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d47941e018
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
86 lines
3.5 KiB
C++
86 lines
3.5 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/audio_processing/test/simulator_buffers.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz,
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int capture_input_sample_rate_hz,
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int render_output_sample_rate_hz,
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int capture_output_sample_rate_hz,
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size_t num_render_input_channels,
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size_t num_capture_input_channels,
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size_t num_render_output_channels,
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size_t num_capture_output_channels) {
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Random rand_gen(42);
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CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels,
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&rand_gen, &render_input_buffer, &render_input_config,
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&render_input, &render_input_samples);
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CreateConfigAndBuffer(render_output_sample_rate_hz,
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num_render_output_channels, &rand_gen,
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&render_output_buffer, &render_output_config,
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&render_output, &render_output_samples);
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CreateConfigAndBuffer(capture_input_sample_rate_hz,
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num_capture_input_channels, &rand_gen,
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&capture_input_buffer, &capture_input_config,
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&capture_input, &capture_input_samples);
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CreateConfigAndBuffer(capture_output_sample_rate_hz,
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num_capture_output_channels, &rand_gen,
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&capture_output_buffer, &capture_output_config,
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&capture_output, &capture_output_samples);
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UpdateInputBuffers();
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}
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SimulatorBuffers::~SimulatorBuffers() = default;
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void SimulatorBuffers::CreateConfigAndBuffer(
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int sample_rate_hz,
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size_t num_channels,
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Random* rand_gen,
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std::unique_ptr<AudioBuffer>* buffer,
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StreamConfig* config,
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std::vector<float*>* buffer_data,
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std::vector<float>* buffer_data_samples) {
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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*config = StreamConfig(sample_rate_hz, num_channels, false);
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buffer->reset(
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new AudioBuffer(config->sample_rate_hz(), config->num_channels(),
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config->sample_rate_hz(), config->num_channels(),
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config->sample_rate_hz(), config->num_channels()));
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buffer_data_samples->resize(samples_per_channel * num_channels);
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for (auto& v : *buffer_data_samples) {
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v = rand_gen->Rand<float>();
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}
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buffer_data->resize(num_channels);
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for (size_t ch = 0; ch < num_channels; ++ch) {
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(*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
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}
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}
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void SimulatorBuffers::UpdateInputBuffers() {
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test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
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capture_input_buffer.get());
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test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
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render_input_buffer.get());
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}
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} // namespace test
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} // namespace webrtc
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