media/audio/wav_audio_handler.cc - chromium/src - Git at Google

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.

 #include "media/audio/wav_audio_handler.h"

 #include <algorithm>
 #include <cstring>

 #include "base/big_endian.h"
 #include "base/logging.h"
 #include "base/memory/ptr_util.h"
 #include "base/notreached.h"
 #include "base/sys_byteorder.h"
 #include "build/build_config.h"
 #include "media/base/audio_bus.h"
 #include "media/base/limits.h"

 namespace media {
 namespace {

 const char kChunkId[] = "RIFF";
 const char kFormat[] = "WAVE";
 const char kFmtSubchunkId[] = "fmt ";
 const char kDataSubchunkId[] = "data";

 // The size of a chunk header in wav file format. A chunk header consists of a
 // tag ('fmt ' or 'data') and 4 bytes of chunk length.
 const size_t kChunkHeaderSize = 8;

 // The minimum size of 'fmt' chunk.
 const size_t kFmtChunkMinimumSize = 16;
 const size_t kFmtChunkExtensibleMinimumSize = 40;

 // The offsets of 'fmt' fields.
 const size_t kAudioFormatOffset = 0;
 const size_t kChannelOffset = 2;
 const size_t kSampleRateOffset = 4;
 const size_t kBitsPerSampleOffset = 14;
 const size_t kValidBitsPerSampleOffset = 18;
 const size_t kSubFormatOffset = 24;

 // A convenience struct for passing WAV parameters around. AudioParameters is
 // too heavyweight for this. Keep this class internal to this implementation.
 struct WavAudioParameters {
   WavAudioHandler::AudioFormat audio_format;
   uint16_t num_channels;
   uint32_t sample_rate;
   uint16_t bits_per_sample;
   uint16_t valid_bits_per_sample;
   bool is_extensible;
 };

 bool ParamsAreValid(const WavAudioParameters& params) {
   return (
       // Check number of channels
       params.num_channels != 0u &&
       params.num_channels <= static_cast<uint16_t>(limits::kMaxChannels) &&
       // Check sample rate
       params.sample_rate != 0u &&
       (
           // Check bits per second for PCM data
           (params.audio_format ==
                WavAudioHandler::AudioFormat::kAudioFormatPCM &&
            (params.bits_per_sample == 8u || params.bits_per_sample == 16u ||
             params.bits_per_sample == 32u)) ||
           // Check bits per second for float data
           (params.audio_format ==
                WavAudioHandler::AudioFormat::kAudioFormatFloat &&
            (params.bits_per_sample == 32u || params.bits_per_sample == 64u))) &&
       // Check extensible format bps
       (!params.is_extensible ||
        params.valid_bits_per_sample == params.bits_per_sample));
 }

 // Reads an integer from |data| with |offset|.
 template <typename T>
 T ReadInt(const base::StringPiece& data, size_t offset) {
   CHECK_LE(offset + sizeof(T), data.size());
   T result;
   memcpy(&result, data.data() + offset, sizeof(T));
 #if !defined(ARCH_CPU_LITTLE_ENDIAN)
   result = base::ByteSwap(result);
 #endif
   return result;
 }

 // Parse a "fmt " chunk from wav data into its parameters.
 bool ParseFmtChunk(const base::StringPiece data, WavAudioParameters* params) {
   DCHECK(params);

   // If the chunk is too small, return false.
   if (data.size() < kFmtChunkMinimumSize) {
     LOG(ERROR) << "Data size " << data.size() << " is too short.";
     return false;
   }

   // Read in serialized parameters.
   params->audio_format =
       ReadInt<WavAudioHandler::AudioFormat>(data, kAudioFormatOffset);
   params->num_channels = ReadInt<uint16_t>(data, kChannelOffset);
   params->sample_rate = ReadInt<uint32_t>(data, kSampleRateOffset);
   params->bits_per_sample = ReadInt<uint16_t>(data, kBitsPerSampleOffset);

   if (params->audio_format ==
       WavAudioHandler::AudioFormat::kAudioFormatExtensible) {
     if (data.size() < kFmtChunkExtensibleMinimumSize) {
       LOG(ERROR) << "Data size " << data.size() << " is too short.";
       return false;
     }

     params->is_extensible = true;
     params->audio_format =
         ReadInt<WavAudioHandler::AudioFormat>(data, kSubFormatOffset);
     params->valid_bits_per_sample =
         ReadInt<uint16_t>(data, kValidBitsPerSampleOffset);
   } else {
     params->is_extensible = false;
   }
   return true;
 }

 bool ParseWavData(const base::StringPiece wav_data,
                   base::StringPiece* audio_data_out,
                   WavAudioParameters* params_out) {
   DCHECK(audio_data_out);
   DCHECK(params_out);

   // The header should look like: |R|I|F|F|1|2|3|4|W|A|V|E|
   base::BigEndianReader reader(wav_data.data(), wav_data.size());

   // Read the chunk ID and compare to "RIFF".
   base::StringPiece chunk_id;
   if (!reader.ReadPiece(&chunk_id, 4) || chunk_id != kChunkId) {
     DLOG(ERROR) << "missing or incorrect chunk ID in wav header";
     return false;
   }
   // The RIFF chunk length comes next, but we don't actually care what it says.
   if (!reader.Skip(sizeof(uint32_t))) {
     DLOG(ERROR) << "missing length in wav header";
     return false;
   }
   // Read format and compare to "WAVE".
   base::StringPiece format;
   if (!reader.ReadPiece(&format, 4) || format != kFormat) {
     DLOG(ERROR) << "missing or incorrect format ID in wav header";
     return false;
   }

   bool got_format = false;
   // If the number of remaining bytes is smaller than |kChunkHeaderSize|, it's
   // just junk at the end.
   while (reader.remaining() >= kChunkHeaderSize) {
     // We should be at the beginning of a subsection. The next 8 bytes are the
     // header and should look like: "|f|m|t| |1|2|3|4|" or "|d|a|t|a|1|2|3|4|".
     base::StringPiece chunk_fmt;
     uint32_t chunk_length;
     if (!reader.ReadPiece(&chunk_fmt, 4) || !reader.ReadU32(&chunk_length))
       break;
     chunk_length = base::ByteSwap(chunk_length);
     // Read |chunk_length| bytes of payload. If that is impossible, try to read
     // all remaining bytes as the payload.
     base::StringPiece chunk_payload;
     if (!reader.ReadPiece(&chunk_payload, chunk_length) &&
         !reader.ReadPiece(&chunk_payload, reader.remaining())) {
       break;
     }

     // Parse the subsection header, handling it if it is a "data" or "fmt "
     // chunk. Skip it otherwise.
     if (chunk_fmt == kFmtSubchunkId) {
       got_format = true;
       if (!ParseFmtChunk(chunk_payload, params_out))
         return false;
     } else if (chunk_fmt == kDataSubchunkId) {
       *audio_data_out = chunk_payload;
     } else {
       DVLOG(1) << "Skipping unknown data chunk: " << chunk_fmt << ".";
     }
   }

   // Check that data format has been read in and is valid.
   if (!got_format) {
     LOG(ERROR) << "Invalid: No \"" << kFmtSubchunkId << "\" header found!";
     return false;
   } else if (!ParamsAreValid(*params_out)) {
     LOG(ERROR) << "Format is invalid. "
                << "num_channels: " << params_out->num_channels << " "
                << "sample_rate: " << params_out->sample_rate << " "
                << "bits_per_sample: " << params_out->bits_per_sample << " "
                << "valid_bits_per_sample: " << params_out->valid_bits_per_sample
                << " "
                << "is_extensible: " << params_out->is_extensible;
     return false;
   }
   return true;
 }

 }  // namespace

 WavAudioHandler::WavAudioHandler(base::StringPiece audio_data,
                                  uint16_t num_channels,
                                  uint32_t sample_rate,
                                  uint16_t bits_per_sample,
                                  AudioFormat audio_format)
     : data_(audio_data),
       num_channels_(num_channels),
       sample_rate_(sample_rate),
       bits_per_sample_(bits_per_sample),
       audio_format_(audio_format) {
   DCHECK_NE(num_channels_, 0u);
   DCHECK_NE(sample_rate_, 0u);
   DCHECK_NE(bits_per_sample_, 0u);
   total_frames_ = data_.size() * 8 / num_channels_ / bits_per_sample_;
 }

 WavAudioHandler::~WavAudioHandler() = default;

 // static
 std::unique_ptr<WavAudioHandler> WavAudioHandler::Create(
     const base::StringPiece wav_data) {
   WavAudioParameters params;
   base::StringPiece audio_data;

   // Attempt to parse the WAV data.
   if (!ParseWavData(wav_data, &audio_data, &params))
     return nullptr;

   return base::WrapUnique(
       new WavAudioHandler(audio_data, params.num_channels, params.sample_rate,
                           params.bits_per_sample, params.audio_format));
 }

 bool WavAudioHandler::AtEnd(size_t cursor) const {
   return data_.size() <= cursor;
 }

 bool WavAudioHandler::CopyTo(AudioBus* bus,
                              size_t cursor,
                              size_t* bytes_written) const {
   if (!bus)
     return false;
   if (bus->channels() != num_channels_) {
     DVLOG(1) << "Number of channels mismatch.";
     return false;
   }
   if (AtEnd(cursor)) {
     bus->Zero();
     return true;
   }
   const int bytes_per_frame = num_channels_ * bits_per_sample_ / 8;
   const int remaining_frames = (data_.size() - cursor) / bytes_per_frame;
   const int frames = std::min(bus->frames(), remaining_frames);
   const auto* source = data_.data() + cursor;

   switch (audio_format_) {
     case AudioFormat::kAudioFormatPCM:
       switch (bits_per_sample_) {
         case 8:
           bus->FromInterleaved<UnsignedInt8SampleTypeTraits>(
               reinterpret_cast<const uint8_t*>(source), frames);
           break;
         case 16:
           bus->FromInterleaved<SignedInt16SampleTypeTraits>(
               reinterpret_cast<const int16_t*>(source), frames);
           break;
         case 32:
           bus->FromInterleaved<SignedInt32SampleTypeTraits>(
               reinterpret_cast<const int32_t*>(source), frames);
           break;
         default:
           NOTREACHED()
               << "Unsupported bytes per sample encountered for integer PCM: "
               << bytes_per_frame;
           bus->ZeroFrames(frames);
       }
       break;
     case AudioFormat::kAudioFormatFloat:
       switch (bits_per_sample_) {
         case 32:
           bus->FromInterleaved<Float32SampleTypeTraitsNoClip>(
               reinterpret_cast<const float*>(source), frames);
           break;
         case 64:
           bus->FromInterleaved<Float64SampleTypeTraits>(
               reinterpret_cast<const double*>(source), frames);
           break;
         default:
           NOTREACHED()
               << "Unsupported bytes per sample encountered for float PCM: "
               << bytes_per_frame;
           bus->ZeroFrames(frames);
       }
       break;
     default:
       NOTREACHED() << "Unsupported audio format encountered: "
                    << static_cast<uint16_t>(audio_format_);
       bus->ZeroFrames(frames);
   }

   *bytes_written = frames * bytes_per_frame;
   bus->ZeroFramesPartial(frames, bus->frames() - frames);
   return true;
 }

 base::TimeDelta WavAudioHandler::GetDuration() const {
   return base::TimeDelta::FromSecondsD(total_frames_ /
                                        static_cast<double>(sample_rate_));
 }

 }  // namespace media
	// Copyright 2013 The Chromium Authors. All rights reserved.
	// Use of this source code is governed by a BSD-style license that can be
	// found in the LICENSE file.

	#include "media/audio/wav_audio_handler.h"

	#include <algorithm>
	#include <cstring>

	#include "base/big_endian.h"
	#include "base/logging.h"
	#include "base/memory/ptr_util.h"
	#include "base/notreached.h"
	#include "base/sys_byteorder.h"
	#include "build/build_config.h"
	#include "media/base/audio_bus.h"
	#include "media/base/limits.h"

	namespace media {
	namespace {

	const char kChunkId[] = "RIFF";
	const char kFormat[] = "WAVE";
	const char kFmtSubchunkId[] = "fmt ";
	const char kDataSubchunkId[] = "data";

	// The size of a chunk header in wav file format. A chunk header consists of a
	// tag ('fmt ' or 'data') and 4 bytes of chunk length.
	const size_t kChunkHeaderSize = 8;

	// The minimum size of 'fmt' chunk.
	const size_t kFmtChunkMinimumSize = 16;
	const size_t kFmtChunkExtensibleMinimumSize = 40;

	// The offsets of 'fmt' fields.
	const size_t kAudioFormatOffset = 0;
	const size_t kChannelOffset = 2;
	const size_t kSampleRateOffset = 4;
	const size_t kBitsPerSampleOffset = 14;
	const size_t kValidBitsPerSampleOffset = 18;
	const size_t kSubFormatOffset = 24;

	// A convenience struct for passing WAV parameters around. AudioParameters is
	// too heavyweight for this. Keep this class internal to this implementation.
	struct WavAudioParameters {
	WavAudioHandler::AudioFormat audio_format;
	uint16_t num_channels;
	uint32_t sample_rate;
	uint16_t bits_per_sample;
	uint16_t valid_bits_per_sample;
	bool is_extensible;
	};

	bool ParamsAreValid(const WavAudioParameters& params) {
	return (
	// Check number of channels
	params.num_channels != 0u &&
	params.num_channels <= static_cast<uint16_t>(limits::kMaxChannels) &&
	// Check sample rate
	params.sample_rate != 0u &&
	(
	// Check bits per second for PCM data
	(params.audio_format ==
	WavAudioHandler::AudioFormat::kAudioFormatPCM &&
	(params.bits_per_sample == 8u \|\| params.bits_per_sample == 16u \|\|
	params.bits_per_sample == 32u)) \|\|
	// Check bits per second for float data
	(params.audio_format ==
	WavAudioHandler::AudioFormat::kAudioFormatFloat &&
	(params.bits_per_sample == 32u \|\| params.bits_per_sample == 64u))) &&
	// Check extensible format bps
	(!params.is_extensible \|\|
	params.valid_bits_per_sample == params.bits_per_sample));
	}

	// Reads an integer from \|data\| with \|offset\|.
	template <typename T>
	T ReadInt(const base::StringPiece& data, size_t offset) {
	CHECK_LE(offset + sizeof(T), data.size());
	T result;
	memcpy(&result, data.data() + offset, sizeof(T));
	#if !defined(ARCH_CPU_LITTLE_ENDIAN)
	result = base::ByteSwap(result);
	#endif
	return result;
	}

	// Parse a "fmt " chunk from wav data into its parameters.
	bool ParseFmtChunk(const base::StringPiece data, WavAudioParameters* params) {
	DCHECK(params);

	// If the chunk is too small, return false.
	if (data.size() < kFmtChunkMinimumSize) {
	LOG(ERROR) << "Data size " << data.size() << " is too short.";
	return false;
	}

	// Read in serialized parameters.
	params->audio_format =
	ReadInt<WavAudioHandler::AudioFormat>(data, kAudioFormatOffset);
	params->num_channels = ReadInt<uint16_t>(data, kChannelOffset);
	params->sample_rate = ReadInt<uint32_t>(data, kSampleRateOffset);
	params->bits_per_sample = ReadInt<uint16_t>(data, kBitsPerSampleOffset);

	if (params->audio_format ==
	WavAudioHandler::AudioFormat::kAudioFormatExtensible) {
	if (data.size() < kFmtChunkExtensibleMinimumSize) {
	LOG(ERROR) << "Data size " << data.size() << " is too short.";
	return false;
	}

	params->is_extensible = true;
	params->audio_format =
	ReadInt<WavAudioHandler::AudioFormat>(data, kSubFormatOffset);
	params->valid_bits_per_sample =
	ReadInt<uint16_t>(data, kValidBitsPerSampleOffset);
	} else {
	params->is_extensible = false;
	}
	return true;
	}

	bool ParseWavData(const base::StringPiece wav_data,
	base::StringPiece* audio_data_out,
	WavAudioParameters* params_out) {
	DCHECK(audio_data_out);
	DCHECK(params_out);

	// The header should look like: \|R\|I\|F\|F\|1\|2\|3\|4\|W\|A\|V\|E\|
	base::BigEndianReader reader(wav_data.data(), wav_data.size());

	// Read the chunk ID and compare to "RIFF".
	base::StringPiece chunk_id;
	if (!reader.ReadPiece(&chunk_id, 4) \|\| chunk_id != kChunkId) {
	DLOG(ERROR) << "missing or incorrect chunk ID in wav header";
	return false;
	}
	// The RIFF chunk length comes next, but we don't actually care what it says.
	if (!reader.Skip(sizeof(uint32_t))) {
	DLOG(ERROR) << "missing length in wav header";
	return false;
	}
	// Read format and compare to "WAVE".
	base::StringPiece format;
	if (!reader.ReadPiece(&format, 4) \|\| format != kFormat) {
	DLOG(ERROR) << "missing or incorrect format ID in wav header";
	return false;
	}

	bool got_format = false;
	// If the number of remaining bytes is smaller than \|kChunkHeaderSize\|, it's
	// just junk at the end.
	while (reader.remaining() >= kChunkHeaderSize) {
	// We should be at the beginning of a subsection. The next 8 bytes are the
	// header and should look like: "\|f\|m\|t\| \|1\|2\|3\|4\|" or "\|d\|a\|t\|a\|1\|2\|3\|4\|".
	base::StringPiece chunk_fmt;
	uint32_t chunk_length;
	if (!reader.ReadPiece(&chunk_fmt, 4) \|\| !reader.ReadU32(&chunk_length))
	break;
	chunk_length = base::ByteSwap(chunk_length);
	// Read \|chunk_length\| bytes of payload. If that is impossible, try to read
	// all remaining bytes as the payload.
	base::StringPiece chunk_payload;
	if (!reader.ReadPiece(&chunk_payload, chunk_length) &&
	!reader.ReadPiece(&chunk_payload, reader.remaining())) {
	break;
	}

	// Parse the subsection header, handling it if it is a "data" or "fmt "
	// chunk. Skip it otherwise.
	if (chunk_fmt == kFmtSubchunkId) {
	got_format = true;
	if (!ParseFmtChunk(chunk_payload, params_out))
	return false;
	} else if (chunk_fmt == kDataSubchunkId) {
	*audio_data_out = chunk_payload;
	} else {
	DVLOG(1) << "Skipping unknown data chunk: " << chunk_fmt << ".";
	}
	}

	// Check that data format has been read in and is valid.
	if (!got_format) {
	LOG(ERROR) << "Invalid: No \"" << kFmtSubchunkId << "\" header found!";
	return false;
	} else if (!ParamsAreValid(*params_out)) {
	LOG(ERROR) << "Format is invalid. "
	<< "num_channels: " << params_out->num_channels << " "
	<< "sample_rate: " << params_out->sample_rate << " "
	<< "bits_per_sample: " << params_out->bits_per_sample << " "
	<< "valid_bits_per_sample: " << params_out->valid_bits_per_sample
	<< " "
	<< "is_extensible: " << params_out->is_extensible;
	return false;
	}
	return true;
	}

	} // namespace

	WavAudioHandler::WavAudioHandler(base::StringPiece audio_data,
	uint16_t num_channels,
	uint32_t sample_rate,
	uint16_t bits_per_sample,
	AudioFormat audio_format)
	: data_(audio_data),
	num_channels_(num_channels),
	sample_rate_(sample_rate),
	bits_per_sample_(bits_per_sample),
	audio_format_(audio_format) {
	DCHECK_NE(num_channels_, 0u);
	DCHECK_NE(sample_rate_, 0u);
	DCHECK_NE(bits_per_sample_, 0u);
	total_frames_ = data_.size() * 8 / num_channels_ / bits_per_sample_;
	}

	WavAudioHandler::~WavAudioHandler() = default;

	// static
	std::unique_ptr<WavAudioHandler> WavAudioHandler::Create(
	const base::StringPiece wav_data) {
	WavAudioParameters params;
	base::StringPiece audio_data;

	// Attempt to parse the WAV data.
	if (!ParseWavData(wav_data, &audio_data, &params))
	return nullptr;

	return base::WrapUnique(
	new WavAudioHandler(audio_data, params.num_channels, params.sample_rate,
	params.bits_per_sample, params.audio_format));
	}

	bool WavAudioHandler::AtEnd(size_t cursor) const {
	return data_.size() <= cursor;
	}

	bool WavAudioHandler::CopyTo(AudioBus* bus,
	size_t cursor,
	size_t* bytes_written) const {
	if (!bus)
	return false;
	if (bus->channels() != num_channels_) {
	DVLOG(1) << "Number of channels mismatch.";
	return false;
	}
	if (AtEnd(cursor)) {
	bus->Zero();
	return true;
	}
	const int bytes_per_frame = num_channels_ * bits_per_sample_ / 8;
	const int remaining_frames = (data_.size() - cursor) / bytes_per_frame;
	const int frames = std::min(bus->frames(), remaining_frames);
	const auto* source = data_.data() + cursor;

	switch (audio_format_) {
	case AudioFormat::kAudioFormatPCM:
	switch (bits_per_sample_) {
	case 8:
	bus->FromInterleaved<UnsignedInt8SampleTypeTraits>(
	reinterpret_cast<const uint8_t*>(source), frames);
	break;
	case 16:
	bus->FromInterleaved<SignedInt16SampleTypeTraits>(
	reinterpret_cast<const int16_t*>(source), frames);
	break;
	case 32:
	bus->FromInterleaved<SignedInt32SampleTypeTraits>(
	reinterpret_cast<const int32_t*>(source), frames);
	break;
	default:
	NOTREACHED()
	<< "Unsupported bytes per sample encountered for integer PCM: "
	<< bytes_per_frame;
	bus->ZeroFrames(frames);
	}
	break;
	case AudioFormat::kAudioFormatFloat:
	switch (bits_per_sample_) {
	case 32:
	bus->FromInterleaved<Float32SampleTypeTraitsNoClip>(
	reinterpret_cast<const float*>(source), frames);
	break;
	case 64:
	bus->FromInterleaved<Float64SampleTypeTraits>(
	reinterpret_cast<const double*>(source), frames);
	break;
	default:
	NOTREACHED()
	<< "Unsupported bytes per sample encountered for float PCM: "
	<< bytes_per_frame;
	bus->ZeroFrames(frames);
	}
	break;
	default:
	NOTREACHED() << "Unsupported audio format encountered: "
	<< static_cast<uint16_t>(audio_format_);
	bus->ZeroFrames(frames);
	}

	bytes_written = frames bytes_per_frame;
	bus->ZeroFramesPartial(frames, bus->frames() - frames);
	return true;
	}

	base::TimeDelta WavAudioHandler::GetDuration() const {
	return base::TimeDelta::FromSecondsD(total_frames_ /
	static_cast<double>(sample_rate_));
	}

	} // namespace media