audio_coding: rename interface -> include
BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
This commit is contained in:
Henrik Kjellander
@ -0,0 +1,51 @@
|
||||
/*
|
||||
* Copyright (c) 2015 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_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
|
||||
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
|
||||
|
||||
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
|
||||
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AudioDecoderOpus final : public AudioDecoder {
|
||||
public:
|
||||
explicit AudioDecoderOpus(size_t num_channels);
|
||||
~AudioDecoderOpus() override;
|
||||
|
||||
void Reset() override;
|
||||
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
|
||||
int PacketDurationRedundant(const uint8_t* encoded,
|
||||
size_t encoded_len) const override;
|
||||
bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
|
||||
size_t Channels() const override;
|
||||
|
||||
protected:
|
||||
int DecodeInternal(const uint8_t* encoded,
|
||||
size_t encoded_len,
|
||||
int sample_rate_hz,
|
||||
int16_t* decoded,
|
||||
SpeechType* speech_type) override;
|
||||
int DecodeRedundantInternal(const uint8_t* encoded,
|
||||
size_t encoded_len,
|
||||
int sample_rate_hz,
|
||||
int16_t* decoded,
|
||||
SpeechType* speech_type) override;
|
||||
|
||||
private:
|
||||
OpusDecInst* dec_state_;
|
||||
const size_t channels_;
|
||||
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_DECODER_OPUS_H
|
||||
@ -0,0 +1,102 @@
|
||||
/*
|
||||
* 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_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
|
||||
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
|
||||
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/base/constructormagic.h"
|
||||
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
|
||||
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
struct CodecInst;
|
||||
|
||||
class AudioEncoderOpus final : public AudioEncoder {
|
||||
public:
|
||||
enum ApplicationMode {
|
||||
kVoip = 0,
|
||||
kAudio = 1,
|
||||
};
|
||||
|
||||
struct Config {
|
||||
bool IsOk() const;
|
||||
int frame_size_ms = 20;
|
||||
int num_channels = 1;
|
||||
int payload_type = 120;
|
||||
ApplicationMode application = kVoip;
|
||||
int bitrate_bps = 64000;
|
||||
bool fec_enabled = false;
|
||||
int max_playback_rate_hz = 48000;
|
||||
int complexity = kDefaultComplexity;
|
||||
bool dtx_enabled = false;
|
||||
|
||||
private:
|
||||
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
|
||||
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
|
||||
// default, to save encoder complexity.
|
||||
static const int kDefaultComplexity = 5;
|
||||
#else
|
||||
static const int kDefaultComplexity = 9;
|
||||
#endif
|
||||
};
|
||||
|
||||
explicit AudioEncoderOpus(const Config& config);
|
||||
explicit AudioEncoderOpus(const CodecInst& codec_inst);
|
||||
~AudioEncoderOpus() override;
|
||||
|
||||
size_t MaxEncodedBytes() const override;
|
||||
int SampleRateHz() const override;
|
||||
int NumChannels() const override;
|
||||
size_t Num10MsFramesInNextPacket() const override;
|
||||
size_t Max10MsFramesInAPacket() const override;
|
||||
int GetTargetBitrate() const override;
|
||||
|
||||
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
||||
const int16_t* audio,
|
||||
size_t max_encoded_bytes,
|
||||
uint8_t* encoded) override;
|
||||
|
||||
void Reset() override;
|
||||
bool SetFec(bool enable) override;
|
||||
|
||||
// Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
|
||||
// being inactive. During that, it still sends 2 packets (one for content, one
|
||||
// for signaling) about every 400 ms.
|
||||
bool SetDtx(bool enable) override;
|
||||
|
||||
bool SetApplication(Application application) override;
|
||||
void SetMaxPlaybackRate(int frequency_hz) override;
|
||||
void SetProjectedPacketLossRate(double fraction) override;
|
||||
void SetTargetBitrate(int target_bps) override;
|
||||
|
||||
// Getters for testing.
|
||||
double packet_loss_rate() const { return packet_loss_rate_; }
|
||||
ApplicationMode application() const { return config_.application; }
|
||||
bool dtx_enabled() const { return config_.dtx_enabled; }
|
||||
|
||||
private:
|
||||
int Num10msFramesPerPacket() const;
|
||||
int SamplesPer10msFrame() const;
|
||||
bool RecreateEncoderInstance(const Config& config);
|
||||
|
||||
Config config_;
|
||||
double packet_loss_rate_;
|
||||
std::vector<int16_t> input_buffer_;
|
||||
OpusEncInst* inst_;
|
||||
uint32_t first_timestamp_in_buffer_;
|
||||
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
|
||||
349
webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h
Normal file
349
webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h
Normal file
@ -0,0 +1,349 @@
|
||||
/*
|
||||
* 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_CODECS_OPUS_INCLUDE_OPUS_INTERFACE_H_
|
||||
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INTERFACE_H_
|
||||
|
||||
#include <stddef.h>
|
||||
|
||||
#include "webrtc/typedefs.h"
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
// Opaque wrapper types for the codec state.
|
||||
typedef struct WebRtcOpusEncInst OpusEncInst;
|
||||
typedef struct WebRtcOpusDecInst OpusDecInst;
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_EncoderCreate(...)
|
||||
*
|
||||
* This function create an Opus encoder.
|
||||
*
|
||||
* Input:
|
||||
* - channels : number of channels.
|
||||
* - application : 0 - VOIP applications.
|
||||
* Favor speech intelligibility.
|
||||
* 1 - Audio applications.
|
||||
* Favor faithfulness to the original input.
|
||||
*
|
||||
* Output:
|
||||
* - inst : a pointer to Encoder context that is created
|
||||
* if success.
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
|
||||
int32_t channels,
|
||||
int32_t application);
|
||||
|
||||
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_Encode(...)
|
||||
*
|
||||
* This function encodes audio as a series of Opus frames and inserts
|
||||
* it into a packet. Input buffer can be any length.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
* - audio_in : Input speech data buffer
|
||||
* - samples : Samples per channel in audio_in
|
||||
* - length_encoded_buffer : Output buffer size
|
||||
*
|
||||
* Output:
|
||||
* - encoded : Output compressed data buffer
|
||||
*
|
||||
* Return value : >=0 - Length (in bytes) of coded data
|
||||
* -1 - Error
|
||||
*/
|
||||
int WebRtcOpus_Encode(OpusEncInst* inst,
|
||||
const int16_t* audio_in,
|
||||
size_t samples,
|
||||
size_t length_encoded_buffer,
|
||||
uint8_t* encoded);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_SetBitRate(...)
|
||||
*
|
||||
* This function adjusts the target bitrate of the encoder.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
* - rate : New target bitrate
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_SetPacketLossRate(...)
|
||||
*
|
||||
* This function configures the encoder's expected packet loss percentage.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
* - loss_rate : loss percentage in the range 0-100, inclusive.
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_SetMaxPlaybackRate(...)
|
||||
*
|
||||
* Configures the maximum playback rate for encoding. Due to hardware
|
||||
* limitations, the receiver may render audio up to a playback rate. Opus
|
||||
* encoder can use this information to optimize for network usage and encoding
|
||||
* complexity. This will affect the audio bandwidth in the coded audio. However,
|
||||
* the input/output sample rate is not affected.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
* - frequency_hz : Maximum playback rate in Hz.
|
||||
* This parameter can take any value. The relation
|
||||
* between the value and the Opus internal mode is
|
||||
* as following:
|
||||
* frequency_hz <= 8000 narrow band
|
||||
* 8000 < frequency_hz <= 12000 medium band
|
||||
* 12000 < frequency_hz <= 16000 wide band
|
||||
* 16000 < frequency_hz <= 24000 super wide band
|
||||
* frequency_hz > 24000 full band
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
|
||||
|
||||
/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
|
||||
* is needed. It might not be very useful since there are not many use cases and
|
||||
* the caller can always maintain the states. */
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_EnableFec()
|
||||
*
|
||||
* This function enables FEC for encoding.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DisableFec()
|
||||
*
|
||||
* This function disables FEC for encoding.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_EnableDtx()
|
||||
*
|
||||
* This function enables Opus internal DTX for encoding.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DisableDtx()
|
||||
*
|
||||
* This function disables Opus internal DTX for encoding.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
|
||||
|
||||
/*
|
||||
* WebRtcOpus_SetComplexity(...)
|
||||
*
|
||||
* This function adjusts the computational complexity. The effect is the same as
|
||||
* calling the complexity setting of Opus as an Opus encoder related CTL.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Encoder context
|
||||
* - complexity : New target complexity (0-10, inclusive)
|
||||
*
|
||||
* Return value : 0 - Success
|
||||
* -1 - Error
|
||||
*/
|
||||
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
|
||||
|
||||
int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
|
||||
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DecoderChannels(...)
|
||||
*
|
||||
* This function returns the number of channels created for Opus decoder.
|
||||
*/
|
||||
int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DecoderInit(...)
|
||||
*
|
||||
* This function resets state of the decoder.
|
||||
*
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
*/
|
||||
void WebRtcOpus_DecoderInit(OpusDecInst* inst);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_Decode(...)
|
||||
*
|
||||
* This function decodes an Opus packet into one or more audio frames at the
|
||||
* ACM interface's sampling rate (32 kHz).
|
||||
*
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
* - encoded : Encoded data
|
||||
* - encoded_bytes : Bytes in encoded vector
|
||||
*
|
||||
* Output:
|
||||
* - decoded : The decoded vector
|
||||
* - audio_type : 1 normal, 2 CNG (for Opus it should
|
||||
* always return 1 since we're not using Opus's
|
||||
* built-in DTX/CNG scheme)
|
||||
*
|
||||
* Return value : >0 - Samples per channel in decoded vector
|
||||
* -1 - Error
|
||||
*/
|
||||
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
|
||||
size_t encoded_bytes, int16_t* decoded,
|
||||
int16_t* audio_type);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DecodePlc(...)
|
||||
*
|
||||
* This function processes PLC for opus frame(s).
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
* - number_of_lost_frames : Number of PLC frames to produce
|
||||
*
|
||||
* Output:
|
||||
* - decoded : The decoded vector
|
||||
*
|
||||
* Return value : >0 - number of samples in decoded PLC vector
|
||||
* -1 - Error
|
||||
*/
|
||||
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
|
||||
int number_of_lost_frames);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DecodeFec(...)
|
||||
*
|
||||
* This function decodes the FEC data from an Opus packet into one or more audio
|
||||
* frames at the ACM interface's sampling rate (32 kHz).
|
||||
*
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
* - encoded : Encoded data
|
||||
* - encoded_bytes : Bytes in encoded vector
|
||||
*
|
||||
* Output:
|
||||
* - decoded : The decoded vector (previous frame)
|
||||
*
|
||||
* Return value : >0 - Samples per channel in decoded vector
|
||||
* 0 - No FEC data in the packet
|
||||
* -1 - Error
|
||||
*/
|
||||
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
|
||||
size_t encoded_bytes, int16_t* decoded,
|
||||
int16_t* audio_type);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_DurationEst(...)
|
||||
*
|
||||
* This function calculates the duration of an opus packet.
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
* - payload : Encoded data pointer
|
||||
* - payload_length_bytes : Bytes of encoded data
|
||||
*
|
||||
* Return value : The duration of the packet, in samples per
|
||||
* channel.
|
||||
*/
|
||||
int WebRtcOpus_DurationEst(OpusDecInst* inst,
|
||||
const uint8_t* payload,
|
||||
size_t payload_length_bytes);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_PlcDuration(...)
|
||||
*
|
||||
* This function calculates the duration of a frame returned by packet loss
|
||||
* concealment (PLC).
|
||||
*
|
||||
* Input:
|
||||
* - inst : Decoder context
|
||||
*
|
||||
* Return value : The duration of a frame returned by PLC, in
|
||||
* samples per channel.
|
||||
*/
|
||||
int WebRtcOpus_PlcDuration(OpusDecInst* inst);
|
||||
|
||||
/* TODO(minyue): Check whether it is needed to add a decoder context to the
|
||||
* arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
|
||||
* the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used.
|
||||
* So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_FecDurationEst(...)
|
||||
*
|
||||
* This function calculates the duration of the FEC data within an opus packet.
|
||||
* Input:
|
||||
* - payload : Encoded data pointer
|
||||
* - payload_length_bytes : Bytes of encoded data
|
||||
*
|
||||
* Return value : >0 - The duration of the FEC data in the
|
||||
* packet in samples per channel.
|
||||
* 0 - No FEC data in the packet.
|
||||
*/
|
||||
int WebRtcOpus_FecDurationEst(const uint8_t* payload,
|
||||
size_t payload_length_bytes);
|
||||
|
||||
/****************************************************************************
|
||||
* WebRtcOpus_PacketHasFec(...)
|
||||
*
|
||||
* This function detects if an opus packet has FEC.
|
||||
* Input:
|
||||
* - payload : Encoded data pointer
|
||||
* - payload_length_bytes : Bytes of encoded data
|
||||
*
|
||||
* Return value : 0 - the packet does NOT contain FEC.
|
||||
* 1 - the packet contains FEC.
|
||||
*/
|
||||
int WebRtcOpus_PacketHasFec(const uint8_t* payload,
|
||||
size_t payload_length_bytes);
|
||||
|
||||
#ifdef __cplusplus
|
||||
} // extern "C"
|
||||
#endif
|
||||
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_OPUS_INCLUDE_H_
|
||||
Reference in New Issue
Block a user