New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.
This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.
Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.
R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
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/*
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* Copyright (c) 2014 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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#ifndef WEBRTC_COMMON_AUDIO_WAV_WRITER_H_
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#define WEBRTC_COMMON_AUDIO_WAV_WRITER_H_
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#ifdef __cplusplus
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#include <stdint.h>
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#include <cstddef>
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#include <string>
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namespace webrtc {
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// Simple C++ class for writing 16-bit PCM WAV files. All error handling is
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Add CHECK and friends from Chromium.
Replace FATAL_ERROR_IF with the more familiar (to Chromium developers)
CHECK and DCHECK. The full Chromium implementation is fairly elaborate
but I copied enough to get us most of the benefits. I believe the main
missing component is a more advanced stack dump. For this bit I relied
on the V8 implementation.
There are a few minor modifications from the Chromium original:
- The FatalMessage class is specialized for logging fatal error
messages and aborting. Chromium uses the general LogMessage class,
which we could consider moving towards in the future.
- NOTIMPLEMENTED() and NOTREACHED() have been removed, partly because
I don't want to rely on our logging.h until base/ and system_wrappers/
are consolidated.
- FATAL() replaces LOG(FATAL).
Minor modifications from V8's stack dump:
- If parsing of a stack trace symbol fails, just print the unparsed
symbol. (I noticed this happened on Mac.)
- Use __GLIBCXX__ and __UCLIBC__. This is from examining the backtrace
use in Chromium.
UNREACHABLE() has been removed because its behavior is different than
Chromium's NOTREACHED(), which is bound to cause confusion. The few uses
were replaced with FATAL(), matching the previous behavior.
Add a NO_RETURN macro, allowing us to remove unreachable return
statements following a CHECK/FATAL.
TESTED=the addition of dummy CHECK, DCHECK, CHECK_EQ and FATAL did the
did the right things. Stack traces work on Mac, but I don't get symbols
on Linux.
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7003 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 16:28:26 +00:00
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// by calls to CHECK(), making it unsuitable for anything but debug code.
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New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.
This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.
Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.
R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
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class WavFile {
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public:
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// Open a new WAV file for writing.
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WavFile(const std::string& filename, int sample_rate, int num_channels);
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// Close the WAV file, after writing its header.
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~WavFile();
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// Write additional samples to the file. Each sample is in the range
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// [-32768,32767], and there must be the previously specified number of
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// interleaved channels.
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void WriteSamples(const float* samples, size_t num_samples);
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2014-09-02 07:51:51 +00:00
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void WriteSamples(const int16_t* samples, size_t num_samples);
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New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.
This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.
Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.
R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
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2014-08-25 06:26:04 +00:00
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int sample_rate() const { return sample_rate_; }
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int num_channels() const { return num_channels_; }
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uint32_t num_samples() const { return num_samples_; }
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New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.
This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.
Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.
R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
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private:
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void Close();
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const int sample_rate_;
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const int num_channels_;
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uint32_t num_samples_; // total number of samples written to file
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FILE* file_handle_; // output file, owned by this class
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};
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} // namespace webrtc
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extern "C" {
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#endif // __cplusplus
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// C wrappers for the WavFile class.
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typedef struct rtc_WavFile rtc_WavFile;
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rtc_WavFile* rtc_WavOpen(const char* filename,
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int sample_rate,
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int num_channels);
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void rtc_WavClose(rtc_WavFile* wf);
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void rtc_WavWriteSamples(rtc_WavFile* wf,
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const float* samples,
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size_t num_samples);
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2014-08-25 06:26:04 +00:00
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int rtc_WavSampleRate(const rtc_WavFile* wf);
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int rtc_WavNumChannels(const rtc_WavFile* wf);
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uint32_t rtc_WavNumSamples(const rtc_WavFile* wf);
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New utility class for easy debug dumping to WAV files
There are currently a number of places in the code where we dump audio
data in various stages of processing for debug purposes. Currently
these all write raw, uncompressed PCM files, which isn't supported by
the most common audio players, and requires the user to supply
metadata such as sample rate, sample size and endianness, etc.
This patch adds a simple class that makes it easy to write WAV files
instead. WAV files still contain the same uncompressed PCM data, but
they have a small header that contains all the requisite metadata, and
are supported by virtually all audio players.
Since some of the debug code that will be writing WAV files is written
in plain C, a C API is included as well.
R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
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#ifdef __cplusplus
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} // extern "C"
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#endif
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#endif // WEBRTC_COMMON_AUDIO_WAV_WRITER_H_
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