Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

156 lines
4.6 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
#include <math.h>
#include <iterator>
#include <limits>
#include <string>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
class RawFile final {
public:
explicit RawFile(const std::string& filename);
~RawFile();
void WriteSamples(const int16_t* samples, size_t num_samples);
void WriteSamples(const float* samples, size_t num_samples);
private:
FILE* file_handle_;
RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
};
// Reads ChannelBuffers from a provided WavReader.
class ChannelBufferWavReader final {
public:
explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
// Reads data from the file according to the |buffer| format. Returns false if
// a full buffer can't be read from the file.
bool Read(ChannelBuffer<float>* buffer);
private:
rtc::scoped_ptr<WavReader> file_;
std::vector<float> interleaved_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
};
// Writes ChannelBuffers to a provided WavWriter.
class ChannelBufferWavWriter final {
public:
explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
void Write(const ChannelBuffer<float>& buffer);
private:
rtc::scoped_ptr<WavWriter> file_;
std::vector<float> interleaved_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
};
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
RawFile* raw_file);
void WriteFloatData(const float* const* data,
int samples_per_channel,
int num_channels,
WavWriter* wav_file,
RawFile* raw_file);
// Exits on failure; do not use in unit tests.
FILE* OpenFile(const std::string& filename, const char* mode);
int SamplesFromRate(int rate);
void SetFrameSampleRate(AudioFrame* frame,
int sample_rate_hz);
template <typename T>
void SetContainerFormat(int sample_rate_hz,
int num_channels,
AudioFrame* frame,
rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
SetFrameSampleRate(frame, sample_rate_hz);
frame->num_channels_ = num_channels;
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
}
AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
template <typename T>
float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (int i = 0; i < length; ++i) {
T error = ref[i] - test[i];
mse += error * error;
*variance += ref[i] * ref[i];
mean += ref[i];
}
mse /= length;
*variance /= length;
mean /= length;
*variance -= mean * mean;
float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(*variance / mse);
return snr;
}
// Returns a vector<T> parsed from whitespace delimited values in to_parse,
// or an empty vector if the string could not be parsed.
template<typename T>
std::vector<T> ParseList(const std::string& to_parse) {
std::vector<T> values;
std::istringstream str(to_parse);
std::copy(
std::istream_iterator<T>(str),
std::istream_iterator<T>(),
std::back_inserter(values));
return values;
}
// Parses the array geometry from the command line.
//
// If a vector with size != num_mics is returned, an error has occurred and an
// appropriate error message has been printed to stdout.
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
size_t num_mics);
// Same as above, but without the num_mics check for when it isn't available.
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_