mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-24 16:57:50 +00:00
d83a3d71bc
Merge in RedPhone // FREEBIE
153 lines
4.9 KiB
C++
153 lines
4.9 KiB
C++
/*
|
|
* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
|
|
|
|
#include <stdlib.h> // NULL
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
enum NetEqDecoder {
|
|
kDecoderPCMu,
|
|
kDecoderPCMa,
|
|
kDecoderPCMu_2ch,
|
|
kDecoderPCMa_2ch,
|
|
kDecoderILBC,
|
|
kDecoderISAC,
|
|
kDecoderISACswb,
|
|
kDecoderISACfb,
|
|
kDecoderPCM16B,
|
|
kDecoderPCM16Bwb,
|
|
kDecoderPCM16Bswb32kHz,
|
|
kDecoderPCM16Bswb48kHz,
|
|
kDecoderPCM16B_2ch,
|
|
kDecoderPCM16Bwb_2ch,
|
|
kDecoderPCM16Bswb32kHz_2ch,
|
|
kDecoderPCM16Bswb48kHz_2ch,
|
|
kDecoderPCM16B_5ch,
|
|
kDecoderG722,
|
|
kDecoderG722_2ch,
|
|
kDecoderRED,
|
|
kDecoderAVT,
|
|
kDecoderCNGnb,
|
|
kDecoderCNGwb,
|
|
kDecoderCNGswb32kHz,
|
|
kDecoderCNGswb48kHz,
|
|
kDecoderArbitrary,
|
|
kDecoderOpus,
|
|
kDecoderOpus_2ch,
|
|
kDecoderCELT_32,
|
|
kDecoderCELT_32_2ch,
|
|
};
|
|
|
|
// This is the interface class for decoders in NetEQ. Each codec type will have
|
|
// and implementation of this class.
|
|
class AudioDecoder {
|
|
public:
|
|
enum SpeechType {
|
|
kSpeech = 1,
|
|
kComfortNoise = 2
|
|
};
|
|
|
|
// Used by PacketDuration below. Save the value -1 for errors.
|
|
enum { kNotImplemented = -2 };
|
|
|
|
explicit AudioDecoder(enum NetEqDecoder type)
|
|
: codec_type_(type),
|
|
channels_(1),
|
|
state_(NULL) {
|
|
}
|
|
|
|
virtual ~AudioDecoder() {}
|
|
|
|
// Decodes |encode_len| bytes from |encoded| and writes the result in
|
|
// |decoded|. The number of samples from all channels produced is in
|
|
// the return value. If the decoder produced comfort noise, |speech_type|
|
|
// is set to kComfortNoise, otherwise it is kSpeech.
|
|
virtual int Decode(const uint8_t* encoded, size_t encoded_len,
|
|
int16_t* decoded, SpeechType* speech_type) = 0;
|
|
|
|
// Same as Decode(), but interfaces to the decoders redundant decode function.
|
|
// The default implementation simply calls the regular Decode() method.
|
|
virtual int DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
|
|
int16_t* decoded, SpeechType* speech_type);
|
|
|
|
// Indicates if the decoder implements the DecodePlc method.
|
|
virtual bool HasDecodePlc() const;
|
|
|
|
// Calls the packet-loss concealment of the decoder to update the state after
|
|
// one or several lost packets.
|
|
virtual int DecodePlc(int num_frames, int16_t* decoded);
|
|
|
|
// Initializes the decoder.
|
|
virtual int Init() = 0;
|
|
|
|
// Notifies the decoder of an incoming packet to NetEQ.
|
|
virtual int IncomingPacket(const uint8_t* payload,
|
|
size_t payload_len,
|
|
uint16_t rtp_sequence_number,
|
|
uint32_t rtp_timestamp,
|
|
uint32_t arrival_timestamp);
|
|
|
|
// Returns the last error code from the decoder.
|
|
virtual int ErrorCode();
|
|
|
|
// Returns the duration in samples of the payload in |encoded| which is
|
|
// |encoded_len| bytes long. Returns kNotImplemented if no duration estimate
|
|
// is available, or -1 in case of an error.
|
|
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len);
|
|
|
|
// Returns the duration in samples of the redandant payload in |encoded| which
|
|
// is |encoded_len| bytes long. Returns kNotImplemented if no duration
|
|
// estimate is available, or -1 in case of an error.
|
|
virtual int PacketDurationRedundant(const uint8_t* encoded,
|
|
size_t encoded_len) const;
|
|
|
|
// Detects whether a packet has forward error correction. The packet is
|
|
// comprised of the samples in |encoded| which is |encoded_len| bytes long.
|
|
// Returns true if the packet has FEC and false otherwise.
|
|
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
|
|
|
|
virtual NetEqDecoder codec_type() const;
|
|
|
|
// Returns the underlying decoder state.
|
|
void* state() { return state_; }
|
|
|
|
// Returns true if |codec_type| is supported.
|
|
static bool CodecSupported(NetEqDecoder codec_type);
|
|
|
|
// Returns the sample rate for |codec_type|.
|
|
static int CodecSampleRateHz(NetEqDecoder codec_type);
|
|
|
|
// Creates an AudioDecoder object of type |codec_type|. Returns NULL for
|
|
// for unsupported codecs, and when creating an AudioDecoder is not
|
|
// applicable (e.g., for RED and DTMF/AVT types).
|
|
static AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type);
|
|
|
|
size_t channels() const { return channels_; }
|
|
|
|
protected:
|
|
static SpeechType ConvertSpeechType(int16_t type);
|
|
|
|
enum NetEqDecoder codec_type_;
|
|
size_t channels_;
|
|
void* state_;
|
|
|
|
private:
|
|
DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
|