blob: 63639860f45715662c49ef2c9822569bbf356412 [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/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
namespace webrtc {
struct CodecInst;
class AudioEncoderIlbc final : public AudioEncoder {
struct Config {
bool IsOk() const;
int payload_type = 102;
int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms.
// Note that frame size 40 ms produces encodings with two 20 ms frames in
// them, and frame size 60 ms consists of two 30 ms frames.
explicit AudioEncoderIlbc(const Config& config);
explicit AudioEncoderIlbc(const CodecInst& codec_inst);
~AudioEncoderIlbc() override;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
void Reset() override;
size_t RequiredOutputSizeBytes() const;
static const size_t kMaxSamplesPerPacket = 480;
const Config config_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
int16_t input_buffer_[kMaxSamplesPerPacket];
IlbcEncoderInstance* encoder_;
} // namespace webrtc