Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(942)

Side by Side Diff: webrtc/common_audio/wav_file.cc

Issue 1534193008: Misc. small cleanups (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Unnecessary parens Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/common_audio/wav_file.h ('k') | webrtc/common_audio/wav_file_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "webrtc/common_audio/wav_file.h" 11 #include "webrtc/common_audio/wav_file.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <cstdio> 14 #include <cstdio>
15 #include <limits> 15 #include <limits>
16 #include <sstream> 16 #include <sstream>
17 17
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/safe_conversions.h" 19 #include "webrtc/base/safe_conversions.h"
20 #include "webrtc/common_audio/include/audio_util.h" 20 #include "webrtc/common_audio/include/audio_util.h"
21 #include "webrtc/common_audio/wav_header.h" 21 #include "webrtc/common_audio/wav_header.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 // We write 16-bit PCM WAV files. 25 // We write 16-bit PCM WAV files.
26 static const WavFormat kWavFormat = kWavFormatPcm; 26 static const WavFormat kWavFormat = kWavFormatPcm;
27 static const int kBytesPerSample = 2; 27 static const size_t kBytesPerSample = 2;
28 28
29 // Doesn't take ownership of the file handle and won't close it. 29 // Doesn't take ownership of the file handle and won't close it.
30 class ReadableWavFile : public ReadableWav { 30 class ReadableWavFile : public ReadableWav {
31 public: 31 public:
32 explicit ReadableWavFile(FILE* file) : file_(file) {} 32 explicit ReadableWavFile(FILE* file) : file_(file) {}
33 virtual size_t Read(void* buf, size_t num_bytes) { 33 virtual size_t Read(void* buf, size_t num_bytes) {
34 return fread(buf, 1, num_bytes, file_); 34 return fread(buf, 1, num_bytes, file_);
35 } 35 }
36 36
37 private: 37 private:
38 FILE* file_; 38 FILE* file_;
39 }; 39 };
40 40
41 std::string WavFile::FormatAsString() const { 41 std::string WavFile::FormatAsString() const {
42 std::ostringstream s; 42 std::ostringstream s;
43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels() 43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
44 << ", Duration: " 44 << ", Duration: "
45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s"; 45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
46 return s.str(); 46 return s.str();
47 } 47 }
48 48
49 WavReader::WavReader(const std::string& filename) 49 WavReader::WavReader(const std::string& filename)
50 : file_handle_(fopen(filename.c_str(), "rb")) { 50 : file_handle_(fopen(filename.c_str(), "rb")) {
51 RTC_CHECK(file_handle_) << "Could not open wav file for reading."; 51 RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
52 52
53 ReadableWavFile readable(file_handle_); 53 ReadableWavFile readable(file_handle_);
54 WavFormat format; 54 WavFormat format;
55 int bytes_per_sample; 55 size_t bytes_per_sample;
56 RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format, 56 RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
57 &bytes_per_sample, &num_samples_)); 57 &bytes_per_sample, &num_samples_));
58 num_samples_remaining_ = num_samples_; 58 num_samples_remaining_ = num_samples_;
59 RTC_CHECK_EQ(kWavFormat, format); 59 RTC_CHECK_EQ(kWavFormat, format);
60 RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample); 60 RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample);
61 } 61 }
62 62
63 WavReader::~WavReader() { 63 WavReader::~WavReader() {
64 Close(); 64 Close();
65 } 65 }
66 66
67 size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) { 67 size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
68 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN 68 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
69 #error "Need to convert samples to big-endian when reading from WAV file" 69 #error "Need to convert samples to big-endian when reading from WAV file"
70 #endif 70 #endif
71 // There could be metadata after the audio; ensure we don't read it. 71 // There could be metadata after the audio; ensure we don't read it.
72 num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples), 72 num_samples = std::min(num_samples, num_samples_remaining_);
73 num_samples_remaining_);
74 const size_t read = 73 const size_t read =
75 fread(samples, sizeof(*samples), num_samples, file_handle_); 74 fread(samples, sizeof(*samples), num_samples, file_handle_);
76 // If we didn't read what was requested, ensure we've reached the EOF. 75 // If we didn't read what was requested, ensure we've reached the EOF.
77 RTC_CHECK(read == num_samples || feof(file_handle_)); 76 RTC_CHECK(read == num_samples || feof(file_handle_));
78 RTC_CHECK_LE(read, num_samples_remaining_); 77 RTC_CHECK_LE(read, num_samples_remaining_);
79 num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read); 78 num_samples_remaining_ -= read;
80 return read; 79 return read;
81 } 80 }
82 81
83 size_t WavReader::ReadSamples(size_t num_samples, float* samples) { 82 size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
84 static const size_t kChunksize = 4096 / sizeof(uint16_t); 83 static const size_t kChunksize = 4096 / sizeof(uint16_t);
85 size_t read = 0; 84 size_t read = 0;
86 for (size_t i = 0; i < num_samples; i += kChunksize) { 85 for (size_t i = 0; i < num_samples; i += kChunksize) {
87 int16_t isamples[kChunksize]; 86 int16_t isamples[kChunksize];
88 size_t chunk = std::min(kChunksize, num_samples - i); 87 size_t chunk = std::min(kChunksize, num_samples - i);
89 chunk = ReadSamples(chunk, isamples); 88 chunk = ReadSamples(chunk, isamples);
(...skipping 29 matching lines...) Expand all
119 Close(); 118 Close();
120 } 119 }
121 120
122 void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) { 121 void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
123 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN 122 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
124 #error "Need to convert samples to little-endian when writing to WAV file" 123 #error "Need to convert samples to little-endian when writing to WAV file"
125 #endif 124 #endif
126 const size_t written = 125 const size_t written =
127 fwrite(samples, sizeof(*samples), num_samples, file_handle_); 126 fwrite(samples, sizeof(*samples), num_samples, file_handle_);
128 RTC_CHECK_EQ(num_samples, written); 127 RTC_CHECK_EQ(num_samples, written);
129 num_samples_ += static_cast<uint32_t>(written); 128 num_samples_ += written;
130 RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() || 129 RTC_CHECK(num_samples_ >= written); // detect size_t overflow
131 num_samples_ >= written); // detect uint32_t overflow
132 } 130 }
133 131
134 void WavWriter::WriteSamples(const float* samples, size_t num_samples) { 132 void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
135 static const size_t kChunksize = 4096 / sizeof(uint16_t); 133 static const size_t kChunksize = 4096 / sizeof(uint16_t);
136 for (size_t i = 0; i < num_samples; i += kChunksize) { 134 for (size_t i = 0; i < num_samples; i += kChunksize) {
137 int16_t isamples[kChunksize]; 135 int16_t isamples[kChunksize];
138 const size_t chunk = std::min(kChunksize, num_samples - i); 136 const size_t chunk = std::min(kChunksize, num_samples - i);
139 FloatS16ToS16(samples + i, chunk, isamples); 137 FloatS16ToS16(samples + i, chunk, isamples);
140 WriteSamples(isamples, chunk); 138 WriteSamples(isamples, chunk);
141 } 139 }
(...skipping 29 matching lines...) Expand all
171 } 169 }
172 170
173 int rtc_WavSampleRate(const rtc_WavWriter* wf) { 171 int rtc_WavSampleRate(const rtc_WavWriter* wf) {
174 return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate(); 172 return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
175 } 173 }
176 174
177 int rtc_WavNumChannels(const rtc_WavWriter* wf) { 175 int rtc_WavNumChannels(const rtc_WavWriter* wf) {
178 return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels(); 176 return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
179 } 177 }
180 178
181 uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf) { 179 size_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
182 return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples(); 180 return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
183 } 181 }
OLDNEW
« no previous file with comments | « webrtc/common_audio/wav_file.h ('k') | webrtc/common_audio/wav_file_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698