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: Reverted unnecessary change to buffer_unittest.cc Created 4 years, 10 months ago

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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..35831ca683a938702d9ca9d21078cb5ac25e47fd 100644

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

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

@@ -100,7 +100,6 @@ 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),

@@ -115,8 +114,8 @@ 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];

+ // Make sure the encode buffer is empty at the start of each test.

+ encoded_.Clear();

kwiberg-webrtc 2016/02/25 00:29:04 Can you allocate a Buffer on the stack for each te

ossu 2016/02/25 10:39:51 Sure.

// Logging to view input and output in Matlab.

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

DataLog::CreateLog();

@@ -128,9 +127,6 @@ class AudioDecoderTest : public ::testing::Test {

virtual void TearDown() {

delete decoder_;

decoder_ = NULL;

- // Delete arrays.

- delete [] encoded_;

- encoded_ = NULL;

// Close log.

DataLog::ReturnLog();

}

@@ -141,7 +137,7 @@ class AudioDecoderTest : public ::testing::Test {

// implementations are gone.

virtual int EncodeFrame(const int16_t* input,

size_t input_len_samples,

- uint8_t* output) {

+ rtc::Buffer* output) {

encoded_info_.encoded_bytes = 0;

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

RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),

@@ -162,7 +158,7 @@ class AudioDecoderTest : public ::testing::Test {

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);

@@ -179,6 +175,7 @@ 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_.Clear();

encoded_bytes_ = 0u;

InitEncoder();

std::vector<int16_t> input;

@@ -191,12 +188,12 @@ 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);

@@ -226,12 +223,13 @@ 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_);

+ encoded_.Clear();

+ 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 +237,7 @@ 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 +254,13 @@ 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_);

+ encoded_.Clear();

+ 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);

@@ -273,7 +273,7 @@ class AudioDecoderTest : public ::testing::Test {

test::ResampleInputAudioFile input_audio_;

int codec_input_rate_hz_;

- uint8_t* encoded_;

+ rtc::Buffer encoded_;

size_t frame_size_;

size_t data_length_;

size_t encoded_bytes_;

@@ -348,11 +348,13 @@ 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_);

+ encoded_.Clear();

+ 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);