
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
91 lines
2.9 KiB
C++
91 lines
2.9 KiB
C++
/*
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* Copyright (c) 2014 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/modules/audio_processing/beamformer/pcm_utils.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/channel_buffer.h"
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namespace webrtc {
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size_t PcmRead(FILE* file,
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size_t length,
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int num_channels,
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int16_t* const* buffer) {
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CHECK_GE(num_channels, 1);
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scoped_ptr<int16_t[]> interleaved_buffer(new int16_t[length]);
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size_t elements_read = fread(interleaved_buffer.get(), sizeof(int16_t),
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length, file);
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if (elements_read != length) {
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// This is only an error if we haven't reached the end of the file.
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CHECK_NE(0, feof(file));
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}
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Deinterleave(interleaved_buffer.get(),
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static_cast<int>(elements_read) / num_channels,
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num_channels,
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buffer);
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return elements_read;
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}
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size_t PcmReadToFloat(FILE* file,
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size_t length,
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int num_channels,
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float* const* buffer) {
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CHECK_GE(num_channels, 1);
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int num_frames = static_cast<int>(length) / num_channels;
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scoped_ptr<ChannelBuffer<int16_t> > deinterleaved_buffer(
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new ChannelBuffer<int16_t>(num_frames, num_channels));
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size_t elements_read =
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PcmRead(file, length, num_channels, deinterleaved_buffer->channels());
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for (int i = 0; i < num_channels; ++i) {
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S16ToFloat(deinterleaved_buffer->channels()[i], num_frames, buffer[i]);
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}
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return elements_read;
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}
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void PcmWrite(FILE* file,
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size_t length,
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int num_channels,
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const int16_t* const* buffer) {
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CHECK_GE(num_channels, 1);
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scoped_ptr<int16_t[]> interleaved_buffer(new int16_t[length]);
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Interleave(buffer,
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static_cast<int>(length) / num_channels,
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num_channels,
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interleaved_buffer.get());
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CHECK_EQ(length,
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fwrite(interleaved_buffer.get(), sizeof(int16_t), length, file));
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}
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void PcmWriteFromFloat(FILE* file,
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size_t length,
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int num_channels,
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const float* const* buffer) {
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CHECK_GE(num_channels, 1);
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int num_frames = static_cast<int>(length) / num_channels;
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scoped_ptr<ChannelBuffer<int16_t> > deinterleaved_buffer(
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new ChannelBuffer<int16_t>(num_frames, num_channels));
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for (int i = 0; i < num_channels; ++i) {
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FloatToS16(buffer[i], num_frames, deinterleaved_buffer->channels()[i]);
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}
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PcmWrite(file, length, num_channels, deinterleaved_buffer->channels());
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}
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} // namespace webrtc
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