blob: c836503f2bec0f6664d3407d964eae4e4a32718e [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 <memory>
#include <utility>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/units/time_delta.h"
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class AudioEncoderG722Impl final : public AudioEncoder {
AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
~AudioEncoderG722Impl() override;
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void Reset() override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
rtc::Buffer encoded_buffer; // Already encoded.
size_t SamplesPerChannel() const;
const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const std::unique_ptr<EncoderState[]> encoders_;
rtc::Buffer interleave_buffer_;
} // namespace webrtc