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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc | 147 |
1 files changed, 147 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc new file mode 100644 index 0000000000..4477e8a5f8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -0,0 +1,147 @@ +/* + * 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 "modules/audio_coding/codecs/opus/opus_interface.h" +#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h" + +using ::std::string; + +namespace webrtc { + +static const int kOpusBlockDurationMs = 20; +static const int kOpusSamplingKhz = 48; + +class OpusSpeedTest : public AudioCodecSpeedTest { + protected: + OpusSpeedTest(); + void SetUp() override; + void TearDown() override; + float EncodeABlock(int16_t* in_data, + uint8_t* bit_stream, + size_t max_bytes, + size_t* encoded_bytes) override; + float DecodeABlock(const uint8_t* bit_stream, + size_t encoded_bytes, + int16_t* out_data) override; + WebRtcOpusEncInst* opus_encoder_; + WebRtcOpusDecInst* opus_decoder_; +}; + +OpusSpeedTest::OpusSpeedTest() + : AudioCodecSpeedTest(kOpusBlockDurationMs, + kOpusSamplingKhz, + kOpusSamplingKhz), + opus_encoder_(NULL), + opus_decoder_(NULL) {} + +void OpusSpeedTest::SetUp() { + AudioCodecSpeedTest::SetUp(); + // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode. + int app = channels_ == 1 ? 0 : 1; + /* Create encoder memory. */ + EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000)); + EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000)); + /* Set bitrate. */ + EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_)); +} + +void OpusSpeedTest::TearDown() { + AudioCodecSpeedTest::TearDown(); + /* Free memory. */ + EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); + EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); +} + +float OpusSpeedTest::EncodeABlock(int16_t* in_data, + uint8_t* bit_stream, + size_t max_bytes, + size_t* encoded_bytes) { + clock_t clocks = clock(); + int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_, + max_bytes, bit_stream); + clocks = clock() - clocks; + EXPECT_GT(value, 0); + *encoded_bytes = static_cast<size_t>(value); + return 1000.0 * clocks / CLOCKS_PER_SEC; +} + +float OpusSpeedTest::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 = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data, + &audio_type); + clocks = clock() - clocks; + EXPECT_EQ(output_length_sample_, static_cast<size_t>(value)); + return 1000.0 * clocks / CLOCKS_PER_SEC; +} + +/* Test audio length in second. */ +constexpr size_t kDurationSec = 400; + +#define ADD_TEST(complexity) \ + TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \ + /* Set complexity. */ \ + printf("Setting complexity to %d ...\n", complexity); \ + EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \ + EncodeDecode(kDurationSec); \ + } + +ADD_TEST(10) +ADD_TEST(9) +ADD_TEST(8) +ADD_TEST(7) +ADD_TEST(6) +ADD_TEST(5) +ADD_TEST(4) +ADD_TEST(3) +ADD_TEST(2) +ADD_TEST(1) +ADD_TEST(0) + +#define ADD_BANDWIDTH_TEST(bandwidth) \ + TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) { \ + /* Set bandwidth. */ \ + printf("Setting bandwidth to %d ...\n", bandwidth); \ + EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, bandwidth)); \ + EncodeDecode(kDurationSec); \ + } + +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND) +ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND) + +// List all test cases: (channel, bit rat, filename, extension). +const coding_param param_set[] = { + std::make_tuple(1, + 64000, + string("audio_coding/speech_mono_32_48kHz"), + string("pcm"), + true), + std::make_tuple(1, + 32000, + string("audio_coding/speech_mono_32_48kHz"), + string("pcm"), + true), + std::make_tuple(2, + 64000, + string("audio_coding/music_stereo_48kHz"), + string("pcm"), + true)}; + +INSTANTIATE_TEST_SUITE_P(AllTest, + OpusSpeedTest, + ::testing::ValuesIn(param_set)); + +} // namespace webrtc |