Add support for WAV output in audioproc
The default output is a WAV file, except if the --pcm_output flag is set.
BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index e5204da..61edd8f 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -8,7 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <limits>
+
#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/wav_writer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
@@ -19,6 +23,64 @@
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
+class RawFile {
+ public:
+ RawFile(const std::string& filename)
+ : file_handle_(fopen(filename.c_str(), "wb")) {}
+
+ ~RawFile() {
+ fclose(file_handle_);
+ }
+
+ void WriteSamples(const int16_t* samples, size_t num_samples) {
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert samples to little-endian when writing to PCM file"
+#endif
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+ }
+
+ void WriteSamples(const float* samples, size_t num_samples) {
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+ }
+
+ private:
+ FILE* file_handle_;
+};
+
+static inline void WriteIntData(const int16_t* data,
+ size_t length,
+ WavFile* wav_file,
+ RawFile* raw_file) {
+ if (wav_file) {
+ wav_file->WriteSamples(data, length);
+ }
+ if (raw_file) {
+ raw_file->WriteSamples(data, length);
+ }
+}
+
+static inline void WriteFloatData(const float* const* data,
+ size_t samples_per_channel,
+ int num_channels,
+ WavFile* wav_file,
+ RawFile* raw_file) {
+ size_t length = num_channels * samples_per_channel;
+ scoped_ptr<float[]> buffer(new float[length]);
+ Interleave(data, samples_per_channel, num_channels, buffer.get());
+ if (raw_file) {
+ raw_file->WriteSamples(buffer.get(), length);
+ }
+ // TODO(aluebs): Use ScaleToInt16Range() from audio_util
+ for (size_t i = 0; i < length; ++i) {
+ buffer[i] = buffer[i] > 0 ?
+ buffer[i] * std::numeric_limits<int16_t>::max() :
+ -buffer[i] * std::numeric_limits<int16_t>::min();
+ }
+ if (wav_file) {
+ wav_file->WriteSamples(buffer.get(), length);
+ }
+}
+
// Exits on failure; do not use in unit tests.
static inline FILE* OpenFile(const std::string& filename, const char* mode) {
FILE* file = fopen(filename.c_str(), mode);