| /* |
| * Copyright (c) 2013 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 <memory> |
| #include <string> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h" |
| #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| |
| namespace webrtc { |
| |
| using test::AudioLoop; |
| using ::testing::TestWithParam; |
| using ::testing::Values; |
| using ::testing::Combine; |
| |
| // Maximum number of bytes in output bitstream. |
| const size_t kMaxBytes = 1000; |
| // Sample rate of Opus. |
| const size_t kOpusRateKhz = 48; |
| // Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz. |
| const size_t kOpus20msFrameSamples = kOpusRateKhz * 20; |
| // Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz. |
| const size_t kOpus10msFrameSamples = kOpusRateKhz * 10; |
| |
| class OpusTest : public TestWithParam<::testing::tuple<int, int>> { |
| protected: |
| OpusTest(); |
| |
| void TestDtxEffect(bool dtx, int block_length_ms); |
| |
| // Prepare |speech_data_| for encoding, read from a hard-coded file. |
| // After preparation, |speech_data_.GetNextBlock()| returns a pointer to a |
| // block of |block_length_ms| milliseconds. The data is looped every |
| // |loop_length_ms| milliseconds. |
| void PrepareSpeechData(size_t channel, |
| int block_length_ms, |
| int loop_length_ms); |
| |
| int EncodeDecode(WebRtcOpusEncInst* encoder, |
| rtc::ArrayView<const int16_t> input_audio, |
| WebRtcOpusDecInst* decoder, |
| int16_t* output_audio, |
| int16_t* audio_type); |
| |
| void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder, |
| opus_int32 expect, int32_t set); |
| |
| void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels, |
| uint16_t bound) const; |
| |
| WebRtcOpusEncInst* opus_encoder_; |
| WebRtcOpusDecInst* opus_decoder_; |
| |
| AudioLoop speech_data_; |
| uint8_t bitstream_[kMaxBytes]; |
| size_t encoded_bytes_; |
| size_t channels_; |
| int application_; |
| }; |
| |
| OpusTest::OpusTest() |
| : opus_encoder_(NULL), |
| opus_decoder_(NULL), |
| encoded_bytes_(0), |
| channels_(static_cast<size_t>(::testing::get<0>(GetParam()))), |
| application_(::testing::get<1>(GetParam())) { |
| } |
| |
| void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms, |
| int loop_length_ms) { |
| const std::string file_name = |
| webrtc::test::ResourcePath((channel == 1) ? |
| "audio_coding/testfile32kHz" : |
| "audio_coding/teststereo32kHz", "pcm"); |
| if (loop_length_ms < block_length_ms) { |
| loop_length_ms = block_length_ms; |
| } |
| EXPECT_TRUE(speech_data_.Init(file_name, |
| loop_length_ms * kOpusRateKhz * channel, |
| block_length_ms * kOpusRateKhz * channel)); |
| } |
| |
| void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder, |
| opus_int32 expect, |
| int32_t set) { |
| opus_int32 bandwidth; |
| EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set)); |
| opus_encoder_ctl(opus_encoder_->encoder, |
| OPUS_GET_MAX_BANDWIDTH(&bandwidth)); |
| EXPECT_EQ(expect, bandwidth); |
| } |
| |
| void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples, |
| size_t channels, uint16_t bound) const { |
| for (size_t i = 0; i < samples; ++i) { |
| for (size_t c = 0; c < channels; ++c) { |
| ASSERT_GE(audio[i * channels + c], -bound); |
| ASSERT_LE(audio[i * channels + c], bound); |
| } |
| } |
| } |
| |
| int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder, |
| rtc::ArrayView<const int16_t> input_audio, |
| WebRtcOpusDecInst* decoder, |
| int16_t* output_audio, |
| int16_t* audio_type) { |
| int encoded_bytes_int = WebRtcOpus_Encode( |
| encoder, input_audio.data(), |
| rtc::CheckedDivExact(input_audio.size(), channels_), |
| kMaxBytes, bitstream_); |
| EXPECT_GE(encoded_bytes_int, 0); |
| encoded_bytes_ = static_cast<size_t>(encoded_bytes_int); |
| int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_); |
| int act_len = WebRtcOpus_Decode(decoder, bitstream_, |
| encoded_bytes_, output_audio, |
| audio_type); |
| EXPECT_EQ(est_len, act_len); |
| return act_len; |
| } |
| |
| // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when |
| // they should not. This test is signal dependent. |
| void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) { |
| PrepareSpeechData(channels_, block_length_ms, 2000); |
| const size_t samples = kOpusRateKhz * block_length_ms; |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); |
| |
| // Set bitrate. |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, |
| channels_ == 1 ? 32000 : 64000)); |
| |
| // Set input audio as silence. |
| std::vector<int16_t> silence(samples * channels_, 0); |
| |
| // Setting DTX. |
| EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) : |
| WebRtcOpus_DisableDtx(opus_encoder_)); |
| |
| int16_t audio_type; |
| int16_t* output_data_decode = new int16_t[samples * channels_]; |
| |
| for (int i = 0; i < 100; ++i) { |
| EXPECT_EQ(samples, |
| static_cast<size_t>(EncodeDecode( |
| opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_, |
| output_data_decode, &audio_type))); |
| // If not DTX, it should never enter DTX mode. If DTX, we do not care since |
| // whether it enters DTX depends on the signal type. |
| if (!dtx) { |
| EXPECT_GT(encoded_bytes_, 1U); |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| } |
| } |
| |
| // We input some silent segments. In DTX mode, the encoder will stop sending. |
| // However, DTX may happen after a while. |
| for (int i = 0; i < 30; ++i) { |
| EXPECT_EQ(samples, |
| static_cast<size_t>(EncodeDecode( |
| opus_encoder_, silence, opus_decoder_, output_data_decode, |
| &audio_type))); |
| if (!dtx) { |
| EXPECT_GT(encoded_bytes_, 1U); |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| } else if (encoded_bytes_ == 1) { |
| EXPECT_EQ(1, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(1, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(2, audio_type); // Comfort noise. |
| break; |
| } |
| } |
| |
| // When Opus is in DTX, it wakes up in a regular basis. It sends two packets, |
| // one with an arbitrary size and the other of 1-byte, then stops sending for |
| // a certain number of frames. |
| |
| // |max_dtx_frames| is the maximum number of frames Opus can stay in DTX. |
| const int max_dtx_frames = 400 / block_length_ms + 1; |
| |
| // We run |kRunTimeMs| milliseconds of pure silence. |
| const int kRunTimeMs = 4500; |
| |
| // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in |
| // Opus needs time to adapt), the absolute values of DTX decoded signal are |
| // bounded by |kOutputValueBound|. |
| const int kCheckTimeMs = 4000; |
| |
| #if defined(OPUS_FIXED_POINT) |
| // Fixed-point Opus generates a random (comfort) noise, which has a less |
| // predictable value bound than its floating-point Opus. This value depends on |
| // input signal, and the time window for checking the output values (between |
| // |kCheckTimeMs| and |kRunTimeMs|). |
| const uint16_t kOutputValueBound = 30; |
| |
| #else |
| const uint16_t kOutputValueBound = 2; |
| #endif |
| |
| int time = 0; |
| while (time < kRunTimeMs) { |
| // DTX mode is maintained for maximum |max_dtx_frames| frames. |
| int i = 0; |
| for (; i < max_dtx_frames; ++i) { |
| time += block_length_ms; |
| EXPECT_EQ(samples, |
| static_cast<size_t>(EncodeDecode( |
| opus_encoder_, silence, opus_decoder_, output_data_decode, |
| &audio_type))); |
| if (dtx) { |
| if (encoded_bytes_ > 1) |
| break; |
| EXPECT_EQ(0U, encoded_bytes_) // Send 0 byte. |
| << "Opus should have entered DTX mode."; |
| EXPECT_EQ(1, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(1, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(2, audio_type); // Comfort noise. |
| if (time >= kCheckTimeMs) { |
| CheckAudioBounded(output_data_decode, samples, channels_, |
| kOutputValueBound); |
| } |
| } else { |
| EXPECT_GT(encoded_bytes_, 1U); |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| } |
| } |
| |
| if (dtx) { |
| // With DTX, Opus must stop transmission for some time. |
| EXPECT_GT(i, 1); |
| } |
| |
| // We expect a normal payload. |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| |
| // Enters DTX again immediately. |
| time += block_length_ms; |
| EXPECT_EQ(samples, |
| static_cast<size_t>(EncodeDecode( |
| opus_encoder_, silence, opus_decoder_, output_data_decode, |
| &audio_type))); |
| if (dtx) { |
| EXPECT_EQ(1U, encoded_bytes_); // Send 1 byte. |
| EXPECT_EQ(1, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(1, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(2, audio_type); // Comfort noise. |
| if (time >= kCheckTimeMs) { |
| CheckAudioBounded(output_data_decode, samples, channels_, |
| kOutputValueBound); |
| } |
| } else { |
| EXPECT_GT(encoded_bytes_, 1U); |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| } |
| } |
| |
| silence[0] = 10000; |
| if (dtx) { |
| // Verify that encoder/decoder can jump out from DTX mode. |
| EXPECT_EQ(samples, |
| static_cast<size_t>(EncodeDecode( |
| opus_encoder_, silence, opus_decoder_, output_data_decode, |
| &audio_type))); |
| EXPECT_GT(encoded_bytes_, 1U); |
| EXPECT_EQ(0, opus_encoder_->in_dtx_mode); |
| EXPECT_EQ(0, opus_decoder_->in_dtx_mode); |
| EXPECT_EQ(0, audio_type); // Speech. |
| } |
| |
| // Free memory. |
| delete[] output_data_decode; |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| // Test failing Create. |
| TEST(OpusTest, OpusCreateFail) { |
| WebRtcOpusEncInst* opus_encoder; |
| WebRtcOpusDecInst* opus_decoder; |
| |
| // Test to see that an invalid pointer is caught. |
| EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0)); |
| // Invalid channel number. |
| EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 3, 0)); |
| // Invalid applciation mode. |
| EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2)); |
| |
| EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1)); |
| // Invalid channel number. |
| EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 3)); |
| } |
| |
| // Test failing Free. |
| TEST(OpusTest, OpusFreeFail) { |
| // Test to see that an invalid pointer is caught. |
| EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL)); |
| EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL)); |
| } |
| |
| // Test normal Create and Free. |
| TEST_P(OpusTest, OpusCreateFree) { |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); |
| EXPECT_TRUE(opus_encoder_ != NULL); |
| EXPECT_TRUE(opus_decoder_ != NULL); |
| // Free encoder and decoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusEncodeDecode) { |
| PrepareSpeechData(channels_, 20, 20); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, |
| channels_)); |
| |
| // Set bitrate. |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, |
| channels_ == 1 ? 32000 : 64000)); |
| |
| // Check number of channels for decoder. |
| EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_)); |
| |
| // Check application mode. |
| opus_int32 app; |
| opus_encoder_ctl(opus_encoder_->encoder, |
| OPUS_GET_APPLICATION(&app)); |
| EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO, |
| app); |
| |
| // Encode & decode. |
| int16_t audio_type; |
| int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_]; |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>( |
| EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), |
| opus_decoder_, output_data_decode, &audio_type))); |
| |
| // Free memory. |
| delete[] output_data_decode; |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusSetBitRate) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000)); |
| |
| // Create encoder memory, try with different bitrates. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000)); |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000)); |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000)); |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 600000)); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusSetComplexity) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9)); |
| |
| // Create encoder memory, try with different complexities. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| |
| EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0)); |
| EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10)); |
| EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 11)); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusForceChannels) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 1)); |
| |
| ASSERT_EQ(0, |
| WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_)); |
| |
| if (channels_ == 2) { |
| EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3)); |
| EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2)); |
| EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1)); |
| EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0)); |
| } else { |
| EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 2)); |
| EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1)); |
| EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0)); |
| } |
| |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| // Encode and decode one frame, initialize the decoder and |
| // decode once more. |
| TEST_P(OpusTest, OpusDecodeInit) { |
| PrepareSpeechData(channels_, 20, 20); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); |
| |
| // Encode & decode. |
| int16_t audio_type; |
| int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_]; |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>( |
| EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), |
| opus_decoder_, output_data_decode, &audio_type))); |
| |
| WebRtcOpus_DecoderInit(opus_decoder_); |
| |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>(WebRtcOpus_Decode( |
| opus_decoder_, bitstream_, encoded_bytes_, output_data_decode, |
| &audio_type))); |
| |
| // Free memory. |
| delete[] output_data_decode; |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusEnableDisableFec) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_encoder_)); |
| EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_)); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| |
| EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_)); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusEnableDisableDtx) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_encoder_)); |
| EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_)); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| |
| opus_int32 dtx; |
| |
| // DTX is off by default. |
| opus_encoder_ctl(opus_encoder_->encoder, |
| OPUS_GET_DTX(&dtx)); |
| EXPECT_EQ(0, dtx); |
| |
| // Test to enable DTX. |
| EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_)); |
| opus_encoder_ctl(opus_encoder_->encoder, |
| OPUS_GET_DTX(&dtx)); |
| EXPECT_EQ(1, dtx); |
| |
| // Test to disable DTX. |
| EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_)); |
| opus_encoder_ctl(opus_encoder_->encoder, |
| OPUS_GET_DTX(&dtx)); |
| EXPECT_EQ(0, dtx); |
| |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusDtxOff) { |
| TestDtxEffect(false, 10); |
| TestDtxEffect(false, 20); |
| TestDtxEffect(false, 40); |
| } |
| |
| TEST_P(OpusTest, OpusDtxOn) { |
| TestDtxEffect(true, 10); |
| TestDtxEffect(true, 20); |
| TestDtxEffect(true, 40); |
| } |
| |
| TEST_P(OpusTest, OpusSetPacketLossRate) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50)); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| |
| EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50)); |
| EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1)); |
| EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 101)); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusSetMaxPlaybackRate) { |
| // Test without creating encoder memory. |
| EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000)); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 24000); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 16001); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 16000); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 12001); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 12000); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 8001); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 8000); |
| SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 4000); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| } |
| |
| // Test PLC. |
| TEST_P(OpusTest, OpusDecodePlc) { |
| PrepareSpeechData(channels_, 20, 20); |
| |
| // Create encoder memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); |
| |
| // Set bitrate. |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, |
| channels_== 1 ? 32000 : 64000)); |
| |
| // Check number of channels for decoder. |
| EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_)); |
| |
| // Encode & decode. |
| int16_t audio_type; |
| int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_]; |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>( |
| EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), |
| opus_decoder_, output_data_decode, &audio_type))); |
| |
| // Call decoder PLC. |
| int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_]; |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>(WebRtcOpus_DecodePlc( |
| opus_decoder_, plc_buffer, 1))); |
| |
| // Free memory. |
| delete[] plc_buffer; |
| delete[] output_data_decode; |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| // Duration estimation. |
| TEST_P(OpusTest, OpusDurationEstimation) { |
| PrepareSpeechData(channels_, 20, 20); |
| |
| // Create. |
| EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_)); |
| |
| // 10 ms. We use only first 10 ms of a 20 ms block. |
| auto speech_block = speech_data_.GetNextBlock(); |
| int encoded_bytes_int = WebRtcOpus_Encode( |
| opus_encoder_, speech_block.data(), |
| rtc::CheckedDivExact(speech_block.size(), 2 * channels_), |
| kMaxBytes, bitstream_); |
| EXPECT_GE(encoded_bytes_int, 0); |
| EXPECT_EQ(kOpus10msFrameSamples, |
| static_cast<size_t>(WebRtcOpus_DurationEst( |
| opus_decoder_, bitstream_, |
| static_cast<size_t>(encoded_bytes_int)))); |
| |
| // 20 ms |
| speech_block = speech_data_.GetNextBlock(); |
| encoded_bytes_int = WebRtcOpus_Encode( |
| opus_encoder_, speech_block.data(), |
| rtc::CheckedDivExact(speech_block.size(), channels_), |
| kMaxBytes, bitstream_); |
| EXPECT_GE(encoded_bytes_int, 0); |
| EXPECT_EQ(kOpus20msFrameSamples, |
| static_cast<size_t>(WebRtcOpus_DurationEst( |
| opus_decoder_, bitstream_, |
| static_cast<size_t>(encoded_bytes_int)))); |
| |
| // Free memory. |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| TEST_P(OpusTest, OpusDecodeRepacketized) { |
| const int kPackets = 6; |
| |
| PrepareSpeechData(channels_, 20, 20 * kPackets); |
| |
| // Create encoder memory. |
| ASSERT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, |
| channels_, |
| application_)); |
| ASSERT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, |
| channels_)); |
| |
| // Set bitrate. |
| EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, |
| channels_ == 1 ? 32000 : 64000)); |
| |
| // Check number of channels for decoder. |
| EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_)); |
| |
| // Encode & decode. |
| int16_t audio_type; |
| std::unique_ptr<int16_t[]> output_data_decode( |
| new int16_t[kPackets * kOpus20msFrameSamples * channels_]); |
| OpusRepacketizer* rp = opus_repacketizer_create(); |
| |
| for (int idx = 0; idx < kPackets; idx++) { |
| auto speech_block = speech_data_.GetNextBlock(); |
| encoded_bytes_ = |
| WebRtcOpus_Encode(opus_encoder_, speech_block.data(), |
| rtc::CheckedDivExact(speech_block.size(), channels_), |
| kMaxBytes, bitstream_); |
| EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_)); |
| } |
| |
| encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes); |
| |
| EXPECT_EQ(kOpus20msFrameSamples * kPackets, |
| static_cast<size_t>(WebRtcOpus_DurationEst( |
| opus_decoder_, bitstream_, encoded_bytes_))); |
| |
| EXPECT_EQ(kOpus20msFrameSamples * kPackets, |
| static_cast<size_t>(WebRtcOpus_Decode( |
| opus_decoder_, bitstream_, encoded_bytes_, |
| output_data_decode.get(), &audio_type))); |
| |
| // Free memory. |
| opus_repacketizer_destroy(rp); |
| EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); |
| EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); |
| } |
| |
| INSTANTIATE_TEST_CASE_P(VariousMode, |
| OpusTest, |
| Combine(Values(1, 2), Values(0, 1))); |
| |
| |
| } // namespace webrtc |