Home | History | Annotate | Download | only in source 
      1 /*
      2  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 #include <string>
     11 
     12 #include "testing/gtest/include/gtest/gtest.h"
     13 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
     14 #include "webrtc/test/testsupport/fileutils.h"
     15 
     16 struct WebRtcISACStruct;
     17 
     18 namespace webrtc {
     19 
     20 // Number of samples in a 60 ms, sampled at 32 kHz.
     21 const int kIsacNumberOfSamples = 320 * 6;
     22 // Maximum number of bytes in output bitstream.
     23 const size_t kMaxBytes = 1000;
     24 
     25 class IsacTest : public ::testing::Test {
     26  protected:
     27   IsacTest();
     28   virtual void SetUp();
     29 
     30   WebRtcISACStruct* isac_codec_;
     31 
     32   int16_t speech_data_[kIsacNumberOfSamples];
     33   int16_t output_data_[kIsacNumberOfSamples];
     34   int16_t bitstream_[kMaxBytes / 2];
     35   uint8_t bitstream_small_[7];  // Simulate sync packets.
     36 };
     37 
     38 IsacTest::IsacTest()
     39     : isac_codec_(NULL) {
     40 }
     41 
     42 void IsacTest::SetUp() {
     43   // Read some samples from a speech file, to be used in the encode test.
     44   FILE* input_file;
     45   const std::string file_name =
     46         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
     47   input_file = fopen(file_name.c_str(), "rb");
     48   ASSERT_TRUE(input_file != NULL);
     49   ASSERT_EQ(kIsacNumberOfSamples,
     50             static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
     51                                        kIsacNumberOfSamples, input_file)));
     52   fclose(input_file);
     53   input_file = NULL;
     54 }
     55 
     56 // Test failing Create.
     57 TEST_F(IsacTest, IsacCreateFail) {
     58   // Test to see that an invalid pointer is caught.
     59   EXPECT_EQ(-1, WebRtcIsac_Create(NULL));
     60 }
     61 
     62 // Test failing Free.
     63 TEST_F(IsacTest, IsacFreeFail) {
     64   // Test to see that free function doesn't crash.
     65   EXPECT_EQ(0, WebRtcIsac_Free(NULL));
     66 }
     67 
     68 // Test normal Create and Free.
     69 TEST_F(IsacTest, IsacCreateFree) {
     70   EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
     71   EXPECT_TRUE(isac_codec_ != NULL);
     72   EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));}
     73 
     74 TEST_F(IsacTest, IsacUpdateBWE) {
     75   // Create encoder memory.
     76   EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
     77 
     78   // Init encoder (adaptive mode) and decoder.
     79   WebRtcIsac_EncoderInit(isac_codec_, 0);
     80   WebRtcIsac_DecoderInit(isac_codec_);
     81 
     82   // Encode & decode.
     83   int16_t encoded_bytes;
     84   uint16_t* coded = reinterpret_cast<uint16_t*>(bitstream_);
     85   uint16_t* coded_small = reinterpret_cast<uint16_t*>(bitstream_small_);
     86 
     87   // Test with call with a small packet (sync packet).
     88   EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, coded_small, 7, 1,
     89                                             12345, 56789));
     90 
     91   // Encode 60 ms of data (needed to create a first packet).
     92   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
     93   EXPECT_EQ(0, encoded_bytes);
     94   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
     95   EXPECT_EQ(0, encoded_bytes);
     96   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
     97   EXPECT_EQ(0, encoded_bytes);
     98   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
     99   EXPECT_EQ(0, encoded_bytes);
    100   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
    101   EXPECT_EQ(0, encoded_bytes);
    102   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
    103 
    104   // Call to update bandwidth estimator with real data.
    105   EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, coded, encoded_bytes, 1,
    106                                            12345, 56789));
    107 
    108   // Free memory.
    109   EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
    110 }
    111 
    112 }  // namespace webrtc
    113