zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Update a ton of audio code to use size_t more correctly and in general reduce

Browse Source 
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
This commit is contained in:
Peter Kasting
2015-08-24 14:52:23 -07:00
parent b594041ec8
commit dce40cf804
471 changed files with 3716 additions and 3499 deletions
Download Patch File Download Diff File Expand all files Collapse all files

142
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -25,11 +25,11 @@ using ::testing::Combine;
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 1000;
// Sample rate of Opus.
const int kOpusRateKhz = 48;
const size_t kOpusRateKhz = 48;
// Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
const int kOpus20msFrameSamples = kOpusRateKhz * 20;
const size_t kOpus20msFrameSamples = kOpusRateKhz * 20;
// Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
const int kOpus10msFrameSamples = kOpusRateKhz * 10;
const size_t kOpus10msFrameSamples = kOpusRateKhz * 10;
class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
protected:
@ -45,7 +45,7 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
int EncodeDecode(WebRtcOpusEncInst* encoder,
const int16_t* input_audio,
int input_samples,
size_t input_samples,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
@ -58,7 +58,7 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
int encoded_bytes_;
size_t encoded_bytes_;
int channels_;
int application_;
};
@ -97,15 +97,14 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
const int16_t* input_audio,
int input_samples,
size_t 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);
int encoded_bytes_int = WebRtcOpus_Encode(encoder, input_audio, input_samples,
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
return WebRtcOpus_Decode(decoder, bitstream_,
encoded_bytes_, output_audio,
audio_type);
@ -139,13 +138,14 @@ void OpusTest::TestDtxEffect(bool dtx) {
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
static_cast<size_t>(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_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
@ -156,11 +156,11 @@ void OpusTest::TestDtxEffect(bool dtx) {
// 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));
static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type)));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1);
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
@ -180,17 +180,17 @@ void OpusTest::TestDtxEffect(bool dtx) {
// 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));
static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, kOpus20msFrameSamples,
opus_decoder_, output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(0, encoded_bytes_) // Send 0 byte.
EXPECT_EQ(0U, 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_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
@ -199,27 +199,27 @@ void OpusTest::TestDtxEffect(bool dtx) {
// Quit DTX after 19 frames.
EXPECT_EQ(kOpus20msFrameSamples,
EncodeDecode(opus_encoder_, silence,
kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type));
static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type)));
EXPECT_GT(encoded_bytes_, 1);
EXPECT_GT(encoded_bytes_, 1U);
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));
static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(1, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1U, 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_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
@ -230,10 +230,10 @@ void OpusTest::TestDtxEffect(bool dtx) {
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);
static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, kOpus20msFrameSamples, 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);
EXPECT_EQ(0, audio_type); // Speech.
@ -311,9 +311,10 @@ TEST_P(OpusTest, OpusEncodeDecode) {
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));
static_cast<size_t>(EncodeDecode(
opus_encoder_, speech_data_.GetNextBlock(),
kOpus20msFrameSamples, opus_decoder_, output_data_decode,
&audio_type)));
// Free memory.
delete[] output_data_decode;
@ -370,16 +371,17 @@ TEST_P(OpusTest, OpusDecodeInit) {
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));
static_cast<size_t>(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));
static_cast<size_t>(WebRtcOpus_Decode(
opus_decoder_, bitstream_, encoded_bytes_, output_data_decode,
&audio_type)));
// Free memory.
delete[] output_data_decode;
@ -508,14 +510,16 @@ TEST_P(OpusTest, OpusDecodePlc) {
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));
static_cast<size_t>(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));
static_cast<size_t>(WebRtcOpus_DecodePlc(
opus_decoder_, plc_buffer, 1)));
// Free memory.
delete[] plc_buffer;
@ -535,24 +539,26 @@ 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.
encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus10msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_, 0);
int encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus10msFrameSamples,
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus10msFrameSamples,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
encoded_bytes_));
static_cast<size_t>(WebRtcOpus_DurationEst(
opus_decoder_, bitstream_,
static_cast<size_t>(encoded_bytes_int))));
// 20 ms
encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus20msFrameSamples, kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_, 0);
encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
speech_data_.GetNextBlock(),
kOpus20msFrameSamples,
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
encoded_bytes_));
static_cast<size_t>(WebRtcOpus_DurationEst(
opus_decoder_, bitstream_,
static_cast<size_t>(encoded_bytes_int))));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
@ -595,11 +601,13 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
EXPECT_EQ(kOpus20msFrameSamples * kPackets,
WebRtcOpus_DurationEst(opus_decoder_, bitstream_, encoded_bytes_));
static_cast<size_t>(WebRtcOpus_DurationEst(
opus_decoder_, bitstream_, encoded_bytes_)));
EXPECT_EQ(kOpus20msFrameSamples * kPackets,
WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
output_data_decode.get(), &audio_type));
static_cast<size_t>(WebRtcOpus_Decode(
opus_decoder_, bitstream_, encoded_bytes_,
output_data_decode.get(), &audio_type)));
// Free memory.
opus_repacketizer_destroy(rp);