_; } + + // Returns the initial delay. + // + // This is the delay introduced by the `blocker_` to be able to get and return + // chunks of `chunk_length`, but process blocks of `block_length`. + size_t initial_delay() const { return blocker_.initial_delay(); } + + private: + // Internal middleware callback, given to the blocker. Transforms each block + // and hands it over to the processing method given at construction time. + class BlockThunk : public BlockerCallback { + public: + explicit BlockThunk(LappedTransform* parent) : parent_(parent) {} + + void ProcessBlock(const float* const* input, + size_t num_frames, + size_t num_input_channels, + size_t num_output_channels, + float* const* output) override; + + private: + LappedTransform* const parent_; + } blocker_callback_; + + const size_t num_in_channels_; + const size_t num_out_channels_; + + const size_t block_length_; + const size_t chunk_length_; + + Callback* const block_processor_; + Blocker blocker_; + + // TODO(alessiob): Replace RealFourier with a different FFT library. + std::unique_ptr<RealFourier> fft_; + const size_t cplx_length_; + AlignedArray<float> real_buf_; + AlignedArray<std::complex<float> > cplx_pre_; + AlignedArray<std::complex<float> > cplx_post_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc new file mode 100644 index 0000000000..1003ed52e5 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc @@ -0,0 +1,203 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/opus/test/lapped_transform.h" + +#include <algorithm> +#include <cmath> +#include <cstring> + +#include "test/gtest.h" + +using std::complex; + +namespace { + +class NoopCallback : public webrtc::LappedTransform::Callback { + public: + NoopCallback() : block_num_(0) {} + + void ProcessAudioBlock(const complex<float>* const* in_block, + size_t in_channels, + size_t frames, + size_t out_channels, + complex<float>* const* out_block) override { + RTC_CHECK_EQ(in_channels, out_channels); + for (size_t i = 0; i < out_channels; ++i) { + memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames); + } + ++block_num_; + } + + size_t block_num() { return block_num_; } + + private: + size_t block_num_; +}; + +class FftCheckerCallback : public webrtc::LappedTransform::Callback { + public: + FftCheckerCallback() : block_num_(0) {} + + void ProcessAudioBlock(const complex<float>* const* in_block, + size_t in_channels, + size_t frames, + size_t out_channels, + complex<float>* const* out_block) override { + RTC_CHECK_EQ(in_channels, out_channels); + + size_t full_length = (frames - 1) * 2; + ++block_num_; + + if (block_num_ > 0) { + ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length), + 1e-5f); + ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f); + for (size_t i = 1; i < frames; ++i) { + ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f); + ASSERT_NEAR(in_block[0][i].imag(), 0.0f, 1e-5f); + } + } + } + + size_t block_num() { return block_num_; } + + private: + size_t block_num_; +}; + +void SetFloatArray(float value, int rows, int cols, float* const* array) { + for (int i = 0; i < rows; ++i) { + for (int j = 0; j < cols; ++j) { + array[i][j] = value; + } + } +} + +} // namespace + +namespace webrtc { + +TEST(LappedTransformTest, Windowless) { + const size_t kChannels = 3; + const size_t kChunkLength = 512; + const size_t kBlockLength = 64; + const size_t kShiftAmount = 64; + NoopCallback noop; + + // Rectangular window. + float window[kBlockLength]; + std::fill(window, &window[kBlockLength], 1.0f); + + LappedTransform trans(kChannels, kChannels, kChunkLength, window, + kBlockLength, kShiftAmount, &noop); + float in_buffer[kChannels][kChunkLength]; + float* in_chunk[kChannels]; + float out_buffer[kChannels][kChunkLength]; + float* out_chunk[kChannels]; + + in_chunk[0] = in_buffer[0]; + in_chunk[1] = in_buffer[1]; + in_chunk[2] = in_buffer[2]; + out_chunk[0] = out_buffer[0]; + out_chunk[1] = out_buffer[1]; + out_chunk[2] = out_buffer[2]; + SetFloatArray(2.0f, kChannels, kChunkLength, in_chunk); + SetFloatArray(-1.0f, kChannels, kChunkLength, out_chunk); + + trans.ProcessChunk(in_chunk, out_chunk); + + for (size_t i = 0; i < kChannels; ++i) { + for (size_t j = 0; j < kChunkLength; ++j) { + ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f); + } + } + + ASSERT_EQ(kChunkLength / kBlockLength, noop.block_num()); +} + +TEST(LappedTransformTest, IdentityProcessor) { + const size_t kChunkLength = 512; + const size_t kBlockLength = 64; + const size_t kShiftAmount = 32; + NoopCallback noop; + + // Identity window for |overlap = block_size / 2|. + float window[kBlockLength]; + std::fill(window, &window[kBlockLength], std::sqrt(0.5f)); + + LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kShiftAmount, + &noop); + float in_buffer[kChunkLength]; + float* in_chunk = in_buffer; + float out_buffer[kChunkLength]; + float* out_chunk = out_buffer; + + SetFloatArray(2.0f, 1, kChunkLength, &in_chunk); + SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk); + + trans.ProcessChunk(&in_chunk, &out_chunk); + + for (size_t i = 0; i < kChunkLength; ++i) { + ASSERT_NEAR(out_chunk[i], (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f, + 1e-5f); + } + + ASSERT_EQ(kChunkLength / kShiftAmount, noop.block_num()); +} + +TEST(LappedTransformTest, Callbacks) { + const size_t kChunkLength = 512; + const size_t kBlockLength = 64; + FftCheckerCallback call; + + // Rectangular window. + float window[kBlockLength]; + std::fill(window, &window[kBlockLength], 1.0f); + + LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kBlockLength, + &call); + float in_buffer[kChunkLength]; + float* in_chunk = in_buffer; + float out_buffer[kChunkLength]; + float* out_chunk = out_buffer; + + SetFloatArray(1.0f, 1, kChunkLength, &in_chunk); + SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk); + + trans.ProcessChunk(&in_chunk, &out_chunk); + + ASSERT_EQ(kChunkLength / kBlockLength, call.block_num()); +} + +TEST(LappedTransformTest, chunk_length) { + const size_t kBlockLength = 64; + FftCheckerCallback call; + const float window[kBlockLength] = {}; + + // Make sure that chunk_length returns the same value passed to the + // LappedTransform constructor. + { + const size_t kExpectedChunkLength = 512; + const LappedTransform trans(1, 1, kExpectedChunkLength, window, + kBlockLength, kBlockLength, &call); + + EXPECT_EQ(kExpectedChunkLength, trans.chunk_length()); + } + { + const size_t kExpectedChunkLength = 160; + const LappedTransform trans(1, 1, kExpectedChunkLength, window, + kBlockLength, kBlockLength, &call); + + EXPECT_EQ(kExpectedChunkLength, trans.chunk_length()); + } +} + +} // namespace webrtc |