audio_converter_unittest.cc File Reference

#include <cmath>
#include <algorithm>
#include <memory>
#include <vector>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/common_audio/audio_converter.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/test/gtest.h"

Namespaces
	webrtc

Typedefs
typedef std::unique_ptr< ChannelBuffer< float > >	webrtc::ScopedBuffer

Functions
ScopedBuffer	webrtc::CreateBuffer (const std::vector< float > &data, size_t frames)
void	webrtc::VerifyParams (const ChannelBuffer< float > &ref, const ChannelBuffer< float > &test)
float	webrtc::ComputeSNR (const ChannelBuffer< float > &ref, const ChannelBuffer< float > &test, size_t expected_delay)
void	webrtc::RunAudioConverterTest (size_t src_channels, int src_sample_rate_hz, size_t dst_channels, int dst_sample_rate_hz)
	webrtc::TEST (AudioConverterTest, ConversionsPassSNRThreshold)