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Reland "Upconvert various types to int.", misc. codecs portion.

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This reverts portions of commit cb180976dd, which
reverted commit 83ad33a8ae.  Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded.

The original commit message is below:

Upconvert various types to int.

Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.

Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."

This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t).  Other locations will be converted to size_t in a separate change.

BUG=none
TBR=kwiberg

Review URL: https://codereview.webrtc.org/1179093003

Cr-Commit-Position: refs/heads/master@{#9424}
This commit is contained in:
Peter Kasting
2015-06-11 19:02:46 -07:00
parent a8b335c709
commit bba7807078
9 changed files with 56 additions and 60 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
View File

@@ -68,8 +68,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
* -1 - Error
*/
int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
int16_t quality);
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality);
int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
/****************************************************************************
@@ -103,9 +103,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID);
int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID);
/****************************************************************************
* WebRtcCng_UpdateSid(...)

18
webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
View File

@@ -36,7 +36,7 @@ typedef struct WebRtcCngDecoder_ {
typedef struct WebRtcCngEncoder_ {
int16_t enc_nrOfCoefs;
uint16_t enc_sampfreq;
int enc_sampfreq;
int16_t enc_interval;
int16_t enc_msSinceSID;
int32_t enc_Energy;
@@ -142,8 +142,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
int16_t quality) {
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality) {
int i;
WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
memset(inst, 0, sizeof(WebRtcCngEncoder));
@@ -227,9 +227,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst) {
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID) {
int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
int16_t nrOfSamples, uint8_t* SIDdata,
int16_t* bytesOut, int16_t forceSID) {
WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@@ -388,10 +388,12 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
inst->enc_msSinceSID = 0;
*bytesOut = inst->enc_nrOfCoefs + 1;
inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
inst->enc_msSinceSID +=
(int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
return inst->enc_nrOfCoefs + 1;
} else {
inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
inst->enc_msSinceSID +=
(int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
*bytesOut = 0;
return 0;
}

4
webrtc/modules/audio_coding/codecs/g722/g722_interface.c
View File

@@ -39,7 +39,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
}
}
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
{
// Free encoder memory
return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
@@ -79,7 +79,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
}
}
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
{
// Free encoder memory
return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);

4
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@@ -73,7 +73,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
@@ -142,7 +142,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
* -1 - Error
*/
int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
/****************************************************************************

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_);
int16_t status = WebRtcOpus_Encode(
int 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
*/
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
int 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
*/
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int 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 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames);
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames);
/****************************************************************************
* WebRtcOpus_DecodeFec(...)
@@ -275,9 +275,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
* 0 - No FEC data in the packet
* -1 - Error
*/
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type);
int 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() {
int16_t value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
int 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;
int16_t value_1 = 0, value_2 = 0;
int 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) {
}
}
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int 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;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int 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 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
return decoded_samples;
}
int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int16_t number_of_lost_frames) {
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int number_of_lost_frames) {
int16_t audio_type = 0;
int decoded_samples;
int plc_samples;
@@ -339,9 +339,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
return decoded_samples;
}
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int 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;

16
webrtc/modules/audio_coding/main/test/opus_test.cc
View File

@@ -273,17 +273,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
int16_t bitstream_len_byte;
uint8_t bitstream[kMaxBytes];
for (int i = 0; i < loop_encode; i++) {
if (channels == 1) {
bitstream_len_byte = WebRtcOpus_Encode(
opus_mono_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GE(bitstream_len_byte, 0);
} else {
bitstream_len_byte = WebRtcOpus_Encode(
opus_stereo_encoder_, &audio[read_samples],
frame_length, kMaxBytes, bitstream);
ASSERT_GE(bitstream_len_byte, 0);
}
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);
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.