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Unified Diff: webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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Index: webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 53dc0335a46c57bacd937ffbd27e115e40c92b98..769f0b0fa8f82c5df9476ae3e254c1c282eba382 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -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);
}
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