| /* |
| * 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/interface/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(); |
| virtual void SetUp() OVERRIDE; |
| virtual void TearDown() OVERRIDE; |
| virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, |
| int max_bytes, int* encoded_bytes); |
| virtual float DecodeABlock(const uint8_t* bit_stream, int 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_, STREAM_MAXW16_60MS); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_)); |
| EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1)); |
| EXPECT_EQ(0, 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, |
| int max_bytes, int* encoded_bytes) { |
| // ISAC takes 10 ms everycall |
| const int subblocks = block_duration_ms_ / 10; |
| const int subblock_length = 10 * input_sampling_khz_; |
| int value; |
| |
| 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); |
| } |
| clocks = clock() - clocks; |
| EXPECT_GT(value, 0); |
| assert(value <= max_bytes); |
| *encoded_bytes = value; |
| return 1000.0 * clocks / CLOCKS_PER_SEC; |
| } |
| |
| float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream, int encoded_bytes, |
| int16_t* out_data) { |
| int value; |
| int16_t audio_type; |
| clock_t clocks = clock(); |
| value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, |
| reinterpret_cast<const uint16_t*>(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 |