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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 WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
namespace webrtc {
class CriticalSectionWrapper;
template <typename T>
class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
public:
// For constructing an encoder in instantaneous mode. Allowed combinations
// are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
// - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct Config {
Config();
bool IsOk() const;
int payload_type;
int sample_rate_hz;
int frame_size_ms;
int bit_rate; // Limit on the short-term average bit rate, in bits/second.
int max_bit_rate;
int max_payload_size_bytes;
};
// For constructing an encoder in channel-adaptive mode. Allowed combinations
// are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
// - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct ConfigAdaptive {
ConfigAdaptive();
bool IsOk() const;
int payload_type;
int sample_rate_hz;
int initial_frame_size_ms;
int initial_bit_rate;
int max_bit_rate;
bool enforce_frame_size; // Prevent adaptive changes to the frame size?
int max_payload_size_bytes;
};
explicit AudioEncoderDecoderIsacT(const Config& config);
explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
~AudioEncoderDecoderIsacT() override;
// AudioEncoder public methods.
int SampleRateHz() const override;
int NumChannels() const override;
size_t MaxEncodedBytes() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
// AudioDecoder methods.
bool HasDecodePlc() const override;
int DecodePlc(int num_frames, int16_t* decoded) override;
int Init() override;
int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) override;
int ErrorCode() override;
size_t Channels() const override { return 1; }
protected:
// AudioEncoder protected method.
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
// AudioDecoder protected method.
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
// STREAM_MAXW16_60MS for iSAC fix (60 ms).
static const size_t kSufficientEncodeBufferSizeBytes = 400;
const int payload_type_;
// iSAC encoder/decoder state, guarded by a mutex to ensure that encode calls
// from one thread won't clash with decode calls from another thread.
// Note: PT_GUARDED_BY is disabled since it is not yet supported by clang.
const rtc::scoped_ptr<CriticalSectionWrapper> state_lock_;
typename T::instance_type* isac_state_
GUARDED_BY(state_lock_) /* PT_GUARDED_BY(lock_)*/;
int decoder_sample_rate_hz_ GUARDED_BY(state_lock_);
// Must be acquired before state_lock_.
const rtc::scoped_ptr<CriticalSectionWrapper> lock_;
// Have we accepted input but not yet emitted it in a packet?
bool packet_in_progress_ GUARDED_BY(lock_);
// Timestamp of the first input of the currently in-progress packet.
uint32_t packet_timestamp_ GUARDED_BY(lock_);
// Timestamp of the previously encoded packet.
uint32_t last_encoded_timestamp_ GUARDED_BY(lock_);
DISALLOW_COPY_AND_ASSIGN(AudioEncoderDecoderIsacT);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
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