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
diff --git a/webrtc/common_audio/wav_writer_unittest.cc b/webrtc/common_audio/wav_writer_unittest.cc
new file mode 100644
index 0000000..9efe96e
--- /dev/null
+++ b/webrtc/common_audio/wav_writer_unittest.cc
@@ -0,0 +1,124 @@
+/*
+ * 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.
+ */
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+
+#include <cmath>
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/compile_assert.h"
+#include "webrtc/common_audio/wav_header.h"
+#include "webrtc/common_audio/wav_writer.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+static const float kSamples[] = {0.0, 10.0, 4e4, -1e9};
+
+// Write a tiny WAV file with the C++ interface and verify the result.
+TEST(WavWriterTest, CPP) {
+ const std::string outfile = webrtc::test::OutputPath() + "wavtest1.wav";
+ static const int kNumSamples = 3;
+ {
+ webrtc::WavFile w(outfile, 14099, 1);
+ w.WriteSamples(kSamples, kNumSamples);
+ }
+ static const uint8_t kExpectedContents[] = {
+ 'R', 'I', 'F', 'F',
+ 42, 0, 0, 0, // size of whole file - 8: 6 + 44 - 8
+ 'W', 'A', 'V', 'E',
+ 'f', 'm', 't', ' ',
+ 16, 0, 0, 0, // size of fmt block - 8: 24 - 8
+ 1, 0, // format: PCM (1)
+ 1, 0, // channels: 1
+ 0x13, 0x37, 0, 0, // sample rate: 14099
+ 0x26, 0x6e, 0, 0, // byte rate: 2 * 14099
+ 2, 0, // block align: NumChannels * BytesPerSample
+ 16, 0, // bits per sample: 2 * 8
+ 'd', 'a', 't', 'a',
+ 6, 0, 0, 0, // size of payload: 6
+ 0, 0, // first sample: 0.0
+ 10, 0, // second sample: 10.0
+ 0xff, 0x7f, // third sample: 4e4 (saturated)
+ };
+ static const int kContentSize =
+ webrtc::kWavHeaderSize + kNumSamples * sizeof(int16_t);
+ COMPILE_ASSERT(sizeof(kExpectedContents) == kContentSize, content_size);
+ EXPECT_EQ(size_t(kContentSize), webrtc::test::GetFileSize(outfile));
+ FILE* f = fopen(outfile.c_str(), "rb");
+ ASSERT_TRUE(f);
+ uint8_t contents[kContentSize];
+ ASSERT_EQ(1u, fread(contents, kContentSize, 1, f));
+ EXPECT_EQ(0, fclose(f));
+ EXPECT_EQ(0, memcmp(kExpectedContents, contents, kContentSize));
+}
+
+// Write a tiny WAV file with the C interface and verify the result.
+TEST(WavWriterTest, C) {
+ const std::string outfile = webrtc::test::OutputPath() + "wavtest2.wav";
+ rtc_WavFile *w = rtc_WavOpen(outfile.c_str(), 11904, 2);
+ static const int kNumSamples = 4;
+ rtc_WavWriteSamples(w, &kSamples[0], 2);
+ rtc_WavWriteSamples(w, &kSamples[2], kNumSamples - 2);
+ rtc_WavClose(w);
+ static const uint8_t kExpectedContents[] = {
+ 'R', 'I', 'F', 'F',
+ 44, 0, 0, 0, // size of whole file - 8: 8 + 44 - 8
+ 'W', 'A', 'V', 'E',
+ 'f', 'm', 't', ' ',
+ 16, 0, 0, 0, // size of fmt block - 8: 24 - 8
+ 1, 0, // format: PCM (1)
+ 2, 0, // channels: 2
+ 0x80, 0x2e, 0, 0, // sample rate: 11904
+ 0, 0xba, 0, 0, // byte rate: 2 * 2 * 11904
+ 4, 0, // block align: NumChannels * BytesPerSample
+ 16, 0, // bits per sample: 2 * 8
+ 'd', 'a', 't', 'a',
+ 8, 0, 0, 0, // size of payload: 8
+ 0, 0, // first sample: 0.0
+ 10, 0, // second sample: 10.0
+ 0xff, 0x7f, // third sample: 4e4 (saturated)
+ 0, 0x80, // fourth sample: -1e9 (saturated)
+ };
+ static const int kContentSize =
+ webrtc::kWavHeaderSize + kNumSamples * sizeof(int16_t);
+ COMPILE_ASSERT(sizeof(kExpectedContents) == kContentSize, content_size);
+ EXPECT_EQ(size_t(kContentSize), webrtc::test::GetFileSize(outfile));
+ FILE* f = fopen(outfile.c_str(), "rb");
+ ASSERT_TRUE(f);
+ uint8_t contents[kContentSize];
+ ASSERT_EQ(1u, fread(contents, kContentSize, 1, f));
+ EXPECT_EQ(0, fclose(f));
+ EXPECT_EQ(0, memcmp(kExpectedContents, contents, kContentSize));
+}
+
+// Write a larger WAV file. You can listen to this file to sanity-check it.
+TEST(WavWriterTest, LargeFile) {
+ std::string outfile = webrtc::test::OutputPath() + "wavtest3.wav";
+ static const int kSampleRate = 8000;
+ static const int kNumChannels = 2;
+ static const int kNumSamples = 3 * kSampleRate * kNumChannels;
+ float samples[kNumSamples];
+ for (int i = 0; i < kNumSamples; i += kNumChannels) {
+ // A nice periodic beeping sound.
+ static const double kToneHz = 440;
+ const double t = static_cast<double>(i) / (kNumChannels * kSampleRate);
+ const double x =
+ std::numeric_limits<int16_t>::max() * std::sin(t * kToneHz * 2 * M_PI);
+ samples[i] = std::pow(std::sin(t * 2 * 2 * M_PI), 10) * x;
+ samples[i + 1] = std::pow(std::cos(t * 2 * 2 * M_PI), 10) * x;
+ }
+ {
+ webrtc::WavFile w(outfile, kSampleRate, kNumChannels);
+ w.WriteSamples(samples, kNumSamples);
+ }
+ EXPECT_EQ(sizeof(int16_t) * kNumSamples + webrtc::kWavHeaderSize,
+ webrtc::test::GetFileSize(outfile));
+}
