1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/audio_processing/debug.pb.h"
12 #include "webrtc/modules/audio_processing/common.h"
13 #include "webrtc/modules/audio_processing/include/audio_processing.h"
14 #include "webrtc/modules/interface/module_common_types.h"
15 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
16
17 namespace webrtc {
18
19 static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
20 #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
21
22 // Exits on failure; do not use in unit tests.
OpenFile(const std::string & filename,const char * mode)23 static inline FILE* OpenFile(const std::string& filename, const char* mode) {
24 FILE* file = fopen(filename.c_str(), mode);
25 if (!file) {
26 printf("Unable to open file %s\n", filename.c_str());
27 exit(1);
28 }
29 return file;
30 }
31
SamplesFromRate(int rate)32 static inline int SamplesFromRate(int rate) {
33 return AudioProcessing::kChunkSizeMs * rate / 1000;
34 }
35
SetFrameSampleRate(AudioFrame * frame,int sample_rate_hz)36 static inline void SetFrameSampleRate(AudioFrame* frame,
37 int sample_rate_hz) {
38 frame->sample_rate_hz_ = sample_rate_hz;
39 frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
40 sample_rate_hz / 1000;
41 }
42
43 template <typename T>
SetContainerFormat(int sample_rate_hz,int num_channels,AudioFrame * frame,scoped_ptr<ChannelBuffer<T>> * cb)44 void SetContainerFormat(int sample_rate_hz,
45 int num_channels,
46 AudioFrame* frame,
47 scoped_ptr<ChannelBuffer<T> >* cb) {
48 SetFrameSampleRate(frame, sample_rate_hz);
49 frame->num_channels_ = num_channels;
50 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
51 }
52
LayoutFromChannels(int num_channels)53 static inline AudioProcessing::ChannelLayout LayoutFromChannels(
54 int num_channels) {
55 switch (num_channels) {
56 case 1:
57 return AudioProcessing::kMono;
58 case 2:
59 return AudioProcessing::kStereo;
60 default:
61 assert(false);
62 return AudioProcessing::kMono;
63 }
64 }
65
66 // Allocates new memory in the scoped_ptr to fit the raw message and returns the
67 // number of bytes read.
ReadMessageBytesFromFile(FILE * file,scoped_ptr<uint8_t[]> * bytes)68 static inline size_t ReadMessageBytesFromFile(FILE* file,
69 scoped_ptr<uint8_t[]>* bytes) {
70 // The "wire format" for the size is little-endian. Assume we're running on
71 // a little-endian machine.
72 int32_t size = 0;
73 if (fread(&size, sizeof(size), 1, file) != 1)
74 return 0;
75 if (size <= 0)
76 return 0;
77
78 bytes->reset(new uint8_t[size]);
79 return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
80 }
81
82 // Returns true on success, false on error or end-of-file.
ReadMessageFromFile(FILE * file,::google::protobuf::MessageLite * msg)83 static inline bool ReadMessageFromFile(FILE* file,
84 ::google::protobuf::MessageLite* msg) {
85 scoped_ptr<uint8_t[]> bytes;
86 size_t size = ReadMessageBytesFromFile(file, &bytes);
87 if (!size)
88 return false;
89
90 msg->Clear();
91 return msg->ParseFromArray(bytes.get(), size);
92 }
93
94 } // namespace webrtc
95