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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
111 lines
3.6 KiB
C++
111 lines
3.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <string>
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#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/testsupport/fileutils.h"
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struct WebRtcISACStruct;
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namespace webrtc {
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// Number of samples in a 60 ms, sampled at 32 kHz.
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const int kIsacNumberOfSamples = 320 * 6;
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// Maximum number of bytes in output bitstream.
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const size_t kMaxBytes = 1000;
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class IsacTest : public ::testing::Test {
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protected:
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IsacTest();
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virtual void SetUp();
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WebRtcISACStruct* isac_codec_;
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int16_t speech_data_[kIsacNumberOfSamples];
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int16_t output_data_[kIsacNumberOfSamples];
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uint8_t bitstream_[kMaxBytes];
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uint8_t bitstream_small_[7]; // Simulate sync packets.
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};
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IsacTest::IsacTest()
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: isac_codec_(NULL) {
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}
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void IsacTest::SetUp() {
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// Read some samples from a speech file, to be used in the encode test.
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FILE* input_file;
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const std::string file_name =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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input_file = fopen(file_name.c_str(), "rb");
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ASSERT_TRUE(input_file != NULL);
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ASSERT_EQ(kIsacNumberOfSamples,
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static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
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kIsacNumberOfSamples, input_file)));
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fclose(input_file);
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input_file = NULL;
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}
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// Test failing Create.
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TEST_F(IsacTest, IsacCreateFail) {
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// Test to see that an invalid pointer is caught.
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EXPECT_EQ(-1, WebRtcIsac_Create(NULL));
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}
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// Test failing Free.
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TEST_F(IsacTest, IsacFreeFail) {
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// Test to see that free function doesn't crash.
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EXPECT_EQ(0, WebRtcIsac_Free(NULL));
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}
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// Test normal Create and Free.
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TEST_F(IsacTest, IsacCreateFree) {
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EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
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EXPECT_TRUE(isac_codec_ != NULL);
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EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));}
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TEST_F(IsacTest, IsacUpdateBWE) {
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// Create encoder memory.
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EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
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// Init encoder (adaptive mode) and decoder.
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WebRtcIsac_EncoderInit(isac_codec_, 0);
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WebRtcIsac_DecoderInit(isac_codec_);
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int encoded_bytes;
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// Test with call with a small packet (sync packet).
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EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1,
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12345, 56789));
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// Encode 60 ms of data (needed to create a first packet).
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_EQ(0, encoded_bytes);
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_EQ(0, encoded_bytes);
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_EQ(0, encoded_bytes);
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_EQ(0, encoded_bytes);
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_EQ(0, encoded_bytes);
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encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
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EXPECT_GT(encoded_bytes, 0);
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// Call to update bandwidth estimator with real data.
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EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,
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static_cast<size_t>(encoded_bytes),
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1, 12345, 56789));
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// Free memory.
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EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
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}
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} // namespace webrtc
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