webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc - Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView

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Unified Diff: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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/g711/audio_encoder_pcm.cc

diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

index dde3cc6799871dc1175d215858b1ca1cb6ca6fb2..6930e2c3751a28945174dd8b14049a87b3ecb962 100644

--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

@@ -88,16 +88,13 @@ int AudioEncoderPcm::GetTargetBitrate() const {

AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(

uint32_t rtp_timestamp,

- const int16_t* audio,

+ rtc::ArrayView<const int16_t> audio,

size_t max_encoded_bytes,

uint8_t* encoded) {

- const int num_samples = SampleRateHz() / 100 * NumChannels();

if (speech_buffer_.empty()) {

first_timestamp_in_buffer_ = rtp_timestamp;

}

- for (int i = 0; i < num_samples; ++i) {

- speech_buffer_.push_back(audio[i]);

- }

+ speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());

if (speech_buffer_.size() < full_frame_samples_) {

return EncodedInfo();

}