
This adds functionality from audioproc_float. The geometry parsing code is now shared from test_utils.h. I removed the "mic_spacing" flag from audioproc_float because it's a redundancy that I suspect isn't very useful. Includes a cleanup of the audio_processing test utils. They're now packaged in targets, with the protobuf-using ones split out to avoid requiring users to depend on protobufs. pcm_utils is no longer needed and removed. The primary motivation for this CL is that AudioProcessing currently doesn't support more than two channels and we'd like a way to pass more channels to the beamformer. R=aluebs@webrtc.org, mgraczyk@chromium.org Review URL: https://webrtc-codereview.appspot.com/50899004 Cr-Commit-Position: refs/heads/master@{#9157}
43 lines
1.3 KiB
C++
43 lines
1.3 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/modules/audio_processing/test/protobuf_utils.h"
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namespace webrtc {
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size_t ReadMessageBytesFromFile(FILE* file, rtc::scoped_ptr<uint8_t[]>* bytes) {
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// The "wire format" for the size is little-endian. Assume we're running on
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// a little-endian machine.
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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#error "Need to convert messsage from little-endian."
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#endif
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int32_t size = 0;
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if (fread(&size, sizeof(size), 1, file) != 1)
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return 0;
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if (size <= 0)
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return 0;
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bytes->reset(new uint8_t[size]);
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return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
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}
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// Returns true on success, false on error or end-of-file.
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bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg) {
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rtc::scoped_ptr<uint8_t[]> bytes;
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size_t size = ReadMessageBytesFromFile(file, &bytes);
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if (!size)
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return false;
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msg->Clear();
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return msg->ParseFromArray(bytes.get(), size);
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}
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} // namespace webrtc
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