52bc4f4797
> Removing unused opus wrapper APIs. > > WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit(). > > WebRtcOpus_DecodePlcMaster/Slave() are also removed. > > BUG= > R=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/28139004 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
120 lines
3.9 KiB
C++
120 lines
3.9 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 "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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#include "webrtc/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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virtual void SetUp() OVERRIDE;
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virtual void TearDown() OVERRIDE;
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virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
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int max_bytes, int* encoded_bytes);
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virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
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int16_t* out_data);
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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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}
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void OpusSpeedTest::SetUp() {
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AudioCodecSpeedTest::SetUp();
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/* Create encoder memory. */
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EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_));
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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, uint8_t* bit_stream,
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int max_bytes, int* encoded_bytes) {
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clock_t clocks = clock();
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int value = WebRtcOpus_Encode(opus_encoder_, in_data,
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input_length_sample_, max_bytes,
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bit_stream);
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clocks = clock() - clocks;
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EXPECT_GT(value, 0);
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*encoded_bytes = 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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int encoded_bytes, 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_DecodeNew(opus_decoder_, bit_stream, encoded_bytes,
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out_data, &audio_type);
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clocks = clock() - clocks;
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EXPECT_EQ(output_length_sample_, value);
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return 1000.0 * clocks / CLOCKS_PER_SEC;
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}
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#define ADD_TEST(complexity) \
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TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
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/* Test audio length in second. */ \
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size_t kDurationSec = 400; \
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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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// List all test cases: (channel, bit rat, filename, extension).
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const coding_param param_set[] =
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{::std::tr1::make_tuple(1, 64000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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::std::tr1::make_tuple(1, 32000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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::std::tr1::make_tuple(2, 64000,
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string("audio_coding/music_stereo_48kHz"),
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string("pcm"), true)};
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INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
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::testing::ValuesIn(param_set));
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} // namespace webrtc
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