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libtgvoip/webrtc_dsp/common_audio/wav_header.cc
Grishka 5caaaafa42 Updated WebRTC APM
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-23 04:02:53 +03:00

300 lines
10 KiB
C++

/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Based on the WAV file format documentation at
// https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ and
// http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
#include "common_audio/wav_header.h"
#include <cstring>
#include <limits>
#include <string>
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/sanitizer.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
namespace {
struct ChunkHeader {
uint32_t ID;
uint32_t Size;
};
static_assert(sizeof(ChunkHeader) == 8, "ChunkHeader size");
struct RiffHeader {
ChunkHeader header;
uint32_t Format;
};
// We can't nest this definition in WavHeader, because VS2013 gives an error
// on sizeof(WavHeader::fmt): "error C2070: 'unknown': illegal sizeof operand".
struct FmtSubchunk {
ChunkHeader header;
uint16_t AudioFormat;
uint16_t NumChannels;
uint32_t SampleRate;
uint32_t ByteRate;
uint16_t BlockAlign;
uint16_t BitsPerSample;
};
static_assert(sizeof(FmtSubchunk) == 24, "FmtSubchunk size");
const uint32_t kFmtSubchunkSize = sizeof(FmtSubchunk) - sizeof(ChunkHeader);
// Simple wav header. It does not include chunks that are not essential to read
// audio samples.
struct WavHeader {
WavHeader(const WavHeader&) = default;
WavHeader& operator=(const WavHeader&) = default;
RiffHeader riff;
FmtSubchunk fmt;
struct {
ChunkHeader header;
} data;
};
static_assert(sizeof(WavHeader) == kWavHeaderSize, "no padding in header");
#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
static inline void WriteLE16(uint16_t* f, uint16_t x) {
*f = x;
}
static inline void WriteLE32(uint32_t* f, uint32_t x) {
*f = x;
}
static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
*f = static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 |
static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24;
}
static inline uint16_t ReadLE16(uint16_t x) {
return x;
}
static inline uint32_t ReadLE32(uint32_t x) {
return x;
}
static inline std::string ReadFourCC(uint32_t x) {
return std::string(reinterpret_cast<char*>(&x), 4);
}
#else
#error "Write be-to-le conversion functions"
#endif
static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
return static_cast<uint32_t>(bytes_in_payload + kWavHeaderSize -
sizeof(ChunkHeader));
}
static inline uint32_t ByteRate(size_t num_channels,
int sample_rate,
size_t bytes_per_sample) {
return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
}
static inline uint16_t BlockAlign(size_t num_channels,
size_t bytes_per_sample) {
return static_cast<uint16_t>(num_channels * bytes_per_sample);
}
// Finds a chunk having the sought ID. If found, then |readable| points to the
// first byte of the sought chunk data. If not found, the end of the file is
// reached.
void FindWaveChunk(ChunkHeader* chunk_header,
ReadableWav* readable,
const std::string sought_chunk_id) {
RTC_DCHECK_EQ(sought_chunk_id.size(), 4);
while (!readable->Eof()) {
if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
sizeof(*chunk_header))
return; // EOF.
if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
return; // Sought chunk found.
// Ignore current chunk by skipping its payload.
if (!readable->SeekForward(chunk_header->Size))
return; // EOF or error.
}
return; // EOF.
}
bool ReadFmtChunkData(FmtSubchunk* fmt_subchunk, ReadableWav* readable) {
// Reads "fmt " chunk payload.
if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtSubchunkSize) !=
kFmtSubchunkSize)
return false;
const uint32_t fmt_size = ReadLE32(fmt_subchunk->header.Size);
if (fmt_size != kFmtSubchunkSize) {
// There is an optional two-byte extension field permitted to be present
// with PCM, but which must be zero.
int16_t ext_size;
if (kFmtSubchunkSize + sizeof(ext_size) != fmt_size)
return false;
if (readable->Read(&ext_size, sizeof(ext_size)) != sizeof(ext_size))
return false;
if (ext_size != 0)
return false;
}
return true;
}
} // namespace
bool CheckWavParameters(size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
size_t num_samples) {
// num_channels, sample_rate, and bytes_per_sample must be positive, must fit
// in their respective fields, and their product must fit in the 32-bit
// ByteRate field.
if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
return false;
if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
return false;
if (num_channels > std::numeric_limits<uint16_t>::max())
return false;
if (static_cast<uint64_t>(bytes_per_sample) * 8 >
std::numeric_limits<uint16_t>::max())
return false;
if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample >
std::numeric_limits<uint32_t>::max())
return false;
// format and bytes_per_sample must agree.
switch (format) {
case kWavFormatPcm:
// Other values may be OK, but for now we're conservative:
if (bytes_per_sample != 1 && bytes_per_sample != 2)
return false;
break;
case kWavFormatALaw:
case kWavFormatMuLaw:
if (bytes_per_sample != 1)
return false;
break;
default:
return false;
}
// The number of bytes in the file, not counting the first ChunkHeader, must
// be less than 2^32; otherwise, the ChunkSize field overflows.
const size_t header_size = kWavHeaderSize - sizeof(ChunkHeader);
const size_t max_samples =
(std::numeric_limits<uint32_t>::max() - header_size) / bytes_per_sample;
if (num_samples > max_samples)
return false;
// Each channel must have the same number of samples.
if (num_samples % num_channels != 0)
return false;
return true;
}
void WriteWavHeader(uint8_t* buf,
size_t num_channels,
int sample_rate,
WavFormat format,
size_t bytes_per_sample,
size_t num_samples) {
RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
bytes_per_sample, num_samples));
auto header = rtc::MsanUninitialized<WavHeader>({});
const size_t bytes_in_payload = bytes_per_sample * num_samples;
WriteFourCC(&header.riff.header.ID, 'R', 'I', 'F', 'F');
WriteLE32(&header.riff.header.Size, RiffChunkSize(bytes_in_payload));
WriteFourCC(&header.riff.Format, 'W', 'A', 'V', 'E');
WriteFourCC(&header.fmt.header.ID, 'f', 'm', 't', ' ');
WriteLE32(&header.fmt.header.Size, kFmtSubchunkSize);
WriteLE16(&header.fmt.AudioFormat, format);
WriteLE16(&header.fmt.NumChannels, static_cast<uint16_t>(num_channels));
WriteLE32(&header.fmt.SampleRate, sample_rate);
WriteLE32(&header.fmt.ByteRate,
ByteRate(num_channels, sample_rate, bytes_per_sample));
WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
WriteLE16(&header.fmt.BitsPerSample,
static_cast<uint16_t>(8 * bytes_per_sample));
WriteFourCC(&header.data.header.ID, 'd', 'a', 't', 'a');
WriteLE32(&header.data.header.Size, static_cast<uint32_t>(bytes_in_payload));
// Do an extra copy rather than writing everything to buf directly, since buf
// might not be correctly aligned.
memcpy(buf, &header, kWavHeaderSize);
}
bool ReadWavHeader(ReadableWav* readable,
size_t* num_channels,
int* sample_rate,
WavFormat* format,
size_t* bytes_per_sample,
size_t* num_samples) {
auto header = rtc::MsanUninitialized<WavHeader>({});
// Read RIFF chunk.
if (readable->Read(&header.riff, sizeof(header.riff)) != sizeof(header.riff))
return false;
if (ReadFourCC(header.riff.header.ID) != "RIFF")
return false;
if (ReadFourCC(header.riff.Format) != "WAVE")
return false;
// Find "fmt " and "data" chunks. While the official Wave file specification
// does not put requirements on the chunks order, it is uncommon to find the
// "data" chunk before the "fmt " one. The code below fails if this is not the
// case.
FindWaveChunk(&header.fmt.header, readable, "fmt ");
if (ReadFourCC(header.fmt.header.ID) != "fmt ") {
RTC_LOG(LS_ERROR) << "Cannot find 'fmt ' chunk.";
return false;
}
if (!ReadFmtChunkData(&header.fmt, readable)) {
RTC_LOG(LS_ERROR) << "Cannot read 'fmt ' chunk.";
return false;
}
if (readable->Eof()) {
RTC_LOG(LS_ERROR) << "'fmt ' chunk placed after 'data' chunk.";
return false;
}
FindWaveChunk(&header.data.header, readable, "data");
if (ReadFourCC(header.data.header.ID) != "data") {
RTC_LOG(LS_ERROR) << "Cannot find 'data' chunk.";
return false;
}
// Parse needed fields.
*format = static_cast<WavFormat>(ReadLE16(header.fmt.AudioFormat));
*num_channels = ReadLE16(header.fmt.NumChannels);
*sample_rate = ReadLE32(header.fmt.SampleRate);
*bytes_per_sample = ReadLE16(header.fmt.BitsPerSample) / 8;
const size_t bytes_in_payload = ReadLE32(header.data.header.Size);
if (*bytes_per_sample == 0)
return false;
*num_samples = bytes_in_payload / *bytes_per_sample;
if (ReadLE32(header.riff.header.Size) < RiffChunkSize(bytes_in_payload))
return false;
if (ReadLE32(header.fmt.ByteRate) !=
ByteRate(*num_channels, *sample_rate, *bytes_per_sample))
return false;
if (ReadLE16(header.fmt.BlockAlign) !=
BlockAlign(*num_channels, *bytes_per_sample))
return false;
return CheckWavParameters(*num_channels, *sample_rate, *format,
*bytes_per_sample, *num_samples);
}
} // namespace webrtc