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Unified Diff: webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc

Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index 4630e448073d7d64d55a14cc1866fbccde3abd81..c059fc5d0171f5952dbaaf1a44b4a895f160329e 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -10,6 +10,7 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
@@ -44,8 +45,7 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
void PrepareSpeechData(int channel, int block_length_ms, int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
- const int16_t* input_audio,
- size_t input_samples,
+ rtc::ArrayView<const int16_t> input_audio,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
@@ -96,13 +96,14 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
}
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
- const int16_t* input_audio,
- size_t input_samples,
+ 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, input_samples,
- kMaxBytes, bitstream_);
+ int encoded_bytes_int = WebRtcOpus_Encode(
+ encoder, input_audio.data(),
+ rtc::CheckedDivExact(input_audio.size(), static_cast<size_t>(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_);
@@ -129,8 +130,7 @@ void OpusTest::TestDtxEffect(bool dtx) {
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_);
+ std::vector<int16_t> silence(kOpus20msFrameSamples * channels_, 0);
// Setting DTX.
EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
@@ -142,9 +142,8 @@ void OpusTest::TestDtxEffect(bool dtx) {
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ 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) {
@@ -158,10 +157,9 @@ void OpusTest::TestDtxEffect(bool dtx) {
// 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,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, 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);
@@ -183,9 +181,9 @@ void OpusTest::TestDtxEffect(bool dtx) {
// DTX mode is maintained 19 frames.
for (int i = 0; i < 19; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples,
- opus_decoder_, output_data_decode, &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(0U, encoded_bytes_) // Send 0 byte.
<< "Opus should have entered DTX mode.";
@@ -201,10 +199,9 @@ void OpusTest::TestDtxEffect(bool dtx) {
}
// Quit DTX after 19 frames.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, 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);
@@ -212,10 +209,9 @@ void OpusTest::TestDtxEffect(bool dtx) {
EXPECT_EQ(0, audio_type); // Speech.
// Enters DTX again immediately.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, 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);
@@ -232,10 +228,9 @@ void OpusTest::TestDtxEffect(bool dtx) {
silence[0] = 10000;
if (dtx) {
// Verify that encoder/decoder can jump out from DTX mode.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, 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);
@@ -244,7 +239,6 @@ void OpusTest::TestDtxEffect(bool dtx) {
// Free memory.
delete[] output_data_decode;
- delete[] silence;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
@@ -314,10 +308,9 @@ TEST_P(OpusTest, OpusEncodeDecode) {
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(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
// Free memory.
delete[] output_data_decode;
@@ -374,10 +367,9 @@ TEST_P(OpusTest, OpusDecodeInit) {
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(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
WebRtcOpus_DecoderInit(opus_decoder_);
@@ -513,10 +505,9 @@ TEST_P(OpusTest, OpusDecodePlc) {
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(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ 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_];
@@ -542,10 +533,12 @@ TEST_P(OpusTest, OpusDurationEstimation) {
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// 10 ms. We use only first 10 ms of a 20 ms block.
- int encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus10msFrameSamples,
- kMaxBytes, bitstream_);
+ auto speech_block = speech_data_.GetNextBlock();
+ int encoded_bytes_int = WebRtcOpus_Encode(
+ opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(),
+ 2 * static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus10msFrameSamples,
static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -553,10 +546,11 @@ TEST_P(OpusTest, OpusDurationEstimation) {
static_cast<size_t>(encoded_bytes_int))));
// 20 ms
- encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus20msFrameSamples,
- kMaxBytes, bitstream_);
+ speech_block = speech_data_.GetNextBlock();
+ encoded_bytes_int = WebRtcOpus_Encode(
+ opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(), static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus20msFrameSamples,
static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -594,10 +588,12 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
OpusRepacketizer* rp = opus_repacketizer_create();
for (int idx = 0; idx < kPackets; idx++) {
- encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, kMaxBytes,
- bitstream_);
+ auto speech_block = speech_data_.GetNextBlock();
+ encoded_bytes_ =
+ WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(),
+ static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
}
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