webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc - Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView

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Unified Diff: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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/main/acm2/audio_coding_module_impl.cc

diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

index 260f8a8e67113678c7257ea275038f15a4750004..3b8b14015a203271eec9edabdee1b7c775556396 100644

--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

@@ -149,7 +149,9 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {

encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());

encoded_info = audio_encoder->Encode(

- rtp_timestamp, input_data.audio, input_data.length_per_channel,

+ rtp_timestamp, rtc::ArrayView<const int16_t>(

+ input_data.audio, input_data.audio_channel *

+ input_data.length_per_channel),

encode_buffer_.size(), encode_buffer_.data());

encode_buffer_.SetSize(encoded_info.encoded_bytes);

bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);