Address Windows 64-bits warnings.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2203004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4803 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2013-09-20 16:25:28 +00:00
parent 0e63e76781
commit 362a55e7b0
30 changed files with 179 additions and 170 deletions

View File

@ -231,7 +231,8 @@ class AudioDecoderPcmUTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
int enc_len_bytes =
WebRtcG711_EncodeU(NULL, const_cast<int16_t*>(input), input_len_samples,
WebRtcG711_EncodeU(NULL, const_cast<int16_t*>(input),
static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
return enc_len_bytes;
@ -250,7 +251,8 @@ class AudioDecoderPcmATest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
int enc_len_bytes =
WebRtcG711_EncodeA(NULL, const_cast<int16_t*>(input), input_len_samples,
WebRtcG711_EncodeA(NULL, const_cast<int16_t*>(input),
static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
return enc_len_bytes;
@ -269,7 +271,7 @@ class AudioDecoderPcm16BTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
int enc_len_bytes = WebRtcPcm16b_EncodeW16(
const_cast<int16_t*>(input), input_len_samples,
const_cast<int16_t*>(input), static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(2 * input_len_samples, static_cast<size_t>(enc_len_bytes));
return enc_len_bytes;
@ -297,7 +299,8 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
int enc_len_bytes =
WebRtcIlbcfix_Encode(encoder_, input, input_len_samples,
WebRtcIlbcfix_Encode(encoder_, input,
static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(50, enc_len_bytes);
return enc_len_bytes;
@ -475,7 +478,7 @@ class AudioDecoderG722Test : public AudioDecoderTest {
uint8_t* output) {
int enc_len_bytes =
WebRtcG722_Encode(encoder_, const_cast<int16_t*>(input),
input_len_samples,
static_cast<int>(input_len_samples),
reinterpret_cast<int16_t*>(output));
EXPECT_EQ(80, enc_len_bytes);
return enc_len_bytes;
@ -545,15 +548,17 @@ class AudioDecoderOpusTest : public AudioDecoderTest {
// Upsample from 32 to 48 kHz.
Resampler rs;
rs.Reset(32000, 48000, kResamplerSynchronous);
const int max_resamp_len_samples = input_len_samples * 3 / 2;
const int max_resamp_len_samples = static_cast<int>(input_len_samples) *
3 / 2;
int16_t* resamp_input = new int16_t[max_resamp_len_samples];
int resamp_len_samples;
EXPECT_EQ(0, rs.Push(input, input_len_samples, resamp_input,
max_resamp_len_samples, resamp_len_samples));
EXPECT_EQ(0, rs.Push(input, static_cast<int>(input_len_samples),
resamp_input, max_resamp_len_samples,
resamp_len_samples));
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
int enc_len_bytes =
WebRtcOpus_Encode(encoder_, resamp_input,
resamp_len_samples, data_length_, output);
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples,
static_cast<int>(data_length_), output);
EXPECT_GT(enc_len_bytes, 0);
delete [] resamp_input;
return enc_len_bytes;
@ -582,7 +587,7 @@ class AudioDecoderOpusStereoTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
// Create stereo by duplicating each sample in |input|.
const int input_stereo_samples = input_len_samples * 2;
const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
int16_t* input_stereo = new int16_t[input_stereo_samples];
for (size_t i = 0; i < input_len_samples; i++)
input_stereo[i * 2] = input_stereo[i * 2 + 1] = input[i];
@ -597,7 +602,7 @@ class AudioDecoderOpusStereoTest : public AudioDecoderTest {
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
int enc_len_bytes =
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples / 2,
data_length_, output);
static_cast<int16_t>(data_length_), output);
EXPECT_GT(enc_len_bytes, 0);
delete [] resamp_input;
delete [] input_stereo;