Chromium Code Reviews| Index: webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |
| diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |
| index 065dc06817237bfab15a55caf6a1661a68173d4e..7b497fdebe1849dd3e80ac48ce2e0a006eed1ceb 100644 |
| --- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |
| +++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |
| @@ -10,7 +10,7 @@ |
| #include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h" |
| -#include <cstring> |
| +#include <algorithm> |
| #include <limits> |
| #include "webrtc/base/checks.h" |
| #include "webrtc/common_types.h" |
| @@ -91,7 +91,7 @@ int AudioEncoderIlbc::GetTargetBitrate() const { |
| AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal( |
| uint32_t rtp_timestamp, |
| - const int16_t* audio, |
| + rtc::ArrayView<const int16_t> audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded) { |
| RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes()); |
| @@ -101,9 +101,9 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal( |
| first_timestamp_in_buffer_ = rtp_timestamp; |
| // Buffer input. |
| - std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_, |
| - audio, |
| - kSampleRateHz / 100 * sizeof(audio[0])); |
| + RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size()); |
| + std::copy(audio.cbegin(), audio.cend(), |
|
the sun
2015/11/04 12:32:18
What does this compile to? Is std::copy special ca
kwiberg-webrtc
2015/11/04 14:58:11
I assume so. [...] OK, I checked. Clang compiles t
|
| + input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_); |
| // If we don't yet have enough buffered input for a whole packet, we're done |
| // for now. |