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c84f661b10
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
147 lines
5.1 KiB
C++
147 lines
5.1 KiB
C++
/*
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* Copyright (c) 2014 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 "modules/audio_coding/codecs/opus/opus_interface.h"
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#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
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using ::std::string;
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namespace webrtc {
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static const int kOpusBlockDurationMs = 20;
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static const int kOpusSamplingKhz = 48;
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class OpusSpeedTest : public AudioCodecSpeedTest {
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protected:
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OpusSpeedTest();
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void SetUp() override;
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void TearDown() override;
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float EncodeABlock(int16_t* in_data,
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uint8_t* bit_stream,
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size_t max_bytes,
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size_t* encoded_bytes) override;
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float DecodeABlock(const uint8_t* bit_stream,
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size_t encoded_bytes,
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int16_t* out_data) override;
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WebRtcOpusEncInst* opus_encoder_;
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WebRtcOpusDecInst* opus_decoder_;
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};
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OpusSpeedTest::OpusSpeedTest()
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: AudioCodecSpeedTest(kOpusBlockDurationMs,
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kOpusSamplingKhz,
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kOpusSamplingKhz),
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opus_encoder_(NULL),
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opus_decoder_(NULL) {}
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void OpusSpeedTest::SetUp() {
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AudioCodecSpeedTest::SetUp();
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// If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
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int app = channels_ == 1 ? 0 : 1;
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/* Create encoder memory. */
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EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app));
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EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
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/* Set bitrate. */
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EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
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}
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void OpusSpeedTest::TearDown() {
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AudioCodecSpeedTest::TearDown();
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/* Free memory. */
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EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
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EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
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}
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float OpusSpeedTest::EncodeABlock(int16_t* in_data,
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uint8_t* bit_stream,
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size_t max_bytes,
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size_t* encoded_bytes) {
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clock_t clocks = clock();
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int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
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max_bytes, bit_stream);
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clocks = clock() - clocks;
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EXPECT_GT(value, 0);
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*encoded_bytes = static_cast<size_t>(value);
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return 1000.0 * clocks / CLOCKS_PER_SEC;
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}
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float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
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size_t encoded_bytes,
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int16_t* out_data) {
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int value;
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int16_t audio_type;
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clock_t clocks = clock();
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value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
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&audio_type);
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clocks = clock() - clocks;
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EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
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return 1000.0 * clocks / CLOCKS_PER_SEC;
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}
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/* Test audio length in second. */
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constexpr size_t kDurationSec = 400;
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#define ADD_TEST(complexity) \
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TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
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/* Set complexity. */ \
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printf("Setting complexity to %d ...\n", complexity); \
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EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
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EncodeDecode(kDurationSec); \
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}
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ADD_TEST(10);
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ADD_TEST(9);
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ADD_TEST(8);
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ADD_TEST(7);
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ADD_TEST(6);
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ADD_TEST(5);
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ADD_TEST(4);
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ADD_TEST(3);
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ADD_TEST(2);
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ADD_TEST(1);
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ADD_TEST(0);
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#define ADD_BANDWIDTH_TEST(bandwidth) \
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TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) { \
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/* Set bandwidth. */ \
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printf("Setting bandwidth to %d ...\n", bandwidth); \
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EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, bandwidth)); \
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EncodeDecode(kDurationSec); \
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}
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ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND);
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ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND);
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ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND);
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ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND);
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ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND);
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// List all test cases: (channel, bit rat, filename, extension).
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const coding_param param_set[] = {
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std::make_tuple(1,
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64000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"),
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true),
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std::make_tuple(1,
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32000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"),
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true),
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std::make_tuple(2,
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64000,
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string("audio_coding/music_stereo_48kHz"),
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string("pcm"),
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true)};
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INSTANTIATE_TEST_SUITE_P(AllTest,
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OpusSpeedTest,
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::testing::ValuesIn(param_set));
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} // namespace webrtc
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