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/*
* 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.
*/
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
using ::std::string;
namespace webrtc {
static const int kIsacBlockDurationMs = 30;
static const int kIsacInputSamplingKhz = 16;
static const int kIsacOutputSamplingKhz = 16;
class IsacSpeedTest : public AudioCodecSpeedTest {
protected:
IsacSpeedTest();
void SetUp() override;
void TearDown() override;
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes);
virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
int16_t* out_data);
ISACFIX_MainStruct *ISACFIX_main_inst_;
};
IsacSpeedTest::IsacSpeedTest()
: AudioCodecSpeedTest(kIsacBlockDurationMs,
kIsacInputSamplingKhz,
kIsacOutputSamplingKhz),
ISACFIX_main_inst_(NULL) {
}
void IsacSpeedTest::SetUp() {
AudioCodecSpeedTest::SetUp();
// Check whether the allocated buffer for the bit stream is large enough.
EXPECT_GE(max_bytes_, static_cast<size_t>(STREAM_MAXW16_60MS));
// Create encoder memory.
EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_);
// Set bitrate and block length.
EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
block_duration_ms_));
}
void IsacSpeedTest::TearDown() {
AudioCodecSpeedTest::TearDown();
// Free memory.
EXPECT_EQ(0, WebRtcIsacfix_Free(ISACFIX_main_inst_));
}
float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes) {
// ISAC takes 10 ms everycall
const int subblocks = block_duration_ms_ / 10;
const int subblock_length = 10 * input_sampling_khz_;
int value = 0;
clock_t clocks = clock();
size_t pointer = 0;
for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
bit_stream);
if (idx == subblocks - 1)
EXPECT_GT(value, 0);
else
EXPECT_EQ(0, value);
}
clocks = clock() - clocks;
*encoded_bytes = static_cast<size_t>(value);
assert(*encoded_bytes <= max_bytes);
return 1000.0 * clocks / CLOCKS_PER_SEC;
}
float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
size_t encoded_bytes,
int16_t* out_data) {
int value;
int16_t audio_type;
clock_t clocks = clock();
value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
out_data, &audio_type);
clocks = clock() - clocks;
EXPECT_EQ(output_length_sample_, value);
return 1000.0 * clocks / CLOCKS_PER_SEC;
}
TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) {
size_t kDurationSec = 400; // Test audio length in second.
EncodeDecode(kDurationSec);
}
const coding_param param_set[] =
{::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
string("pcm"), true)};
INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
::testing::ValuesIn(param_set));
} // namespace webrtc
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