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

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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

78
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -53,10 +53,9 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
return static_cast<int>(ret);
}
int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
@ -85,10 +84,9 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
return static_cast<int>(ret);
}
int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
@ -120,10 +118,9 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
int16_t ret =
WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded);
size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
*speech_type = ConvertSpeechType(1);
return ret;
return static_cast<int>(ret);
}
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
@ -132,7 +129,7 @@ int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
return static_cast<int>(encoded_len / (2 * Channels()));
}
AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels)
AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(size_t num_channels)
: channels_(num_channels) {
DCHECK(num_channels > 0);
}
@ -163,14 +160,13 @@ int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
}
@ -204,11 +200,10 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 16000);
int16_t temp_type = 1; // Default is speech.
int16_t ret =
WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
size_t ret =
WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
return static_cast<int>(ret);
}
int AudioDecoderG722::Init() {
@ -246,29 +241,24 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
// Decode left and right.
int16_t ret = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
static_cast<int16_t>(encoded_len / 2),
decoded, &temp_type);
if (ret >= 0) {
int decoded_len = ret;
ret = WebRtcG722_Decode(dec_state_right_,
&encoded_deinterleaved[encoded_len / 2],
static_cast<int16_t>(encoded_len / 2),
&decoded[decoded_len], &temp_type);
if (ret == decoded_len) {
ret += decoded_len; // Return total number of samples.
// Interleave output.
for (int k = ret / 2; k < ret; k++) {
int16_t temp = decoded[k];
memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
(ret - k - 1) * sizeof(int16_t));
decoded[2 * k - ret + 1] = temp;
}
size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
encoded_len / 2, decoded, &temp_type);
size_t ret = WebRtcG722_Decode(
dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
encoded_len / 2, &decoded[decoded_len], &temp_type);
if (ret == decoded_len) {
ret += decoded_len; // Return total number of samples.
// Interleave output.
for (size_t k = ret / 2; k < ret; k++) {
int16_t temp = decoded[k];
memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
(ret - k - 1) * sizeof(int16_t));
decoded[2 * k - ret + 1] = temp;
}
}
*speech_type = ConvertSpeechType(temp_type);
delete [] encoded_deinterleaved;
return ret;
return static_cast<int>(ret);
}
size_t AudioDecoderG722Stereo::Channels() const {
@ -312,7 +302,8 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
// Opus
#ifdef WEBRTC_CODEC_OPUS
AudioDecoderOpus::AudioDecoderOpus(int num_channels) : channels_(num_channels) {
AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
: channels_(num_channels) {
DCHECK(num_channels == 1 || num_channels == 2);
WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
}
@ -328,8 +319,7 @@ int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcOpus_Decode(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
int ret = WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded,
&temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
@ -350,8 +340,7 @@ int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
&temp_type);
if (ret > 0)
ret *= static_cast<int>(channels_); // Return total number of samples.
@ -365,8 +354,7 @@ int AudioDecoderOpus::Init() {
int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return WebRtcOpus_DurationEst(dec_state_,
encoded, static_cast<int>(encoded_len));
return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
}
int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
@ -376,13 +364,13 @@ int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
return PacketDuration(encoded, encoded_len);
}
return WebRtcOpus_FecDurationEst(encoded, static_cast<int>(encoded_len));
return WebRtcOpus_FecDurationEst(encoded, encoded_len);
}
bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
int fec;
fec = WebRtcOpus_PacketHasFec(encoded, static_cast<int>(encoded_len));
fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
return (fec == 1);
}