34efc689c2
This new lib contains the bare minimum to implement an iSAC encoder and decoder. The webrtc files have been copied from the revision as the existing imported files (c8b569e0a7ad0b369e15f0197b3a558699ec8efa).
56 lines
1.7 KiB
C++
56 lines
1.7 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/base/checks.h"
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namespace webrtc {
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AudioEncoder::EncodedInfo::EncodedInfo() = default;
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AudioEncoder::EncodedInfo::~EncodedInfo() = default;
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int AudioEncoder::RtpTimestampRateHz() const {
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return SampleRateHz();
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}
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AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t num_samples_per_channel,
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size_t max_encoded_bytes,
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uint8_t* encoded) {
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RTC_CHECK_EQ(num_samples_per_channel,
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static_cast<size_t>(SampleRateHz() / 100));
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EncodedInfo info =
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EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
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RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
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return info;
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}
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bool AudioEncoder::SetFec(bool enable) {
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return !enable;
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}
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bool AudioEncoder::SetDtx(bool enable) {
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return !enable;
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}
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bool AudioEncoder::SetApplication(Application application) {
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return false;
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}
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void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
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void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
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void AudioEncoder::SetTargetBitrate(int target_bps) {}
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} // namespace webrtc
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