webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc - Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

Unified Diff: webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Removed use of <functional> and moved MockAudioEncoderHelper things into the respective MockAudioEn… Created 4 years, 10 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

« webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc ('K') | « webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc ('k') | webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc » ('j') | webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Hide Comments ('s')

Index: webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

index 58dd6c6680f0c7c9f7f1f164f51e96e8f14de26b..b705aea81a5fdd9362e179cb98c3c989e13dafeb 100644

--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -100,10 +100,8 @@ class AudioDecoderTest : public ::testing::Test {

webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),

32000),

codec_input_rate_hz_(32000), // Legacy default value.

- encoded_(NULL),

frame_size_(0),

data_length_(0),

- encoded_bytes_(0),

channels_(1),

payload_type_(17),

decoder_(NULL) {}

@@ -115,8 +113,6 @@ class AudioDecoderTest : public ::testing::Test {

codec_input_rate_hz_ = audio_encoder_->SampleRateHz();

// Create arrays.

ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";

- // Longest encoded data is produced by PCM16b with 2 bytes per sample.

- encoded_ = new uint8_t[data_length_ * 2];

// Logging to view input and output in Matlab.

// Use 'gyp -Denable_data_logging=1' to enable logging.

DataLog::CreateLog();

@@ -128,9 +124,6 @@ class AudioDecoderTest : public ::testing::Test {

virtual void TearDown() {

delete decoder_;

decoder_ = NULL;

- // Delete arrays.

- delete [] encoded_;

- encoded_ = NULL;

// Close log.

DataLog::ReturnLog();

}

@@ -141,15 +134,16 @@ class AudioDecoderTest : public ::testing::Test {

// implementations are gone.

virtual int EncodeFrame(const int16_t* input,

size_t input_len_samples,

- uint8_t* output) {

- encoded_info_.encoded_bytes = 0;

+ rtc::Buffer* output) {

+ AudioEncoder::EncodedInfo encoded_info;

+ encoded_info.encoded_bytes = 0;

hlundin-webrtc 2016/02/29 12:46:47 No need to set encoded_bytes = 0, since the ctor d

ossu 2016/02/29 13:23:01 Acknowledged.

const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;

RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),

input_len_samples);

std::unique_ptr<int16_t[]> interleaved_input(

new int16_t[channels_ * samples_per_10ms]);

for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {

- EXPECT_EQ(0u, encoded_info_.encoded_bytes);

+ EXPECT_EQ(0u, encoded_info.encoded_bytes);

// Duplicate the mono input signal to however many channels the test

// wants.

@@ -157,15 +151,15 @@ class AudioDecoderTest : public ::testing::Test {

samples_per_10ms, channels_,

interleaved_input.get());

- encoded_info_ = audio_encoder_->Encode(

+ encoded_info = audio_encoder_->Encode(

0, rtc::ArrayView<const int16_t>(interleaved_input.get(),

audio_encoder_->NumChannels() *

audio_encoder_->SampleRateHz() /

100),

- data_length_ * 2, output);

+ output);

}

- EXPECT_EQ(payload_type_, encoded_info_.payload_type);

- return static_cast<int>(encoded_info_.encoded_bytes);

+ EXPECT_EQ(payload_type_, encoded_info.payload_type);

+ return static_cast<int>(encoded_info.encoded_bytes);

}

// Encodes and decodes audio. The absolute difference between the input and

@@ -179,7 +173,8 @@ class AudioDecoderTest : public ::testing::Test {

ASSERT_GE(channel_diff_tolerance, 0) <<

"Test must define a channel_diff_tolerance >= 0";

size_t processed_samples = 0u;

- encoded_bytes_ = 0u;

+ rtc::Buffer encoded;

+ size_t encoded_bytes = 0u;

InitEncoder();

std::vector<int16_t> input;

std::vector<int16_t> decoded;

@@ -191,23 +186,23 @@ class AudioDecoderTest : public ::testing::Test {

ASSERT_TRUE(input_audio_.Read(

frame_size_, codec_input_rate_hz_, &input[processed_samples]));

size_t enc_len = EncodeFrame(

- &input[processed_samples], frame_size_, &encoded_[encoded_bytes_]);

+ &input[processed_samples], frame_size_, &encoded);

// Make sure that frame_size_ * channels_ samples are allocated and free.

decoded.resize((processed_samples + frame_size_) * channels_, 0);

AudioDecoder::SpeechType speech_type;

size_t dec_len = decoder_->Decode(

- &encoded_[encoded_bytes_], enc_len, codec_input_rate_hz_,

+ &encoded.data()[encoded_bytes], enc_len, codec_input_rate_hz_,

frame_size_ * channels_ * sizeof(int16_t),

&decoded[processed_samples * channels_], &speech_type);

EXPECT_EQ(frame_size_ * channels_, dec_len);

- encoded_bytes_ += enc_len;

+ encoded_bytes += enc_len;

processed_samples += frame_size_;

}

// For some codecs it doesn't make sense to check expected number of bytes,

// since the number can vary for different platforms. Opus and iSAC are

// such codecs. In this case expected_bytes is set to 0.

if (expected_bytes) {

- EXPECT_EQ(expected_bytes, encoded_bytes_);

+ EXPECT_EQ(expected_bytes, encoded_bytes);

}

CompareInputOutput(

input, decoded, processed_samples, channels_, tolerance, delay);

@@ -226,12 +221,13 @@ class AudioDecoderTest : public ::testing::Test {

std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);

ASSERT_TRUE(

input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));

- size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);

+ rtc::Buffer encoded;

+ size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);

size_t dec_len;

AudioDecoder::SpeechType speech_type1, speech_type2;

decoder_->Reset();

std::unique_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);

- dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,

+ dec_len = decoder_->Decode(encoded.data(), enc_len, codec_input_rate_hz_,

frame_size_ * channels_ * sizeof(int16_t),

output1.get(), &speech_type1);

ASSERT_LE(dec_len, frame_size_ * channels_);

@@ -239,7 +235,7 @@ class AudioDecoderTest : public ::testing::Test {

// Re-init decoder and decode again.

decoder_->Reset();

std::unique_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);

- dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,

+ dec_len = decoder_->Decode(encoded.data(), enc_len, codec_input_rate_hz_,

frame_size_ * channels_ * sizeof(int16_t),

output2.get(), &speech_type2);

ASSERT_LE(dec_len, frame_size_ * channels_);

@@ -256,11 +252,13 @@ class AudioDecoderTest : public ::testing::Test {

std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);

ASSERT_TRUE(

input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));

- size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);

+ rtc::Buffer encoded;

+ size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);

AudioDecoder::SpeechType speech_type;

decoder_->Reset();

std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);

- size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,

+ size_t dec_len = decoder_->Decode(encoded.data(), enc_len,

+ codec_input_rate_hz_,

frame_size_ * channels_ * sizeof(int16_t),

output.get(), &speech_type);

EXPECT_EQ(frame_size_ * channels_, dec_len);

@@ -273,13 +271,10 @@ class AudioDecoderTest : public ::testing::Test {

test::ResampleInputAudioFile input_audio_;

int codec_input_rate_hz_;

- uint8_t* encoded_;

size_t frame_size_;

size_t data_length_;

- size_t encoded_bytes_;

size_t channels_;

const int payload_type_;

- AudioEncoder::EncodedInfo encoded_info_;

AudioDecoder* decoder_;

std::unique_ptr<AudioEncoder> audio_encoder_;

};

@@ -348,11 +343,13 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {

std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);

ASSERT_TRUE(

input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));

- size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);

+ rtc::Buffer encoded;

+ size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);

AudioDecoder::SpeechType speech_type;

decoder_->Reset();

std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);

- size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,

+ size_t dec_len = decoder_->Decode(encoded.data(), enc_len,

+ codec_input_rate_hz_,

frame_size_ * channels_ * sizeof(int16_t),

output.get(), &speech_type);

EXPECT_EQ(frame_size_, dec_len);