Update a ton of audio code to use size_t more correctly and in general reduce
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}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
index dc59984..b5a86d0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
@@ -93,7 +93,8 @@
AudioFrame output_frame;
EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
- const int samples_per_block = output_freq_hz_ * 10 / 1000;
+ const size_t samples_per_block =
+ static_cast<size_t>(output_freq_hz_ * 10 / 1000);
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
if (expected_output_channels_ != kArbitraryChannels) {
if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
index dd570e6..2a0bbe1 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
@@ -160,7 +160,8 @@
AudioFrame output_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
- const int samples_per_block = output_freq_hz_ * 10 / 1000;
+ const size_t samples_per_block =
+ static_cast<size_t>(output_freq_hz_ * 10 / 1000);
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
if (exptected_output_channels_ != kArbitraryChannels) {
if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 4c11197..1cefeb6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -344,7 +344,7 @@
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
enum NetEqOutputType type;
- int samples_per_channel;
+ size_t samples_per_channel;
int num_channels;
bool return_silence = false;
@@ -394,7 +394,7 @@
}
// NetEq always returns 10 ms of audio.
- current_sample_rate_hz_ = samples_per_channel * 100;
+ current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
// Update if resampling is required.
bool need_resampling = (desired_freq_hz != -1) &&
@@ -403,18 +403,19 @@
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
- samples_per_channel =
+ int samples_per_channel_int =
resampler_.Resample10Msec(last_audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
temp_output);
- if (samples_per_channel < 0) {
+ if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
return -1;
}
+ samples_per_channel = static_cast<size_t>(samples_per_channel_int);
}
// The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
@@ -422,17 +423,18 @@
// TODO(henrik.lundin) Glitches in the output may appear if the output rate
// from NetEq changes. See WebRTC issue 3923.
if (need_resampling) {
- samples_per_channel =
+ int samples_per_channel_int =
resampler_.Resample10Msec(audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_frame->data_);
- if (samples_per_channel < 0) {
+ if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return -1;
}
+ samples_per_channel = static_cast<size_t>(samples_per_channel_int);
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
@@ -448,7 +450,7 @@
audio_frame->num_channels_ = num_channels;
audio_frame->samples_per_channel_ = samples_per_channel;
- audio_frame->sample_rate_hz_ = samples_per_channel * 100;
+ audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
// Should set |vad_activity| before calling SetAudioFrameActivityAndType().
audio_frame->vad_activity_ = previous_audio_activity_;
@@ -787,10 +789,11 @@
frame->sample_rate_hz_ = current_sample_rate_hz_;
}
- frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms.
+ frame->samples_per_channel_ =
+ static_cast<size_t>(frame->sample_rate_hz_ / 100); // Always 10 ms.
frame->speech_type_ = AudioFrame::kCNG;
frame->vad_activity_ = AudioFrame::kVadPassive;
- int samples = frame->samples_per_channel_ * frame->num_channels_;
+ size_t samples = frame->samples_per_channel_ * frame->num_channels_;
memset(frame->data_, 0, samples * sizeof(int16_t));
return true;
}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
index 97d87b1..2650725 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -29,9 +29,9 @@
int in_freq_hz,
int out_freq_hz,
int num_audio_channels,
- int out_capacity_samples,
+ size_t out_capacity_samples,
int16_t* out_audio) {
- int in_length = in_freq_hz * num_audio_channels / 100;
+ size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
int out_length = out_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
@@ -39,7 +39,7 @@
return -1;
}
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
- return in_length / num_audio_channels;
+ return static_cast<int>(in_length / num_audio_channels);
}
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
index a8fc6b6..a19b0c4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
@@ -26,7 +26,7 @@
int in_freq_hz,
int out_freq_hz,
int num_audio_channels,
- int out_capacity_samples,
+ size_t out_capacity_samples,
int16_t* out_audio);
private:
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
index b96db6b..91df16f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
@@ -29,7 +29,8 @@
: clock_(0),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
- input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
+ input_block_size_samples_(
+ static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
codec_registered_(false),
test_duration_ms_(test_duration_ms),
frame_type_(kAudioFrameSpeech),
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
index 4c4db5b..09fe9e6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
@@ -63,7 +63,7 @@
rtc::scoped_ptr<AudioCoding> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
- const int input_block_size_samples_;
+ const size_t input_block_size_samples_;
AudioFrame input_frame_;
bool codec_registered_;
int test_duration_ms_;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
index 1819d59..74e98d9 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
@@ -31,7 +31,8 @@
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
- input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
+ input_block_size_samples_(
+ static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
codec_registered_(false),
test_duration_ms_(test_duration_ms),
frame_type_(kAudioFrameSpeech),
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
index 8cdc298..008e264 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
@@ -71,7 +71,7 @@
rtc::scoped_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
- const int input_block_size_samples_;
+ const size_t input_block_size_samples_;
AudioFrame input_frame_;
bool codec_registered_;
int test_duration_ms_;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 32d60a7..46980d3 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -76,22 +76,24 @@
}
// Stereo-to-mono can be used as in-place.
-int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
+int DownMix(const AudioFrame& frame,
+ size_t length_out_buff,
+ int16_t* out_buff) {
if (length_out_buff < frame.samples_per_channel_) {
return -1;
}
- for (int n = 0; n < frame.samples_per_channel_; ++n)
+ for (size_t n = 0; n < frame.samples_per_channel_; ++n)
out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
return 0;
}
// Mono-to-stereo can be used as in-place.
-int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
+int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
if (length_out_buff < frame.samples_per_channel_) {
return -1;
}
- for (int n = frame.samples_per_channel_; n > 0; --n) {
- int i = n - 1;
+ for (size_t n = frame.samples_per_channel_; n != 0; --n) {
+ size_t i = n - 1;
int16_t sample = frame.data_[i];
out_buff[2 * i + 1] = sample;
out_buff[2 * i] = sample;
@@ -338,11 +340,10 @@
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
InputData* input_data) {
- if (audio_frame.samples_per_channel_ <= 0) {
+ if (audio_frame.samples_per_channel_ == 0) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot Add 10 ms audio, payload length is negative or "
- "zero");
+ "Cannot Add 10 ms audio, payload length is zero");
return -1;
}
@@ -354,7 +355,7 @@
}
// If the length and frequency matches. We currently just support raw PCM.
- if ((audio_frame.sample_rate_hz_ / 100) !=
+ if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
audio_frame.samples_per_channel_) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot Add 10 ms audio, input frequency and length doesn't"
@@ -477,17 +478,19 @@
// The result of the resampler is written to output frame.
dest_ptr_audio = preprocess_frame_.data_;
- preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec(
+ int samples_per_channel = resampler_.Resample10Msec(
src_ptr_audio, in_frame.sample_rate_hz_,
codec_manager_.CurrentEncoder()->SampleRateHz(),
preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
dest_ptr_audio);
- if (preprocess_frame_.samples_per_channel_ < 0) {
+ if (samples_per_channel < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot add 10 ms audio, resampling failed");
return -1;
}
+ preprocess_frame_.samples_per_channel_ =
+ static_cast<size_t>(samples_per_channel);
preprocess_frame_.sample_rate_hz_ =
codec_manager_.CurrentEncoder()->SampleRateHz();
}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index beb49bc..c451854 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -248,7 +248,7 @@
struct InputData {
uint32_t input_timestamp;
const int16_t* audio;
- uint16_t length_per_channel;
+ size_t length_per_channel;
uint8_t audio_channel;
// If a re-mix is required (up or down), this buffer will store a re-mixed
// version of the input.
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
index eea51a3..418ddd1 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
@@ -272,7 +272,8 @@
EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0);
- EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
+ EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+ audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 0af6af8..e5371d0 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -314,7 +314,8 @@
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0);
- EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
+ EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+ audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
index cad6ee9..7b9c7ed 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
@@ -326,10 +326,10 @@
// Make up a CodecInst.
send_codec_inst_.channels = external_speech_encoder->NumChannels();
send_codec_inst_.plfreq = external_speech_encoder->SampleRateHz();
- send_codec_inst_.pacsize =
- rtc::CheckedDivExact(external_speech_encoder->Max10MsFramesInAPacket() *
- send_codec_inst_.plfreq,
- 100);
+ send_codec_inst_.pacsize = rtc::CheckedDivExact(
+ static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
+ send_codec_inst_.plfreq),
+ 100);
send_codec_inst_.pltype = -1; // Not valid.
send_codec_inst_.rate = -1; // Not valid.
static const char kName[] = "external";
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc
index 4b08f75..d0ae783 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc
@@ -150,7 +150,7 @@
}
} else {
int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
- for (int k = 0; k < audio_frame.samples_per_channel_; k++) {
+ for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
stereo_audio[k << 1] = audio_frame.data_[k];
stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
}
@@ -172,7 +172,7 @@
}
}
-void PCMFile::Write10MsData(int16_t* playout_buffer, uint16_t length_smpls) {
+void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
length_smpls) {
return;
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/main/test/PCMFile.h
index c4487b8..8353898 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.h
@@ -36,7 +36,7 @@
int32_t Read10MsData(AudioFrame& audio_frame);
- void Write10MsData(int16_t *playout_buffer, uint16_t length_smpls);
+ void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
void Write10MsData(AudioFrame& audio_frame);
uint16_t PayloadLength10Ms() const;
diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc b/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
index b28c510..134d975 100644
--- a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
@@ -159,13 +159,13 @@
while (!_inFile.EndOfFile()) {
_inFile.Read10MsData(audioFrame);
- for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
+ for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
audioFrame.data_[n] = (int16_t) floor(
audioFrame.data_[n] * leftPanning + 0.5);
}
CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
- for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
+ for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
audioFrame.data_[n] = (int16_t) floor(
audioFrame.data_[n] * rightToLeftRatio + 0.5);
}
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
index ffbbc8c..0bac401 100644
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
@@ -32,9 +32,9 @@
namespace {
double FrameRms(AudioFrame& frame) {
- int samples = frame.num_channels_ * frame.samples_per_channel_;
+ size_t samples = frame.num_channels_ * frame.samples_per_channel_;
double rms = 0;
- for (int n = 0; n < samples; ++n)
+ for (size_t n = 0; n < samples; ++n)
rms += frame.data_[n] * frame.data_[n];
rms /= samples;
rms = sqrt(rms);
@@ -132,9 +132,9 @@
in_audio_frame.sample_rate_hz_ = codec.plfreq;
in_audio_frame.num_channels_ = codec.channels;
in_audio_frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms.
- int samples = in_audio_frame.num_channels_ *
+ size_t samples = in_audio_frame.num_channels_ *
in_audio_frame.samples_per_channel_;
- for (int n = 0; n < samples; ++n) {
+ for (size_t n = 0; n < samples; ++n) {
in_audio_frame.data_[n] = kAmp;
}
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index c61d25a..79124aa 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -270,14 +270,14 @@
if (loop_encode > 0) {
const int kMaxBytes = 1000; // Maximum number of bytes for one packet.
- int16_t bitstream_len_byte;
+ size_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
for (int i = 0; i < loop_encode; i++) {
int bitstream_len_byte_int = WebRtcOpus_Encode(
(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
&audio[read_samples], frame_length, kMaxBytes, bitstream);
ASSERT_GE(bitstream_len_byte_int, 0);
- bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
+ bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
@@ -341,7 +341,8 @@
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
// Write stand-alone speech to file.
- out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
+ out_file_standalone_.Write10MsData(
+ out_audio, static_cast<size_t>(decoded_samples) * channels);
if (audio_frame.timestamp_ > start_time_stamp) {
// Number of channels should be the same for both stand-alone and
