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. |