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/*
* 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/opus/include/opus_interface.h"
#include "webrtc/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;
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes);
virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
int16_t* out_data);
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));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
/* 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_, value);
return 1000.0 * clocks / CLOCKS_PER_SEC;
}
#define ADD_TEST(complexity) \
TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
/* Test audio length in second. */ \
size_t kDurationSec = 400; \
/* 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);
// List all test cases: (channel, bit rat, filename, extension).
const coding_param param_set[] =
{::std::tr1::make_tuple(1, 64000,
string("audio_coding/speech_mono_32_48kHz"),
string("pcm"), true),
::std::tr1::make_tuple(1, 32000,
string("audio_coding/speech_mono_32_48kHz"),
string("pcm"), true),
::std::tr1::make_tuple(2, 64000,
string("audio_coding/music_stereo_48kHz"),
string("pcm"), true)};
INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
::testing::ValuesIn(param_set));
} // namespace webrtc
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