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WebRtcG722_Decode: Input array should be const uint8_t[]

Browse Source 
BUG=909
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38799004

Cr-Commit-Position: refs/heads/master@{#8224}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8224 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org 2015-02-02 08:58:03 +00:00
parent 026b892e72
commit a1dfbf1e5c
4 changed files with 15 additions and 18 deletions
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4
webrtc/modules/audio_coding/codecs/g722/g722_interface.c
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@ -86,7 +86,7 @@ int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
}
int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
int16_t *encoded,
const uint8_t *encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
@ -94,7 +94,7 @@ int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
// Decode the G.722 encoder stream
*speechType=G722_WEBRTC_SPEECH;
return WebRtc_g722_decode((G722DecoderState*) G722dec_inst,
decoded, (uint8_t*) encoded, len);
decoded, encoded, len);
}
int16_t WebRtcG722_Version(char *versionStr, short len)

2
webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
View File

@ -168,7 +168,7 @@ int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
*/
int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
int16_t *encoded,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);

5
webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
View File

@ -124,9 +124,8 @@ int main(int argc, char* argv[])
/* G.722 encoding + decoding */
stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
err = WebRtcG722_Decode(G722dec_inst,
reinterpret_cast<int16_t*>(streamdata),
stream_len, decoded, speechType);
err = WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
speechType);
/* Stop clock after call to encoder and decoder */
runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);

22
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
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@ -140,10 +140,9 @@ AudioDecoderG722::~AudioDecoderG722() {
int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG722_Decode(
dec_state_,
const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)),
static_cast<int16_t>(encoded_len), decoded, &temp_type);
int16_t ret =
WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
@ -176,16 +175,15 @@ int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
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_,
reinterpret_cast<int16_t*>(encoded_deinterleaved),
static_cast<int16_t>(encoded_len / 2), decoded, &temp_type);
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_,
reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]),
static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type);
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) {
decoded_len += ret;
// Interleave output.