blob: 6d9b013ae5ad245821edd572fa9b13077c9798bc [file] [log] [blame]
/*
* Copyright (c) 2013 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.
*/
#include <string>
#include "modules/audio_coding/codecs/isac/main/include/isac.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
struct WebRtcISACStruct;
namespace webrtc {
// Number of samples in a 60 ms, sampled at 32 kHz.
const int kIsacNumberOfSamples = 320 * 6;
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 1000;
class IsacTest : public ::testing::Test {
protected:
IsacTest();
virtual void SetUp();
WebRtcISACStruct* isac_codec_;
int16_t speech_data_[kIsacNumberOfSamples];
int16_t output_data_[kIsacNumberOfSamples];
uint8_t bitstream_[kMaxBytes];
uint8_t bitstream_small_[7]; // Simulate sync packets.
};
IsacTest::IsacTest() : isac_codec_(NULL) {}
void IsacTest::SetUp() {
// Read some samples from a speech file, to be used in the encode test.
FILE* input_file;
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
input_file = fopen(file_name.c_str(), "rb");
ASSERT_TRUE(input_file != NULL);
ASSERT_EQ(kIsacNumberOfSamples,
static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
kIsacNumberOfSamples, input_file)));
fclose(input_file);
input_file = NULL;
}
// Test failing Create.
TEST_F(IsacTest, IsacCreateFail) {
// Test to see that an invalid pointer is caught.
EXPECT_EQ(-1, WebRtcIsac_Create(NULL));
}
// Test failing Free.
TEST_F(IsacTest, IsacFreeFail) {
// Test to see that free function doesn't crash.
EXPECT_EQ(0, WebRtcIsac_Free(NULL));
}
// Test normal Create and Free.
TEST_F(IsacTest, IsacCreateFree) {
EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
EXPECT_TRUE(isac_codec_ != NULL);
EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
}
TEST_F(IsacTest, IsacUpdateBWE) {
// Create encoder memory.
EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
// Init encoder (adaptive mode) and decoder.
WebRtcIsac_EncoderInit(isac_codec_, 0);
WebRtcIsac_DecoderInit(isac_codec_);
int encoded_bytes;
// Test with call with a small packet (sync packet).
EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1,
12345, 56789));
// Encode 60 ms of data (needed to create a first packet).
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_GT(encoded_bytes, 0);
// Call to update bandwidth estimator with real data.
EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,
static_cast<size_t>(encoded_bytes),
1, 12345, 56789));
// Free memory.
EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
}
} // namespace webrtc