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/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/time_delta.h"
namespace webrtc {
class AudioEncoderPcm : public AudioEncoder {
public:
struct Config {
public:
bool IsOk() const;
int frame_size_ms;
size_t num_channels;
int payload_type;
protected:
explicit Config(int pt)
: frame_size_ms(20), num_channels(1), payload_type(pt) {}
};
~AudioEncoderPcm() override;
int SampleRateHz() const override;
size_t NumChannels() 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;
protected:
AudioEncoderPcm(const Config& config, int sample_rate_hz);
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
virtual size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) = 0;
virtual size_t BytesPerSample() const = 0;
// Used to set EncodedInfoLeaf::encoder_type in
// AudioEncoderPcm::EncodeImpl
virtual AudioEncoder::CodecType GetCodecType() const = 0;
private:
const int sample_rate_hz_;
const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
const size_t full_frame_samples_;
std::vector<int16_t> speech_buffer_;
uint32_t first_timestamp_in_buffer_;
};
class AudioEncoderPcmA final : public AudioEncoderPcm {
public:
struct Config : public AudioEncoderPcm::Config {
Config() : AudioEncoderPcm::Config(8) {}
};
explicit AudioEncoderPcmA(const Config& config)
: AudioEncoderPcm(config, kSampleRateHz) {}
AudioEncoderPcmA(const AudioEncoderPcmA&) = delete;
AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete;
protected:
size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
size_t BytesPerSample() const override;
AudioEncoder::CodecType GetCodecType() const override;
private:
static const int kSampleRateHz = 8000;
};
class AudioEncoderPcmU final : public AudioEncoderPcm {
public:
struct Config : public AudioEncoderPcm::Config {
Config() : AudioEncoderPcm::Config(0) {}
};
explicit AudioEncoderPcmU(const Config& config)
: AudioEncoderPcm(config, kSampleRateHz) {}
AudioEncoderPcmU(const AudioEncoderPcmU&) = delete;
AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete;
protected:
size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
size_t BytesPerSample() const override;
AudioEncoder::CodecType GetCodecType() const override;
private:
static const int kSampleRateHz = 8000;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_