zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Pass audio to AudioEncoder::Encode() in an ArrayView

Browse Source 
Instead of in separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1418423010

Cr-Commit-Position: refs/heads/master@{#10535}
This commit is contained in:
kwiberg
2015-11-06 01:21:35 -08:00
committed by Commit bot
parent 1a4e9d7572
commit 288886b2ec
32 changed files with 183 additions and 142 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
View File

@ -97,7 +97,7 @@ int AudioEncoderCng::GetTargetBitrate() const {
AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
uint32_t rtp_timestamp,
const int16_t* audio,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_CHECK_GE(max_encoded_bytes,
@ -106,9 +106,8 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
RTC_CHECK_EQ(speech_buffer_.size(),
rtp_timestamps_.size() * samples_per_10ms_frame);
rtp_timestamps_.push_back(rtp_timestamp);
for (size_t i = 0; i < samples_per_10ms_frame; ++i) {
speech_buffer_.push_back(audio[i]);
}
RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
if (rtp_timestamps_.size() < frames_to_encode) {
return EncodedInfo();
@ -242,9 +241,12 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
const size_t samples_per_10ms_frame = SamplesPer10msFrame();
AudioEncoder::EncodedInfo info;
for (size_t i = 0; i < frames_to_encode; ++i) {
info = speech_encoder_->Encode(
rtp_timestamps_.front(), &speech_buffer_[i * samples_per_10ms_frame],
samples_per_10ms_frame, max_encoded_bytes, encoded);
info =
speech_encoder_->Encode(rtp_timestamps_.front(),
rtc::ArrayView<const int16_t>(
&speech_buffer_[i * samples_per_10ms_frame],
samples_per_10ms_frame),
max_encoded_bytes, encoded);
if (i + 1 == frames_to_encode) {
RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
} else {

6
webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -75,8 +75,10 @@ class AudioEncoderCngTest : public ::testing::Test {
void Encode() {
ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_,
encoded_.size(), &encoded_[0]);
encoded_info_ = cng_->Encode(
timestamp_,
rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
encoded_.size(), &encoded_[0]);
timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
}

2
webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
View File

@ -57,7 +57,7 @@ class AudioEncoderCng final : public AudioEncoder {
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;