webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc - Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView

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Unified Diff: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebase Created 5 years, 1 month ago

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Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

index eac7412178f671229420ba73b3951f08e3c7592e..3daf3f94e17a28baf028aabb3a635c5258c4ff7a 100644

--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

@@ -132,13 +132,13 @@ int AudioEncoderOpus::GetTargetBitrate() const {

AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(

uint32_t rtp_timestamp,

- const int16_t* audio,

+ rtc::ArrayView<const int16_t> audio,

size_t max_encoded_bytes,

uint8_t* encoded) {

if (input_buffer_.empty())

first_timestamp_in_buffer_ = rtp_timestamp;

- input_buffer_.insert(input_buffer_.end(), audio,

- audio + SamplesPer10msFrame());

+ RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());

+ input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());

if (input_buffer_.size() <

(static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {

return EncodedInfo();