blob: 028d2ec544c0081e57a337476a93bffc79d5fb9b [file] [log] [blame]
/*
* 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 <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.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 int kOpusRateKhz = 48;
// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
const int kOpus20msFrameSamples = kOpusRateKhz * 20;
// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
const int kOpus10msFrameSamples = kOpusRateKhz * 10;
class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
protected:
OpusTest();
void TestDtxEffect(bool dtx);
// 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(int channel, int block_length_ms, int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
const int16_t* input_audio,
int input_samples,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
opus_int32 expect, int32_t set);
WebRtcOpusEncInst* opus_encoder_;
WebRtcOpusDecInst* opus_decoder_;
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
int encoded_bytes_;
int channels_;
int application_;
};
OpusTest::OpusTest()
: opus_encoder_(NULL),
opus_decoder_(NULL),
encoded_bytes_(0),
channels_(::testing::get<0>(GetParam())),
application_(::testing::get<1>(GetParam())) {
}
void OpusTest::PrepareSpeechData(int 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);
}
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
const int16_t* input_audio,
int input_samples,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type) {
encoded_bytes_ = WebRtcOpus_Encode(encoder,
input_audio,
input_samples, kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_, 0);
return WebRtcOpus_Decode(decoder, bitstream_,
encoded_bytes_, output_audio,
audio_type);
}
// 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) {
PrepareSpeechData(channels_, 20, 2000);
// 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.
int16_t* silence = new int16_t[kOpus20msFrameSamples * channels_];
memset(silence, 0, sizeof(int16_t) * kOpus20msFrameSamples * channels_);
// 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[kOpus20msFrameSamples * channels_];
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, 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_, 1);
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(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1);
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
// 19 frames.
const int cycles = 5;
for (int j = 0; j < cycles; ++j) {
// DTX mode is maintained 19 frames.
for (int i = 0; i < 19; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
if (dtx) {
EXPECT_EQ(0, 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.
} else {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
// Quit DTX after 19 frames.
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
EXPECT_GT(encoded_bytes_, 1);
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.
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
if (dtx) {
EXPECT_EQ(1, 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.
} else {
EXPECT_GT(encoded_bytes_, 1);
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(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
EXPECT_GT(encoded_bytes_, 1);
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;
delete[] silence;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
// Test failing Create.
TEST(OpusTest, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, 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, DISABLED_ON_IOS(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_));
}
// Encode and decode one frame, initialize the decoder and
// decode once more.
TEST_P(OpusTest, DISABLED_ON_IOS(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,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_decoder_));
EXPECT_EQ(kOpus20msFrameSamples,
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, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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, DISABLED_ON_IOS(OpusDtxOff)) {
TestDtxEffect(false);
}
TEST_P(OpusTest, DISABLED_ON_IOS(OpusDtxOn)) {
TestDtxEffect(true);
}
TEST_P(OpusTest, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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, DISABLED_ON_IOS(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,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
// Call decoder PLC.
int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
EXPECT_EQ(kOpus20msFrameSamples,
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, DISABLED_ON_IOS(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.
encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus10msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_, 0);
EXPECT_EQ(kOpus10msFrameSamples,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
encoded_bytes_));
// 20 ms
encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_, 0);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
encoded_bytes_));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST_P(OpusTest, DISABLED_ON_IOS(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;
rtc::scoped_ptr<int16_t[]> output_data_decode(
new int16_t[kPackets * kOpus20msFrameSamples * channels_]);
OpusRepacketizer* rp = opus_repacketizer_create();
for (int idx = 0; idx < kPackets; idx++) {
encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus20msFrameSamples, 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,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_, encoded_bytes_));
EXPECT_EQ(kOpus20msFrameSamples * kPackets,
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