blob: 388b0ff61c6d2b9d02ccebccd7779a329a52bb69 [file] [log] [blame]
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/base/checks.h"
namespace webrtc {
AudioEncoder::EncodedInfo::EncodedInfo() = default;
AudioEncoder::EncodedInfo::~EncodedInfo() = default;
int AudioEncoder::RtpTimestampRateHz() const {
return SampleRateHz();
AudioEncoder::EncodedInfo AudioEncoder::Encode(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
EncodedInfo info =
EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
return info;
bool AudioEncoder::SetFec(bool enable) {
return !enable;
bool AudioEncoder::SetDtx(bool enable) {
return !enable;
bool AudioEncoder::SetApplication(Application application) {
return false;
void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
void AudioEncoder::SetTargetBitrate(int target_bps) {}
} // namespace webrtc