Chromium Code Reviews| 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 99ff95a2ec1378d65bcdaa0bfa383e9f29697961..6a9b953f23a3a9d9db5f683a3820afebb9b49a2e 100644 |
| --- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| @@ -256,16 +256,14 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded, |
| static_cast<int16_t>(encoded_len / 2), |
| &decoded[decoded_len], &temp_type); |
| if (ret == decoded_len) { |
| - decoded_len += ret; |
| + ret += decoded_len; // Return total number of samples. |
| // Interleave output. |
| - for (int k = decoded_len / 2; k < decoded_len; k++) { |
| + for (int k = ret / 2; k < ret; k++) { |
| int16_t temp = decoded[k]; |
| - memmove(&decoded[2 * k - decoded_len + 2], |
| - &decoded[2 * k - decoded_len + 1], |
| - (decoded_len - k - 1) * sizeof(int16_t)); |
| - decoded[2 * k - decoded_len + 1] = temp; |
| + memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1], |
| + (ret - k - 1) * sizeof(int16_t)); |
| + decoded[2 * k - ret + 1] = temp; |
| } |
|
kwiberg-webrtc
2015/07/12 18:40:30
Hmm. Maybe better, but still messy. How about usin
Peter Kasting
2015/07/13 02:41:24
I don't think that makes life better, due to the w
|
| - ret = decoded_len; // Return total number of samples. |
| } |
| } |
| *speech_type = ConvertSpeechType(temp_type); |