blob: ae82509ef44b0f0efebd751c6019c4be1631778e [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() : EncodedInfoLeaf() {
}
AudioEncoder::EncodedInfo::~EncodedInfo() {
}
bool AudioEncoder::Encode(uint32_t rtp_timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(SampleRateHz() / 100));
bool ret =
EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info);
CHECK_LE(info->encoded_bytes, max_encoded_bytes);
return ret;
}
int AudioEncoder::RtpTimestampRateHz() const {
return SampleRateHz();
}
} // namespace webrtc