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Adding DTX to WebRTC Opus wrapper

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This is a step toward adding Opus DTX support in WebRTC.

Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See

https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html

We transmit the first 1-byte packet to let decoder be in-sync

BUG=webrtc:1014
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org 2014-12-09 13:27:39 +00:00
parent 5c3ee4bce6
commit 4321f175f1
5 changed files with 390 additions and 108 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

28
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@ -39,7 +39,7 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
* Output:
* - encoded : Output compressed data buffer
*
* Return value : >0 - Length (in bytes) of coded data
* Return value : >=0 - Length (in bytes) of coded data
* -1 - Error
*/
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
@ -130,6 +130,32 @@ int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
*/
int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
/****************************************************************************
* WebRtcOpus_EnableDtx()
*
* This function enables Opus internal DTX for encoding.
*
* Input:
* - inst : Encoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
/****************************************************************************
* WebRtcOpus_DisableDtx()
*
* This function disables Opus internal DTX for encoding.
*
* Input:
* - inst : Encoder context
*
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
/*
* WebRtcOpus_SetComplexity(...)
*

2
webrtc/modules/audio_coding/codecs/opus/opus_inst.h
View File

@ -15,12 +15,14 @@
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
int in_dtx_mode;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
int prev_decoded_samples;
int channels;
int in_dtx_mode;
};

96
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -43,6 +43,7 @@ int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
state->encoder = opus_encoder_create(48000, channels, application,
&error);
state->in_dtx_mode = 0;
if (error == OPUS_OK && state->encoder != NULL) {
*inst = state;
return 0;
@ -80,9 +81,21 @@ int16_t WebRtcOpus_Encode(OpusEncInst* inst,
encoded,
length_encoded_buffer);
if (res > 0) {
if (res == 1) {
// Indicates DTX since the packet has nothing but a header. In principle,
// there is no need to send this packet. However, we do transmit the first
// occurrence to let the decoder know that the encoder enters DTX mode.
if (inst->in_dtx_mode) {
return 0;
} else {
inst->in_dtx_mode = 1;
return 1;
}
} else if (res > 1) {
inst->in_dtx_mode = 0;
return res;
}
return -1;
}
@ -140,6 +153,22 @@ int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
}
}
int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
} else {
return -1;
}
}
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
} else {
return -1;
}
}
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_COMPLEXITY(complexity));
@ -165,6 +194,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
/* Creation of memory all ok. */
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->in_dtx_mode = 0;
*inst = state;
return 0;
}
@ -195,53 +225,61 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
if (error == OPUS_OK) {
inst->in_dtx_mode = 0;
return 0;
}
return -1;
}
/* For decoder to determine if it is to output speech or comfort noise. */
static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
// Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
// to be so if the following |encoded_byte| are 0 or 1.
if (encoded_bytes == 0 && inst->in_dtx_mode) {
return 2; // Comfort noise.
} else if (encoded_bytes == 1) {
inst->in_dtx_mode = 1;
return 2; // Comfort noise.
} else {
inst->in_dtx_mode = 0;
return 0; // Speech.
}
}
/* |frame_size| is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
static int DecodeNative(OpusDecoder* inst, const uint8_t* encoded,
static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 0);
int16_t* decoded, int16_t* audio_type, int decode_fec) {
int res = opus_decode(inst->decoder, encoded, encoded_bytes,
(opus_int16*)decoded, frame_size, decode_fec);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
if (res > 0) {
return res;
}
if (res <= 0)
return -1;
}
static int DecodeFec(OpusDecoder* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 1);
*audio_type = DetermineAudioType(inst, encoded_bytes);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
if (res > 0) {
return res;
}
return -1;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples = DecodeNative(inst->decoder,
int decoded_samples;
if (encoded_bytes == 0) {
*audio_type = DetermineAudioType(inst, encoded_bytes);
decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1);
} else {
decoded_samples = DecodeNative(inst,
encoded,
encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded,
audio_type);
audio_type,
0);
}
if (decoded_samples < 0) {
return -1;
}
@ -264,8 +302,8 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
decoded_samples = DecodeNative(inst->decoder, NULL, 0, plc_samples,
decoded, &audio_type);
decoded_samples = DecodeNative(inst, NULL, 0, plc_samples,
decoded, &audio_type, 0);
if (decoded_samples < 0) {
return -1;
}
@ -285,8 +323,8 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
decoded_samples = DecodeFec(inst->decoder, encoded, encoded_bytes,
fec_samples, decoded, audio_type);
decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
fec_samples, decoded, audio_type, 1);
if (decoded_samples < 0) {
return -1;
}

344
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -12,34 +12,50 @@
#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"
namespace webrtc {
// Number of samples in a 60 ms stereo frame, sampled at 48 kHz.
const int kOpusMaxFrameSamples = 48 * 60 * 2;
using test::AudioLoop;
// 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 = 48 * 20;
const int kOpus20msFrameSamples = kOpusRateKhz * 20;
// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
const int kOpus10msFrameSamples = 48 * 10;
const int kOpus10msFrameSamples = kOpusRateKhz * 10;
class OpusTest : public ::testing::Test {
protected:
OpusTest();
virtual void SetUp();
void TestSetMaxPlaybackRate(opus_int32 expect, int32_t set);
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,
const int input_samples,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;
WebRtcOpusDecInst* opus_stereo_decoder_;
int16_t speech_data_[kOpusMaxFrameSamples];
int16_t output_data_[kOpusMaxFrameSamples];
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
int encoded_bytes_;
};
OpusTest::OpusTest()
@ -49,17 +65,16 @@ OpusTest::OpusTest()
opus_stereo_decoder_(NULL) {
}
void OpusTest::SetUp() {
FILE* input_file;
void OpusTest::PrepareSpeechData(int channel, int block_length_ms,
int loop_length_ms) {
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
input_file = fopen(file_name.c_str(), "rb");
ASSERT_TRUE(input_file != NULL);
ASSERT_EQ(kOpusMaxFrameSamples,
static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
kOpusMaxFrameSamples, input_file)));
fclose(input_file);
input_file = NULL;
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::TestSetMaxPlaybackRate(opus_int32 expect, int32_t set) {
@ -76,6 +91,155 @@ void OpusTest::TestSetMaxPlaybackRate(opus_int32 expect, int32_t set) {
EXPECT_EQ(expect, bandwidth);
}
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
const int16_t* input_audio,
const int input_samples,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type) {
encoded_bytes_ = WebRtcOpus_Encode(encoder,
input_audio,
input_samples, kMaxBytes,
bitstream_);
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(1, 20, 2000);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1));
// Set bitrate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, 32000));
// Set input audio as silence.
int16_t silence[kOpus20msFrameSamples] = {0};
// Setting DTX.
EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_mono_encoder_) :
WebRtcOpus_DisableDtx(opus_mono_encoder_));
int16_t audio_type;
int16_t output_data_decode[kOpus20msFrameSamples];
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_mono_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_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_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 < 21; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
// For this input signal, DTX happens now.
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
if (dtx) {
EXPECT_EQ(1, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
} else {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
// DTX mode is maintained 400 ms.
for (int i = 0; i < 20; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
if (dtx) {
EXPECT_EQ(0, encoded_bytes_); // Send 0 byte.
EXPECT_EQ(1, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
} else {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
// Quit DTX after 400 ms
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
// Enters DTX again immediately.
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
if (dtx) {
EXPECT_EQ(1, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
} else {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_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_mono_encoder_, silence,
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
EXPECT_GT(encoded_bytes_, 1);
EXPECT_EQ(0, opus_mono_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_mono_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_));
}
// Test failing Create.
TEST_F(OpusTest, OpusCreateFail) {
// Test to see that an invalid pointer is caught.
@ -110,6 +274,8 @@ TEST_F(OpusTest, OpusCreateFree) {
}
TEST_F(OpusTest, OpusEncodeDecodeMono) {
PrepareSpeechData(1, 20, 20);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1));
@ -121,16 +287,12 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode[kOpusMaxFrameSamples];
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
int16_t output_data_decode[kOpus20msFrameSamples];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
encoded_bytes, output_data_decode,
&audio_type));
EncodeDecode(opus_mono_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_));
@ -138,6 +300,8 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
}
TEST_F(OpusTest, OpusEncodeDecodeStereo) {
PrepareSpeechData(2, 20, 20);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
@ -149,16 +313,12 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode[kOpusMaxFrameSamples];
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
int16_t output_data_decode[kOpus20msFrameSamples * 2];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode,
&audio_type));
EncodeDecode(opus_stereo_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_stereo_decoder_,
output_data_decode, &audio_type));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));
@ -207,28 +367,25 @@ TEST_F(OpusTest, OpusSetComplexity) {
// Encode and decode one frame (stereo), initialize the decoder and
// decode once more.
TEST_F(OpusTest, OpusDecodeInit) {
PrepareSpeechData(2, 20, 20);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode[kOpusMaxFrameSamples];
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
int16_t output_data_decode[kOpus20msFrameSamples * 2];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode,
&audio_type));
EncodeDecode(opus_stereo_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_stereo_decoder_,
output_data_decode, &audio_type));
EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_stereo_decoder_));
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode,
encoded_bytes_, output_data_decode,
&audio_type));
// Free memory.
@ -255,6 +412,61 @@ TEST_F(OpusTest, OpusEnableDisableFec) {
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));
}
TEST_F(OpusTest, OpusEnableDisableDtx) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_mono_encoder_));
EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_stereo_encoder_));
// Create encoder memory, try with different bitrates.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
opus_int32 dtx;
// DTX is off by default.
opus_encoder_ctl(opus_mono_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
opus_encoder_ctl(opus_stereo_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Test to enable DTX.
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_mono_encoder_));
opus_encoder_ctl(opus_mono_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(1, dtx);
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_stereo_encoder_));
opus_encoder_ctl(opus_stereo_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(1, dtx);
// Test to disable DTX.
EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_mono_encoder_));
opus_encoder_ctl(opus_mono_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_stereo_encoder_));
opus_encoder_ctl(opus_stereo_encoder_->encoder,
OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_));
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));
}
TEST_F(OpusTest, OpusDtxOff) {
TestDtxEffect(false);
}
TEST_F(OpusTest, OpusDtxOn) {
TestDtxEffect(true);
}
TEST_F(OpusTest, OpusSetPacketLossRate) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_mono_encoder_, 50));
@ -303,6 +515,8 @@ TEST_F(OpusTest, OpusSetMaxPlaybackRate) {
// PLC in mono mode.
TEST_F(OpusTest, OpusDecodePlcMono) {
PrepareSpeechData(1, 20, 20);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1));
@ -314,19 +528,15 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode[kOpusMaxFrameSamples];
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
int16_t output_data_decode[kOpus20msFrameSamples];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
encoded_bytes, output_data_decode,
&audio_type));
EncodeDecode(opus_mono_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_mono_decoder_,
output_data_decode, &audio_type));
// Call decoder PLC.
int16_t plc_buffer[kOpusMaxFrameSamples];
int16_t plc_buffer[kOpus20msFrameSamples];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodePlc(opus_mono_decoder_, plc_buffer, 1));
@ -337,6 +547,8 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
// PLC in stereo mode.
TEST_F(OpusTest, OpusDecodePlcStereo) {
PrepareSpeechData(2, 20, 20);
// Create encoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
@ -348,19 +560,15 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_));
// Encode & decode.
int16_t encoded_bytes;
int16_t audio_type;
int16_t output_data_decode[kOpusMaxFrameSamples];
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
int16_t output_data_decode[kOpus20msFrameSamples * 2];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode,
&audio_type));
EncodeDecode(opus_stereo_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_stereo_decoder_,
output_data_decode, &audio_type));
// Call decoder PLC.
int16_t plc_buffer[kOpusMaxFrameSamples];
int16_t plc_buffer[kOpus20msFrameSamples * 2];
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodePlc(opus_stereo_decoder_, plc_buffer, 1));
@ -371,27 +579,29 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
// Duration estimation.
TEST_F(OpusTest, OpusDurationEstimation) {
PrepareSpeechData(2, 20, 20);
// Create.
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2));
int16_t encoded_bytes;
// 10 ms.
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
// 10 ms. We use only first 10 ms of a 20 ms block.
encoded_bytes_ = WebRtcOpus_Encode(opus_stereo_encoder_,
speech_data_.GetNextBlock(),
kOpus10msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus10msFrameSamples,
WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
encoded_bytes_));
// 20 ms
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
encoded_bytes_ = WebRtcOpus_Encode(opus_stereo_encoder_,
speech_data_.GetNextBlock(),
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DurationEst(opus_stereo_decoder_, bitstream_,
encoded_bytes));
encoded_bytes_));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_));

8
webrtc/modules/audio_coding/neteq/test/neteq_opus_fec_quality_test.cc
View File

@ -111,6 +111,8 @@ static const bool runtime_dummy =
DEFINE_bool(fec, true, "Whether to enable FEC for encoding.");
DEFINE_bool(dtx, true, "Whether to enable DTX for encoding.");
class NetEqOpusFecQualityTest : public NetEqQualityTest {
protected:
NetEqOpusFecQualityTest();
@ -123,6 +125,7 @@ class NetEqOpusFecQualityTest : public NetEqQualityTest {
int channels_;
int bit_rate_kbps_;
bool fec_;
bool dtx_;
int target_loss_rate_;
};
@ -137,6 +140,7 @@ NetEqOpusFecQualityTest::NetEqOpusFecQualityTest()
channels_(FLAGS_channels),
bit_rate_kbps_(FLAGS_bit_rate_kbps),
fec_(FLAGS_fec),
dtx_(FLAGS_dtx),
target_loss_rate_(FLAGS_reported_loss_rate) {
}
@ -149,6 +153,9 @@ void NetEqOpusFecQualityTest::SetUp() {
if (fec_) {
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
}
if (dtx_) {
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
}
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
target_loss_rate_));
NetEqQualityTest::SetUp();
@ -166,7 +173,6 @@ int NetEqOpusFecQualityTest::EncodeBlock(int16_t* in_data,
int value = WebRtcOpus_Encode(opus_encoder_, in_data,
block_size_samples, max_bytes,
payload);
EXPECT_GT(value, 0);
return value;
}