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Revert "Upconvert various types to int."

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This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.

BUG=499241
TBR=hlundin

Review URL: https://codereview.webrtc.org/1179953003

Cr-Commit-Position: refs/heads/master@{#9418}
This commit is contained in:
Peter Kasting
2015-06-11 12:42:27 -07:00
parent 66a641a9c6
commit cb180976dd
61 changed files with 314 additions and 313 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -198,7 +198,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
CHECK_EQ(input_buffer_.size(),
static_cast<size_t>(num_10ms_frames_per_packet_) *
samples_per_10ms_frame_);
int status = WebRtcOpus_Encode(
int16_t status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
static_cast<int16_t>(num_channels_)),

26
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@ -64,11 +64,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
* Return value : >=0 - Length (in bytes) of coded data
* -1 - Error
*/
int WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
/****************************************************************************
* WebRtcOpus_SetBitRate(...)
@ -236,9 +236,9 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
* Return value : >0 - Samples per channel in decoded vector
* -1 - Error
*/
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DecodePlc(...)
@ -254,8 +254,8 @@ int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames);
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DecodeFec(...)
@ -275,9 +275,9 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
* 0 - No FEC data in the packet
* -1 - Error
*/
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
* WebRtcOpus_DurationEst(...)

10
webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -131,10 +131,10 @@ OpusFecTest::OpusFecTest()
}
void OpusFecTest::EncodeABlock() {
int value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
int16_t value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
EXPECT_GT(value, 0);
encoded_bytes_ = value;
@ -142,7 +142,7 @@ void OpusFecTest::EncodeABlock() {
void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
int16_t audio_type;
int value_1 = 0, value_2 = 0;
int16_t value_1 = 0, value_2 = 0;
if (lost_previous) {
// Decode previous frame.

26
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -78,11 +78,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
}
}
int WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int res;
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
@ -291,9 +291,9 @@ static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
return res;
}
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
if (encoded_bytes == 0) {
@ -318,8 +318,8 @@ int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
return decoded_samples;
}
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames) {
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
int16_t audio_type = 0;
int decoded_samples;
int plc_samples;
@ -339,9 +339,9 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
return decoded_samples;
}
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
int fec_samples;