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林若思 2026-02-12 14:59:50 +08:00
parent 3c9ec050c4
commit a894f22ae5
22 changed files with 4477 additions and 184 deletions
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29
app/CMakeLists.txt
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@ -0,0 +1,29 @@
cmake_minimum_required(VERSION 3.18.1)
project(app CXX)
# ========== 关键修正1：源码路径（CMAKE_CURRENT_SOURCE_DIR 就是 src/main/cpp，无需重复加路径） ==========
add_library(
app
SHARED
# 直接写文件名即可，否则会找 src/main/cpp/src/main/cpp/xxx.cpp（路径重复）
${CMAKE_CURRENT_SOURCE_DIR}/src/main/cpp/opus_recorder.cpp
${CMAKE_CURRENT_SOURCE_DIR}/src/main/cpp/opus_decoder.cpp
)
# ========== 关键修正2：正确导入本地 jniLibs 下的 libopus.so ==========
# 1. 声明 opus 为「导入库」（SHARED 对应 .so 动态库）
add_library(opus SHARED IMPORTED)
# 2. 指定 libopus.so 的绝对路径（适配不同架构：arm64-v8a/armeabi-v7a 等）
set_target_properties(opus PROPERTIES
IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/src/main/jniLibs/${ANDROID_ABI}/libopus.so
# 可选：如果需要头文件，添加这行（头文件放 src/main/cpp/include/opus/ 下）
# INTERFACE_INCLUDE_DIRECTORIES ${CMAKE_CURRENT_SOURCE_DIR}/include
)
# ========== 关键修正3：链接导入的 opus 库（直接写库名，不用 opus:: 前缀） ==========
target_link_libraries(
app
PRIVATE
opus # 链接上面声明的本地 libopus.so
log # Android 日志库
)

5
app/build.gradle
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@ -19,7 +19,8 @@ android {
externalNativeBuild {
cmake {
cppFlags ""
arguments += "-DANDROID_STL=c++_shared"
cppFlags += "-std=c++17"
}
}
@ -50,6 +51,7 @@ android {
buildFeatures {
buildConfig = true // 显式启用
prefab = true
}
buildTypes {
@ -182,4 +184,5 @@ dependencies {
implementation files('libs/sherpa-onnx-1.12.23.aar')
implementation(libs.opus.v131)
}

63
app/src/main/assets/sherpa-onnx-kws-zipformer-wenetspeech-3.3M-2024-01-01/keywords.txt
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@ -1,3 +1,60 @@
x iǎo z ì t óng x ué @小智同学
x iǎo zh ì t óng x ué @小智同学
x iǎo zh ì t óng x ié @小智同学
x iǎo zh ì t óng x ué
x iǎo z ì t óng x ué
x iǎo j ì t óng x ué
x iǎo ch ì t óng x ué
x iáo zh ì t óng x ué
x iào zh ì t óng x ué
x iǎo zh í t óng x ué
x iǎo zh ǐ t óng x ué
x iǎo zh ì t ōng x ué
x iǎo zh ì t òng x ué
x iǎo zh ì t óng x uē
x iǎo zh ì t óng x uè
x iǎo zh ì t óng y ué
x iǎo zh i t óng x ué
x iǎo zh ì t óng x uè
x iǎo zh ì t óng x ué
x iǎo z ì t óng x ué
x iǎo j ì t óng x ué
x iǎo ch ì t óng x ué
x iáo zh ì t óng x ué
x iǎo zh í t óng x ué
x iǎo zh ì t ōng x ué
x iǎo zh ì t óng x uè
x iǎo zh ì d óng x ué
x iǎo z ì d óng x ué
x iǎo j ì d óng x ué
x iǎo ch ì d óng x ué
x iáo zh ì d óng x ué
x iào zh ì d óng x ué
x iǎo zh í d óng x ué
x iǎo zh ǐ d óng x ué
x iǎo zh ì d ōng x ué
x iǎo zh ì d òng x ué
x iǎo zh ì d óng x uē
x iǎo zh ì d óng x uè
x iǎo zh ì d óng y ué
x iǎo zh i d óng x ué
x iǎo zh ì d óng x uè
x iǎo zh ì d óng x ué
x iǎo z ì d óng x ué
x iǎo j ì d óng x ué
x iǎo ch ì d óng x ué
x iáo zh ì d óng x ué
x iǎo zh í d óng x ué
x iǎo zh ì d ōng x ué
x iǎo zh ì d óng x uè
s iǎo zh ì t óng x ué
sh iǎo zh ì t óng x ué
h iǎo zh ì t óng x ué
s iǎo zh ì t óng x ué
sh iǎo zh ì t óng x ué
s iǎo zh ì d óng x ué
sh iǎo zh ì d óng x ué
h iǎo zh ì d óng x ué
x iào zh ì t óng x ué
x iào zh ì d óng x ué

981
app/src/main/cpp/opus.h Normal file
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/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus.h
* @brief Opus reference implementation API
*/
#ifndef OPUS_H
#define OPUS_H
#include "opus_types.h"
#include "opus_defines.h"
#ifdef __cplusplus
extern "C" {
#endif
/**
* @mainpage Opus
*
* The Opus codec is designed for interactive speech and audio transmission over the Internet.
* It is designed by the IETF Codec Working Group and incorporates technology from
* Skype's SILK codec and Xiph.Org's CELT codec.
*
* The Opus codec is designed to handle a wide range of interactive audio applications,
* including Voice over IP, videoconferencing, in-game chat, and even remote live music
* performances. It can scale from low bit-rate narrowband speech to very high quality
* stereo music. Its main features are:
* @li Sampling rates from 8 to 48 kHz
* @li Bit-rates from 6 kb/s to 510 kb/s
* @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
* @li Audio bandwidth from narrowband to full-band
* @li Support for speech and music
* @li Support for mono and stereo
* @li Support for multichannel (up to 255 channels)
* @li Frame sizes from 2.5 ms to 60 ms
* @li Good loss robustness and packet loss concealment (PLC)
* @li Floating point and fixed-point implementation
*
* Documentation sections:
* @li @ref opus_encoder
* @li @ref opus_decoder
* @li @ref opus_repacketizer
* @li @ref opus_multistream
* @li @ref opus_libinfo
* @li @ref opus_custom
*/
/** @defgroup opus_encoder Opus Encoder
* @{
*
* @brief This page describes the process and functions used to encode Opus.
*
* Since Opus is a stateful codec, the encoding process starts with creating an encoder
* state. This can be done with:
*
* @code
* int error;
* OpusEncoder *enc;
* enc = opus_encoder_create(Fs, channels, application, &error);
* @endcode
*
* From this point, @c enc can be used for encoding an audio stream. An encoder state
* @b must @b not be used for more than one stream at the same time. Similarly, the encoder
* state @b must @b not be re-initialized for each frame.
*
* While opus_encoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
*
* @code
* int size;
* int error;
* OpusEncoder *enc;
* size = opus_encoder_get_size(channels);
* enc = malloc(size);
* error = opus_encoder_init(enc, Fs, channels, application);
* @endcode
*
* where opus_encoder_get_size() returns the required size for the encoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The encoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* It is possible to change some of the encoder's settings using the opus_encoder_ctl()
* interface. All these settings already default to the recommended value, so they should
* only be changed when necessary. The most common settings one may want to change are:
*
* @code
* opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
* opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
* opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
* @endcode
*
* where
*
* @arg bitrate is in bits per second (b/s)
* @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
* @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
*
* See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
*
* To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
* @code
* len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
* @endcode
*
* where
* <ul>
* <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
* <li>frame_size is the duration of the frame in samples (per channel)</li>
* <li>packet is the byte array to which the compressed data is written</li>
* <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
* Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
* </ul>
*
* opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
* The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
* is 2 bytes or less, then the packet does not need to be transmitted (DTX).
*
* Once the encoder state if no longer needed, it can be destroyed with
*
* @code
* opus_encoder_destroy(enc);
* @endcode
*
* If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
* then no action is required aside from potentially freeing the memory that was manually
* allocated for it (calling free(enc) for the example above)
*
*/
/** Opus encoder state.
* This contains the complete state of an Opus encoder.
* It is position independent and can be freely copied.
* @see opus_encoder_create,opus_encoder_init
*/
typedef struct OpusEncoder OpusEncoder;
/** Gets the size of an <code>OpusEncoder</code> structure.
* @param[in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
/**
*/
/** Allocates and initializes an encoder state.
* There are three coding modes:
*
* @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
* signals. It enhances the input signal by high-pass filtering and
* emphasizing formants and harmonics. Optionally it includes in-band
* forward error correction to protect against packet loss. Use this
* mode for typical VoIP applications. Because of the enhancement,
* even at high bitrates the output may sound different from the input.
*
* @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
* non-voice signals like music. Use this mode for music and mixed
* (music/voice) content, broadcast, and applications requiring less
* than 15 ms of coding delay.
*
* @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
* disables the speech-optimized mode in exchange for slightly reduced delay.
* This mode can only be set on an newly initialized or freshly reset encoder
* because it changes the codec delay.
*
* This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [out] error <tt>int*</tt>: @ref opus_errorcodes
* @note Regardless of the sampling rate and number channels selected, the Opus encoder
* can switch to a lower audio bandwidth or number of channels if the bitrate
* selected is too low. This also means that it is safe to always use 48 kHz stereo input
* and let the encoder optimize the encoding.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
opus_int32 Fs,
int channels,
int application,
int *error
);
/** Initializes a previously allocated encoder state
* The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_encoder_create(),opus_encoder_get_size()
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_encoder_init(
OpusEncoder *st,
opus_int32 Fs,
int channels,
int application
) OPUS_ARG_NONNULL(1);
/** Encodes an Opus frame.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
OpusEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes an Opus frame from floating point input.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
* Samples with a range beyond +/-1.0 are supported but will
* be clipped by decoders using the integer API and should
* only be used if it is known that the far end supports
* extended dynamic range.
* length is frame_size*channels*sizeof(float)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
OpusEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
* @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
/** Perform a CTL function on an Opus encoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusEncoder*</tt>: Encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_encoderctls.
* @see opus_genericctls
* @see opus_encoderctls
*/
OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/** @defgroup opus_decoder Opus Decoder
* @{
*
* @brief This page describes the process and functions used to decode Opus.
*
* The decoding process also starts with creating a decoder
* state. This can be done with:
* @code
* int error;
* OpusDecoder *dec;
* dec = opus_decoder_create(Fs, channels, &error);
* @endcode
* where
* @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
* @li channels is the number of channels (1 or 2)
* @li error will hold the error code in case of failure (or #OPUS_OK on success)
* @li the return value is a newly created decoder state to be used for decoding
*
* While opus_decoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
* @code
* int size;
* int error;
* OpusDecoder *dec;
* size = opus_decoder_get_size(channels);
* dec = malloc(size);
* error = opus_decoder_init(dec, Fs, channels);
* @endcode
* where opus_decoder_get_size() returns the required size for the decoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The decoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
* @code
* frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
* @endcode
* where
*
* @li packet is the byte array containing the compressed data
* @li len is the exact number of bytes contained in the packet
* @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
* @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
*
* opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
* If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
* buffer is too small to hold the decoded audio.
*
* Opus is a stateful codec with overlapping blocks and as a result Opus
* packets are not coded independently of each other. Packets must be
* passed into the decoder serially and in the correct order for a correct
* decode. Lost packets can be replaced with loss concealment by calling
* the decoder with a null pointer and zero length for the missing packet.
*
* A single codec state may only be accessed from a single thread at
* a time and any required locking must be performed by the caller. Separate
* streams must be decoded with separate decoder states and can be decoded
* in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
* defined.
*
*/
/** Opus decoder state.
* This contains the complete state of an Opus decoder.
* It is position independent and can be freely copied.
* @see opus_decoder_create,opus_decoder_init
*/
typedef struct OpusDecoder OpusDecoder;
/** Gets the size of an <code>OpusDecoder</code> structure.
* @param [in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
/** Allocates and initializes a decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
*
* Internally Opus stores data at 48000 Hz, so that should be the default
* value for Fs. However, the decoder can efficiently decode to buffers
* at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
* data at the full sample rate, or knows the compressed data doesn't
* use the full frequency range, it can request decoding at a reduced
* rate. Likewise, the decoder is capable of filling in either mono or
* interleaved stereo pcm buffers, at the caller's request.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
opus_int32 Fs,
int channels,
int *error
);
/** Initializes a previously allocated decoder state.
* The state must be at least the size returned by opus_decoder_get_size().
* This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_init(
OpusDecoder *st,
opus_int32 Fs,
int channels
) OPUS_ARG_NONNULL(1);
/** Decode an Opus packet.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available, the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an Opus packet with floating point output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusDecoder*</tt>: Decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_decoderctls.
* @see opus_genericctls
* @see opus_decoderctls
*/
OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
* @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
/** Parse an opus packet into one or more frames.
* Opus_decode will perform this operation internally so most applications do
* not need to use this function.
* This function does not copy the frames, the returned pointers are pointers into
* the input packet.
* @param [in] data <tt>char*</tt>: Opus packet to be parsed
* @param [in] len <tt>opus_int32</tt>: size of data
* @param [out] out_toc <tt>char*</tt>: TOC pointer
* @param [out] frames <tt>char*[48]</tt> encapsulated frames
* @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
* @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
* @returns number of frames
*/
OPUS_EXPORT int opus_packet_parse(
const unsigned char *data,
opus_int32 len,
unsigned char *out_toc,
const unsigned char *frames[48],
opus_int16 size[48],
int *payload_offset
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
/** Gets the bandwidth of an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
* @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
* @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
* @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
* @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of samples per frame from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet.
* This must contain at least one byte of
* data.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples per frame.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of channels from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @returns Number of channels
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of frames in an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of frames
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
* the signal is already in that range, nothing is done. If there are values
* outside of [-1,1], then the signal is clipped as smoothly as possible to
* both fit in the range and avoid creating excessive distortion in the
* process.
* @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
* @param [in] frame_size <tt>int</tt> Number of samples per channel to process
* @param [in] channels <tt>int</tt>: Number of channels
* @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
*/
OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
/**@}*/
/** @defgroup opus_repacketizer Repacketizer
* @{
*
* The repacketizer can be used to merge multiple Opus packets into a single
* packet or alternatively to split Opus packets that have previously been
* merged. Splitting valid Opus packets is always guaranteed to succeed,
* whereas merging valid packets only succeeds if all frames have the same
* mode, bandwidth, and frame size, and when the total duration of the merged
* packet is no more than 120 ms. The 120 ms limit comes from the
* specification and limits decoder memory requirements at a point where
* framing overhead becomes negligible.
*
* The repacketizer currently only operates on elementary Opus
* streams. It will not manipualte multistream packets successfully, except in
* the degenerate case where they consist of data from a single stream.
*
* The repacketizing process starts with creating a repacketizer state, either
* by calling opus_repacketizer_create() or by allocating the memory yourself,
* e.g.,
* @code
* OpusRepacketizer *rp;
* rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
* if (rp != NULL)
* opus_repacketizer_init(rp);
* @endcode
*
* Then the application should submit packets with opus_repacketizer_cat(),
* extract new packets with opus_repacketizer_out() or
* opus_repacketizer_out_range(), and then reset the state for the next set of
* input packets via opus_repacketizer_init().
*
* For example, to split a sequence of packets into individual frames:
* @code
* unsigned char *data;
* int len;
* while (get_next_packet(&data, &len))
* {
* unsigned char out[1276];
* opus_int32 out_len;
* int nb_frames;
* int err;
* int i;
* err = opus_repacketizer_cat(rp, data, len);
* if (err != OPUS_OK)
* {
* release_packet(data);
* return err;
* }
* nb_frames = opus_repacketizer_get_nb_frames(rp);
* for (i = 0; i < nb_frames; i++)
* {
* out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
* if (out_len < 0)
* {
* release_packet(data);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* }
* opus_repacketizer_init(rp);
* release_packet(data);
* }
* @endcode
*
* Alternatively, to combine a sequence of frames into packets that each
* contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
* @code
* // The maximum number of packets with duration TARGET_DURATION_MS occurs
* // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
* // packets.
* unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
* opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
* int nb_packets;
* unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
* opus_int32 out_len;
* int prev_toc;
* nb_packets = 0;
* while (get_next_packet(data+nb_packets, len+nb_packets))
* {
* int nb_frames;
* int err;
* nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
* if (nb_frames < 1)
* {
* release_packets(data, nb_packets+1);
* return nb_frames;
* }
* nb_frames += opus_repacketizer_get_nb_frames(rp);
* // If adding the next packet would exceed our target, or it has an
* // incompatible TOC sequence, output the packets we already have before
* // submitting it.
* // N.B., The nb_packets > 0 check ensures we've submitted at least one
* // packet since the last call to opus_repacketizer_init(). Otherwise a
* // single packet longer than TARGET_DURATION_MS would cause us to try to
* // output an (invalid) empty packet. It also ensures that prev_toc has
* // been set to a valid value. Additionally, len[nb_packets] > 0 is
* // guaranteed by the call to opus_packet_get_nb_frames() above, so the
* // reference to data[nb_packets][0] should be valid.
* if (nb_packets > 0 && (
* ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
* opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
* TARGET_DURATION_MS*48))
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* if (out_len < 0)
* {
* release_packets(data, nb_packets+1);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* opus_repacketizer_init(rp);
* release_packets(data, nb_packets);
* data[0] = data[nb_packets];
* len[0] = len[nb_packets];
* nb_packets = 0;
* }
* err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
* if (err != OPUS_OK)
* {
* release_packets(data, nb_packets+1);
* return err;
* }
* prev_toc = data[nb_packets][0];
* nb_packets++;
* }
* // Output the final, partial packet.
* if (nb_packets > 0)
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* release_packets(data, nb_packets);
* if (out_len < 0)
* return (int)out_len;
* output_next_packet(out, out_len);
* }
* @endcode
*
* An alternate way of merging packets is to simply call opus_repacketizer_cat()
* unconditionally until it fails. At that point, the merged packet can be
* obtained with opus_repacketizer_out() and the input packet for which
* opus_repacketizer_cat() needs to be re-added to a newly reinitialized
* repacketizer state.
*/
typedef struct OpusRepacketizer OpusRepacketizer;
/** Gets the size of an <code>OpusRepacketizer</code> structure.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
/** (Re)initializes a previously allocated repacketizer state.
* The state must be at least the size returned by opus_repacketizer_get_size().
* This can be used for applications which use their own allocator instead of
* malloc().
* It must also be called to reset the queue of packets waiting to be
* repacketized, which is necessary if the maximum packet duration of 120 ms
* is reached or if you wish to submit packets with a different Opus
* configuration (coding mode, audio bandwidth, frame size, or channel count).
* Failure to do so will prevent a new packet from being added with
* opus_repacketizer_cat().
* @see opus_repacketizer_create
* @see opus_repacketizer_get_size
* @see opus_repacketizer_cat
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
* (re)initialize.
* @returns A pointer to the same repacketizer state that was passed in.
*/
OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Allocates memory and initializes the new repacketizer with
* opus_repacketizer_init().
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
/** Frees an <code>OpusRepacketizer</code> allocated by
* opus_repacketizer_create().
* @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
/** Add a packet to the current repacketizer state.
* This packet must match the configuration of any packets already submitted
* for repacketization since the last call to opus_repacketizer_init().
* This means that it must have the same coding mode, audio bandwidth, frame
* size, and channel count.
* This can be checked in advance by examining the top 6 bits of the first
* byte of the packet, and ensuring they match the top 6 bits of the first
* byte of any previously submitted packet.
* The total duration of audio in the repacketizer state also must not exceed
* 120 ms, the maximum duration of a single packet, after adding this packet.
*
* The contents of the current repacketizer state can be extracted into new
* packets using opus_repacketizer_out() or opus_repacketizer_out_range().
*
* In order to add a packet with a different configuration or to add more
* audio beyond 120 ms, you must clear the repacketizer state by calling
* opus_repacketizer_init().
* If a packet is too large to add to the current repacketizer state, no part
* of it is added, even if it contains multiple frames, some of which might
* fit.
* If you wish to be able to add parts of such packets, you should first use
* another repacketizer to split the packet into pieces and add them
* individually.
* @see opus_repacketizer_out_range
* @see opus_repacketizer_out
* @see opus_repacketizer_init
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
* add the packet.
* @param[in] data <tt>const unsigned char*</tt>: The packet data.
* The application must ensure
* this pointer remains valid
* until the next call to
* opus_repacketizer_init() or
* opus_repacketizer_destroy().
* @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
* @returns An error code indicating whether or not the operation succeeded.
* @retval #OPUS_OK The packet's contents have been added to the repacketizer
* state.
* @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
* the packet's TOC sequence was not compatible
* with previously submitted packets (because
* the coding mode, audio bandwidth, frame size,
* or channel count did not match), or adding
* this packet would increase the total amount of
* audio stored in the repacketizer state to more
* than 120 ms.
*/
OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param begin <tt>int</tt>: The index of the first frame in the current
* repacketizer state to include in the output.
* @param end <tt>int</tt>: One past the index of the last frame in the
* current repacketizer state to include in the
* output.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1276</code> for a single frame,
* or for multiple frames,
* <code>1277*(end-begin)</code>.
* However, <code>1*(end-begin)</code> plus
* the size of all packet data submitted to
* the repacketizer since the last call to
* opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
* frames (begin < 0, begin >= end, or end >
* opus_repacketizer_get_nb_frames()).
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Return the total number of frames contained in packet data submitted to
* the repacketizer state so far via opus_repacketizer_cat() since the last
* call to opus_repacketizer_init() or opus_repacketizer_create().
* This defines the valid range of packets that can be extracted with
* opus_repacketizer_out_range() or opus_repacketizer_out().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
* frames.
* @returns The total number of frames contained in the packet data submitted
* to the repacketizer state.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* This is a convenience routine that returns all the data submitted so far
* in a single packet.
* It is equivalent to calling
* @code
* opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
* data, maxlen)
* @endcode
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
* However,
* <code>1*opus_repacketizer_get_nb_frames(rp)</code>
* plus the size of all packet data
* submitted to the repacketizer since the
* last call to opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_H */

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/* Copyright (c) 2007-2008 CSIRO
Copyright (c) 2007-2009 Xiph.Org Foundation
Copyright (c) 2008-2012 Gregory Maxwell
Written by Jean-Marc Valin and Gregory Maxwell */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
@file opus_custom.h
@brief Opus-Custom reference implementation API
*/
#ifndef OPUS_CUSTOM_H
#define OPUS_CUSTOM_H
#include "opus_defines.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifdef CUSTOM_MODES
# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
#else
# define OPUS_CUSTOM_EXPORT
# ifdef OPUS_BUILD
# define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
# else
# define OPUS_CUSTOM_EXPORT_STATIC
# endif
#endif
/** @defgroup opus_custom Opus Custom
* @{
* Opus Custom is an optional part of the Opus specification and
* reference implementation which uses a distinct API from the regular
* API and supports frame sizes that are not normally supported.\ Use
* of Opus Custom is discouraged for all but very special applications
* for which a frame size different from 2.5, 5, 10, or 20 ms is needed
* (for either complexity or latency reasons) and where interoperability
* is less important.
*
* In addition to the interoperability limitations the use of Opus custom
* disables a substantial chunk of the codec and generally lowers the
* quality available at a given bitrate. Normally when an application needs
* a different frame size from the codec it should buffer to match the
* sizes but this adds a small amount of delay which may be important
* in some very low latency applications. Some transports (especially
* constant rate RF transports) may also work best with frames of
* particular durations.
*
* Libopus only supports custom modes if they are enabled at compile time.
*
* The Opus Custom API is similar to the regular API but the
* @ref opus_encoder_create and @ref opus_decoder_create calls take
* an additional mode parameter which is a structure produced by
* a call to @ref opus_custom_mode_create. Both the encoder and decoder
* must create a mode using the same sample rate (fs) and frame size
* (frame size) so these parameters must either be signaled out of band
* or fixed in a particular implementation.
*
* Similar to regular Opus the custom modes support on the fly frame size
* switching, but the sizes available depend on the particular frame size in
* use. For some initial frame sizes on a single on the fly size is available.
*/
/** Contains the state of an encoder. One encoder state is needed
for each stream. It is initialized once at the beginning of the
stream. Do *not* re-initialize the state for every frame.
@brief Encoder state
*/
typedef struct OpusCustomEncoder OpusCustomEncoder;
/** State of the decoder. One decoder state is needed for each stream.
It is initialized once at the beginning of the stream. Do *not*
re-initialize the state for every frame.
@brief Decoder state
*/
typedef struct OpusCustomDecoder OpusCustomDecoder;
/** The mode contains all the information necessary to create an
encoder. Both the encoder and decoder need to be initialized
with exactly the same mode, otherwise the output will be
corrupted.
@brief Mode configuration
*/
typedef struct OpusCustomMode OpusCustomMode;
/** Creates a new mode struct. This will be passed to an encoder or
* decoder. The mode MUST NOT BE DESTROYED until the encoders and
* decoders that use it are destroyed as well.
* @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
* @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
* packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
* @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
* @return A newly created mode
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
/** Destroys a mode struct. Only call this after all encoders and
* decoders using this mode are destroyed as well.
* @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
/* Encoder */
/** Gets the size of an OpusCustomEncoder structure.
* @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
* @param [in] channels <tt>int</tt>: Number of channels
* @returns size
*/
OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1);
# ifdef CUSTOM_MODES
/** Initializes a previously allocated encoder state
* The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* decoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @return OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
OpusCustomEncoder *st,
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
# endif
#endif
/** Creates a new encoder state. Each stream needs its own encoder
* state (can't be shared across simultaneous streams).
* @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* decoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @param [out] error <tt>int*</tt>: Returns an error code
* @return Newly created encoder state.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
const OpusCustomMode *mode,
int channels,
int *error
) OPUS_ARG_NONNULL(1);
/** Destroys a an encoder state.
* @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
/** Encodes a frame of audio.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
* Samples with a range beyond +/-1.0 are supported but will
* be clipped by decoders using the integer API and should
* only be used if it is known that the far end supports
* extended dynamic range. There must be exactly
* frame_size samples per channel.
* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
* @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
* @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
* (can change from one frame to another)
* @return Number of bytes written to "compressed".
* If negative, an error has occurred (see error codes). It is IMPORTANT that
* the length returned be somehow transmitted to the decoder. Otherwise, no
* decoding is possible.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
OpusCustomEncoder *st,
const float *pcm,
int frame_size,
unsigned char *compressed,
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a frame of audio.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
* There must be exactly frame_size samples per channel.
* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
* @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
* @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
* (can change from one frame to another)
* @return Number of bytes written to "compressed".
* If negative, an error has occurred (see error codes). It is IMPORTANT that
* the length returned be somehow transmitted to the decoder. Otherwise, no
* decoding is possible.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
OpusCustomEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *compressed,
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom encoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @see opus_encoderctls
*/
OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
/* Decoder */
/** Gets the size of an OpusCustomDecoder structure.
* @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
* @param [in] channels <tt>int</tt>: Number of channels
* @returns size
*/
OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1);
/** Initializes a previously allocated decoder state
* The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* encoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @return OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
OpusCustomDecoder *st,
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
#endif
/** Creates a new decoder state. Each stream needs its own decoder state (can't
* be shared across simultaneous streams).
* @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
* stream (must be the same characteristics as used for the encoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @param [out] error <tt>int*</tt>: Returns an error code
* @return Newly created decoder state.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
const OpusCustomMode *mode,
int channels,
int *error
) OPUS_ARG_NONNULL(1);
/** Destroys a an decoder state.
* @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
/** Decode an opus custom frame with floating point output
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>int</tt>: Number of bytes in payload
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel of available space in *pcm.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
OpusCustomDecoder *st,
const unsigned char *data,
int len,
float *pcm,
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an opus custom frame
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>int</tt>: Number of bytes in payload
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel of available space in *pcm.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
OpusCustomDecoder *st,
const unsigned char *data,
int len,
opus_int16 *pcm,
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @see opus_genericctls
*/
OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_CUSTOM_H */

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#include <jni.h>
#include <string>
#include <android/log.h>
#include "opus.h"
#define LOG_TAG "OpusJNI"
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
static OpusDecoder* decoderHandle = nullptr;
extern "C" {
JNIEXPORT jlong JNICALL
Java_info_dourok_voicebot_OpusDecoder_nativeInitDecoder(JNIEnv *env, jobject thiz,
jint sample_rate, jint channels) {
int error;
OpusDecoder *decoder = opus_decoder_create(sample_rate, channels, &error);
if (error != OPUS_OK || decoder == nullptr) {
LOGE("Failed to create decoder: %s", opus_strerror(error));
return 0;
}
LOGI("Opus decoder initialized: sample_rate=%d, channels=%d", sample_rate, channels);
return (jlong)(intptr_t)decoder;
}
JNIEXPORT jint JNICALL
Java_info_dourok_voicebot_OpusDecoder_nativeDecodeBytes(JNIEnv *env, jobject thiz,
jlong decoder_handle,
jbyteArray input_buffer,
jint input_size,
jbyteArray output_buffer,
jint max_output_size) {
OpusDecoder *decoder = (OpusDecoder*)(intptr_t)decoder_handle;
if (decoder == nullptr) {
LOGE("Decoder handle is null");
return -1;
}
jbyte *input = env->GetByteArrayElements(input_buffer, nullptr);
jbyte *output = env->GetByteArrayElements(output_buffer, nullptr);
int frame_size = max_output_size / 2; // 16-bit PCM
int result = opus_decode(decoder, (unsigned char*)input, input_size,
(opus_int16*)output, frame_size, 0);
env->ReleaseByteArrayElements(input_buffer, input, JNI_ABORT);
env->ReleaseByteArrayElements(output_buffer, output, 0);
if (result < 0) {
LOGE("Decoding failed: %s", opus_strerror(result));
return -1;
}
return result * 2; // 返回字节数（每个样本2字节）
}
JNIEXPORT void JNICALL
Java_info_dourok_voicebot_OpusDecoder_nativeReleaseDecoder(JNIEnv *env, jobject thiz,
jlong decoder_handle) {
OpusDecoder *decoder = (OpusDecoder*)(intptr_t)decoder_handle;
if (decoder != nullptr) {
opus_decoder_destroy(decoder);
LOGI("Opus decoder released");
}
}
} // extern "C"

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/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_defines.h
* @brief Opus reference implementation constants
*/
#ifndef OPUS_DEFINES_H
#define OPUS_DEFINES_H
#include "opus_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @defgroup opus_errorcodes Error codes
* @{
*/
/** No error @hideinitializer*/
#define OPUS_OK 0
/** One or more invalid/out of range arguments @hideinitializer*/
#define OPUS_BAD_ARG -1
/** Not enough bytes allocated in the buffer @hideinitializer*/
#define OPUS_BUFFER_TOO_SMALL -2
/** An internal error was detected @hideinitializer*/
#define OPUS_INTERNAL_ERROR -3
/** The compressed data passed is corrupted @hideinitializer*/
#define OPUS_INVALID_PACKET -4
/** Invalid/unsupported request number @hideinitializer*/
#define OPUS_UNIMPLEMENTED -5
/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
#define OPUS_INVALID_STATE -6
/** Memory allocation has failed @hideinitializer*/
#define OPUS_ALLOC_FAIL -7
/**@}*/
/** @cond OPUS_INTERNAL_DOC */
/**Export control for opus functions */
#ifndef OPUS_EXPORT
# if defined(WIN32)
# if defined(OPUS_BUILD) && defined(DLL_EXPORT)
# define OPUS_EXPORT __declspec(dllexport)
# else
# define OPUS_EXPORT
# endif
# elif defined(__GNUC__) && defined(OPUS_BUILD)
# define OPUS_EXPORT __attribute__ ((visibility ("default")))
# else
# define OPUS_EXPORT
# endif
#endif
# if !defined(OPUS_GNUC_PREREQ)
# if defined(__GNUC__)&&defined(__GNUC_MINOR__)
# define OPUS_GNUC_PREREQ(_maj,_min) \
((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
# else
# define OPUS_GNUC_PREREQ(_maj,_min) 0
# endif
# endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(3,0)
# define OPUS_RESTRICT __restrict__
# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
# define OPUS_RESTRICT __restrict
# else
# define OPUS_RESTRICT
# endif
#else
# define OPUS_RESTRICT restrict
#endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(2,7)
# define OPUS_INLINE __inline__
# elif (defined(_MSC_VER))
# define OPUS_INLINE __inline
# else
# define OPUS_INLINE
# endif
#else
# define OPUS_INLINE inline
#endif
/**Warning attributes for opus functions
* NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
* some paranoid null checks. */
#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
#else
# define OPUS_WARN_UNUSED_RESULT
#endif
#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x)))
#else
# define OPUS_ARG_NONNULL(_x)
#endif
/** These are the actual Encoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
#define OPUS_SET_APPLICATION_REQUEST 4000
#define OPUS_GET_APPLICATION_REQUEST 4001
#define OPUS_SET_BITRATE_REQUEST 4002
#define OPUS_GET_BITRATE_REQUEST 4003
#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004
#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005
#define OPUS_SET_VBR_REQUEST 4006
#define OPUS_GET_VBR_REQUEST 4007
#define OPUS_SET_BANDWIDTH_REQUEST 4008
#define OPUS_GET_BANDWIDTH_REQUEST 4009
#define OPUS_SET_COMPLEXITY_REQUEST 4010
#define OPUS_GET_COMPLEXITY_REQUEST 4011
#define OPUS_SET_INBAND_FEC_REQUEST 4012
#define OPUS_GET_INBAND_FEC_REQUEST 4013
#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014
#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015
#define OPUS_SET_DTX_REQUEST 4016
#define OPUS_GET_DTX_REQUEST 4017
#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020
#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021
#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022
#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023
#define OPUS_SET_SIGNAL_REQUEST 4024
#define OPUS_GET_SIGNAL_REQUEST 4025
#define OPUS_GET_LOOKAHEAD_REQUEST 4027
/* #define OPUS_RESET_STATE 4028 */
#define OPUS_GET_SAMPLE_RATE_REQUEST 4029
#define OPUS_GET_FINAL_RANGE_REQUEST 4031
#define OPUS_GET_PITCH_REQUEST 4033
#define OPUS_SET_GAIN_REQUEST 4034
#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */
#define OPUS_SET_LSB_DEPTH_REQUEST 4036
#define OPUS_GET_LSB_DEPTH_REQUEST 4037
#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
#define OPUS_GET_IN_DTX_REQUEST 4049
/** Defines for the presence of extended APIs. */
#define OPUS_HAVE_OPUS_PROJECTION_H
/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
/** @endcond */
/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
* @see opus_genericctls, opus_encoderctls
* @{
*/
/* Values for the various encoder CTLs */
#define OPUS_AUTO -1000 /**<Auto/default setting @hideinitializer*/
#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/
/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
* @hideinitializer */
#define OPUS_APPLICATION_VOIP 2048
/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
* @hideinitializer */
#define OPUS_APPLICATION_AUDIO 2049
/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
* @hideinitializer */
#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
#define OPUS_SIGNAL_VOICE 3001 /**< Signal being encoded is voice */
#define OPUS_SIGNAL_MUSIC 3002 /**< Signal being encoded is music */
#define OPUS_BANDWIDTH_NARROWBAND 1101 /**< 4 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_MEDIUMBAND 1102 /**< 6 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_WIDEBAND 1103 /**< 8 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_SUPERWIDEBAND 1104 /**<12 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_FULLBAND 1105 /**<20 kHz bandpass @hideinitializer*/
#define OPUS_FRAMESIZE_ARG 5000 /**< Select frame size from the argument (default) */
#define OPUS_FRAMESIZE_2_5_MS 5001 /**< Use 2.5 ms frames */
#define OPUS_FRAMESIZE_5_MS 5002 /**< Use 5 ms frames */
#define OPUS_FRAMESIZE_10_MS 5003 /**< Use 10 ms frames */
#define OPUS_FRAMESIZE_20_MS 5004 /**< Use 20 ms frames */
#define OPUS_FRAMESIZE_40_MS 5005 /**< Use 40 ms frames */
#define OPUS_FRAMESIZE_60_MS 5006 /**< Use 60 ms frames */
#define OPUS_FRAMESIZE_80_MS 5007 /**< Use 80 ms frames */
#define OPUS_FRAMESIZE_100_MS 5008 /**< Use 100 ms frames */
#define OPUS_FRAMESIZE_120_MS 5009 /**< Use 120 ms frames */
/**@}*/
/** @defgroup opus_encoderctls Encoder related CTLs
*
* These are convenience macros for use with the \c opus_encode_ctl
* interface. They are used to generate the appropriate series of
* arguments for that call, passing the correct type, size and so
* on as expected for each particular request.
*
* Some usage examples:
*
* @code
* int ret;
* ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
* if (ret != OPUS_OK) return ret;
*
* opus_int32 rate;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* @endcode
*
* @see opus_genericctls, opus_encoder
* @{
*/
/** Configures the encoder's computational complexity.
* The supported range is 0-10 inclusive with 10 representing the highest complexity.
* @see OPUS_GET_COMPLEXITY
* @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
*
* @hideinitializer */
#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
/** Gets the encoder's complexity configuration.
* @see OPUS_SET_COMPLEXITY
* @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
* inclusive.
* @hideinitializer */
#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
/** Configures the bitrate in the encoder.
* Rates from 500 to 512000 bits per second are meaningful, as well as the
* special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
* The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
* rate as it can, which is useful for controlling the rate by adjusting the
* output buffer size.
* @see OPUS_GET_BITRATE
* @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
* is determined based on the number of
* channels and the input sampling rate.
* @hideinitializer */
#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
/** Gets the encoder's bitrate configuration.
* @see OPUS_SET_BITRATE
* @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
* The default is determined based on the
* number of channels and the input
* sampling rate.
* @hideinitializer */
#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables variable bitrate (VBR) in the encoder.
* The configured bitrate may not be met exactly because frames must
* be an integer number of bytes in length.
* @see OPUS_GET_VBR
* @see OPUS_SET_VBR_CONSTRAINT
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
* cause noticeable quality degradation.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
* #OPUS_SET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
/** Determine if variable bitrate (VBR) is enabled in the encoder.
* @see OPUS_SET_VBR
* @see OPUS_GET_VBR_CONSTRAINT
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Hard CBR.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
* #OPUS_GET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables constrained VBR in the encoder.
* This setting is ignored when the encoder is in CBR mode.
* @warning Only the MDCT mode of Opus currently heeds the constraint.
* Speech mode ignores it completely, hybrid mode may fail to obey it
* if the LPC layer uses more bitrate than the constraint would have
* permitted.
* @see OPUS_GET_VBR_CONSTRAINT
* @see OPUS_SET_VBR
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
* frame of buffering delay assuming a transport with a
* serialization speed of the nominal bitrate.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
/** Determine if constrained VBR is enabled in the encoder.
* @see OPUS_SET_VBR_CONSTRAINT
* @see OPUS_GET_VBR
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default).</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
/** Configures mono/stereo forcing in the encoder.
* This can force the encoder to produce packets encoded as either mono or
* stereo, regardless of the format of the input audio. This is useful when
* the caller knows that the input signal is currently a mono source embedded
* in a stereo stream.
* @see OPUS_GET_FORCE_CHANNELS
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
/** Gets the encoder's forced channel configuration.
* @see OPUS_SET_FORCE_CHANNELS
* @param[out] x <tt>opus_int32 *</tt>:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
/** Configures the maximum bandpass that the encoder will select automatically.
* Applications should normally use this instead of #OPUS_SET_BANDWIDTH
* (leaving that set to the default, #OPUS_AUTO). This allows the
* application to set an upper bound based on the type of input it is
* providing, but still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_MAX_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured maximum allowed bandpass.
* @see OPUS_SET_MAX_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Allowed values:
* <dl>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Sets the encoder's bandpass to a specific value.
* This prevents the encoder from automatically selecting the bandpass based
* on the available bitrate. If an application knows the bandpass of the input
* audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
* instead, which still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Configures the type of signal being encoded.
* This is a hint which helps the encoder's mode selection.
* @see OPUS_GET_SIGNAL
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal type.
* @see OPUS_SET_SIGNAL
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's intended application.
* The initial value is a mandatory argument to the encoder_create function.
* @see OPUS_GET_APPLICATION
* @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured application.
* @see OPUS_SET_APPLICATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the total samples of delay added by the entire codec.
* This can be queried by the encoder and then the provided number of samples can be
* skipped on from the start of the decoder's output to provide time aligned input
* and output. From the perspective of a decoding application the real data begins this many
* samples late.
*
* The decoder contribution to this delay is identical for all decoders, but the
* encoder portion of the delay may vary from implementation to implementation,
* version to version, or even depend on the encoder's initial configuration.
* Applications needing delay compensation should call this CTL rather than
* hard-coding a value.
* @param[out] x <tt>opus_int32 *</tt>: Number of lookahead samples
* @hideinitializer */
#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of inband forward error correction (FEC).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_INBAND_FEC
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable inband FEC (default).</dd>
* <dt>1</dt><dd>Enable inband FEC.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of inband forward error correction.
* @see OPUS_SET_INBAND_FEC
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Inband FEC disabled (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's expected packet loss percentage.
* Higher values trigger progressively more loss resistant behavior in the encoder
* at the expense of quality at a given bitrate in the absence of packet loss, but
* greater quality under loss.
* @see OPUS_GET_PACKET_LOSS_PERC
* @param[in] x <tt>opus_int32</tt>: Loss percentage in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured packet loss percentage.
* @see OPUS_SET_PACKET_LOSS_PERC
* @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
* in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of discontinuous transmission (DTX).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_DTX
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable DTX (default).</dd>
* <dt>1</dt><dd>Enabled DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of discontinuous transmission.
* @see OPUS_SET_DTX
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>DTX disabled (default).</dd>
* <dt>1</dt><dd>DTX enabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
/** Configures the depth of signal being encoded.
*
* This is a hint which helps the encoder identify silence and near-silence.
* It represents the number of significant bits of linear intensity below
* which the signal contains ignorable quantization or other noise.
*
* For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
* for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
* for 16-bit linear pcm input with opus_encode_float().
*
* When using opus_encode() instead of opus_encode_float(), or when libopus
* is compiled for fixed-point, the encoder uses the minimum of the value
* set here and the value 16.
*
* @see OPUS_GET_LSB_DEPTH
* @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
* (default: 24).
* @hideinitializer */
#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal depth.
* @see OPUS_SET_LSB_DEPTH
* @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
* 24 (default: 24).
* @hideinitializer */
#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of variable duration frames.
* When variable duration is enabled, the encoder is free to use a shorter frame
* size than the one requested in the opus_encode*() call.
* It is then the user's responsibility
* to verify how much audio was encoded by checking the ToC byte of the encoded
* packet. The part of the audio that was not encoded needs to be resent to the
* encoder for the next call. Do not use this option unless you <b>really</b>
* know what you are doing.
* @see OPUS_GET_EXPERT_FRAME_DURATION
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured use of variable duration frames.
* @see OPUS_SET_EXPERT_FRAME_DURATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables almost all use of prediction, making frames almost
* completely independent. This reduces quality.
* @see OPUS_GET_PREDICTION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable prediction (default).</dd>
* <dt>1</dt><dd>Disable prediction.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured prediction status.
* @see OPUS_SET_PREDICTION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Prediction enabled (default).</dd>
* <dt>1</dt><dd>Prediction disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_genericctls Generic CTLs
*
* These macros are used with the \c opus_decoder_ctl and
* \c opus_encoder_ctl calls to generate a particular
* request.
*
* When called on an \c OpusDecoder they apply to that
* particular decoder instance. When called on an
* \c OpusEncoder they apply to the corresponding setting
* on that encoder instance, if present.
*
* Some usage examples:
*
* @code
* int ret;
* opus_int32 pitch;
* ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
* if (ret == OPUS_OK) return ret;
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
*
* opus_int32 enc_bw, dec_bw;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
* opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
* if (enc_bw != dec_bw) {
* printf("packet bandwidth mismatch!\n");
* }
* @endcode
*
* @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
* @{
*/
/** Resets the codec state to be equivalent to a freshly initialized state.
* This should be called when switching streams in order to prevent
* the back to back decoding from giving different results from
* one at a time decoding.
* @hideinitializer */
#define OPUS_RESET_STATE 4028
/** Gets the final state of the codec's entropy coder.
* This is used for testing purposes,
* The encoder and decoder state should be identical after coding a payload
* (assuming no data corruption or software bugs)
*
* @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
*
* @hideinitializer */
#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
/** Gets the encoder's configured bandpass or the decoder's last bandpass.
* @see OPUS_SET_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Gets the sampling rate the encoder or decoder was initialized with.
* This simply returns the <code>Fs</code> value passed to opus_encoder_init()
* or opus_decoder_init().
* @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
* @hideinitializer
*/
#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables the use of phase inversion for intensity stereo,
* improving the quality of mono downmixes, but slightly reducing normal
* stereo quality. Disabling phase inversion in the decoder does not comply
* with RFC 6716, although it does not cause any interoperability issue and
* is expected to become part of the Opus standard once RFC 6716 is updated
* by draft-ietf-codec-opus-update.
* @see OPUS_GET_PHASE_INVERSION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable phase inversion (default).</dd>
* <dt>1</dt><dd>Disable phase inversion.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured phase inversion status.
* @see OPUS_SET_PHASE_INVERSION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
* <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/** Gets the DTX state of the encoder.
* Returns whether the last encoded frame was either a comfort noise update
* during DTX or not encoded because of DTX.
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>The encoder is not in DTX.</dd>
* <dt>1</dt><dd>The encoder is in DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_decoderctls Decoder related CTLs
* @see opus_genericctls, opus_encoderctls, opus_decoder
* @{
*/
/** Configures decoder gain adjustment.
* Scales the decoded output by a factor specified in Q8 dB units.
* This has a maximum range of -32768 to 32767 inclusive, and returns
* OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
* This setting survives decoder reset.
*
* gain = pow(10, x/(20.0*256))
*
* @param[in] x <tt>opus_int32</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
*
* @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
* @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
* @hideinitializer */
#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the pitch of the last decoded frame, if available.
* This can be used for any post-processing algorithm requiring the use of pitch,
* e.g. time stretching/shortening. If the last frame was not voiced, or if the
* pitch was not coded in the frame, then zero is returned.
*
* This CTL is only implemented for decoder instances.
*
* @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
*
* @hideinitializer */
#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_libinfo Opus library information functions
* @{
*/
/** Converts an opus error code into a human readable string.
*
* @param[in] error <tt>int</tt>: Error number
* @returns Error string
*/
OPUS_EXPORT const char *opus_strerror(int error);
/** Gets the libopus version string.
*
* Applications may look for the substring "-fixed" in the version string to
* determine whether they have a fixed-point or floating-point build at
* runtime.
*
* @returns Version string
*/
OPUS_EXPORT const char *opus_get_version_string(void);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_DEFINES_H */

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/* Copyright (c) 2011 Xiph.Org Foundation
Written by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_multistream.h
* @brief Opus reference implementation multistream API
*/
#ifndef OPUS_MULTISTREAM_H
#define OPUS_MULTISTREAM_H
#include "opus.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** Macros to trigger compilation errors when the wrong types are provided to a
* CTL. */
/**@{*/
#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
/**@}*/
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
/**@}*/
/** @endcond */
/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
* @ref opus_decoderctls may be applied to a multistream encoder or decoder as
* well.
* In addition, you may retrieve the encoder or decoder state for an specific
* stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
* #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
*/
/**@{*/
/** Gets the encoder state for an individual stream of a multistream encoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the encoder.
* @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
* encoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
/** Gets the decoder state for an individual stream of a multistream decoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the decoder.
* @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
* decoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
/**@}*/
/** @defgroup opus_multistream Opus Multistream API
* @{
*
* The multistream API allows individual Opus streams to be combined into a
* single packet, enabling support for up to 255 channels. Unlike an
* elementary Opus stream, the encoder and decoder must negotiate the channel
* configuration before the decoder can successfully interpret the data in the
* packets produced by the encoder. Some basic information, such as packet
* duration, can be computed without any special negotiation.
*
* The format for multistream Opus packets is defined in
* <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
* and is based on the self-delimited Opus framing described in Appendix B of
* <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
* Normal Opus packets are just a degenerate case of multistream Opus packets,
* and can be encoded or decoded with the multistream API by setting
* <code>streams</code> to <code>1</code> when initializing the encoder or
* decoder.
*
* Multistream Opus streams can contain up to 255 elementary Opus streams.
* These may be either "uncoupled" or "coupled", indicating that the decoder
* is configured to decode them to either 1 or 2 channels, respectively.
* The streams are ordered so that all coupled streams appear at the
* beginning.
*
* A <code>mapping</code> table defines which decoded channel <code>i</code>
* should be used for each input/output (I/O) channel <code>j</code>. This table is
* typically provided as an unsigned char array.
* Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
* If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
* encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
* is even, or as the right channel of stream <code>(i/2)</code> if
* <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
* mono in stream <code>(i - coupled_streams)</code>, unless it has the special
* value 255, in which case it is omitted from the encoding entirely (the
* decoder will reproduce it as silence). Each value <code>i</code> must either
* be the special value 255 or be less than <code>streams + coupled_streams</code>.
*
* The output channels specified by the encoder
* should use the
* <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
* channel ordering</a>. A decoder may wish to apply an additional permutation
* to the mapping the encoder used to achieve a different output channel
* order (e.g. for outputing in WAV order).
*
* Each multistream packet contains an Opus packet for each stream, and all of
* the Opus packets in a single multistream packet must have the same
* duration. Therefore the duration of a multistream packet can be extracted
* from the TOC sequence of the first stream, which is located at the
* beginning of the packet, just like an elementary Opus stream:
*
* @code
* int nb_samples;
* int nb_frames;
* nb_frames = opus_packet_get_nb_frames(data, len);
* if (nb_frames < 1)
* return nb_frames;
* nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
* @endcode
*
* The general encoding and decoding process proceeds exactly the same as in
* the normal @ref opus_encoder and @ref opus_decoder APIs.
* See their documentation for an overview of how to use the corresponding
* multistream functions.
*/
/** Opus multistream encoder state.
* This contains the complete state of a multistream Opus encoder.
* It is position independent and can be freely copied.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_init
*/
typedef struct OpusMSEncoder OpusMSEncoder;
/** Opus multistream decoder state.
* This contains the complete state of a multistream Opus decoder.
* It is position independent and can be freely copied.
* @see opus_multistream_decoder_create
* @see opus_multistream_decoder_init
*/
typedef struct OpusMSDecoder OpusMSDecoder;
/**\name Multistream encoder functions */
/**@{*/
/** Gets the size of an OpusMSEncoder structure.
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
int streams,
int coupled_streams
);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a multistream encoder state.
* Call opus_multistream_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(5);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Initialize a previously allocated multistream encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
OPUS_EXPORT int opus_multistream_surround_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
/** Encodes a multistream Opus frame.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
OpusMSEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a multistream Opus frame from floating point input.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
OpusMSEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusMSEncoder</code> allocated by
* opus_multistream_encoder_create().
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
/** Perform a CTL function on a multistream Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Multistream decoder functions */
/**@{*/
/** Gets the size of an <code>OpusMSDecoder</code> structure.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
int streams,
int coupled_streams
);
/** Allocates and initializes a multistream decoder state.
* Call opus_multistream_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_decoder_create
* @see opus_multistream_deocder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_decoder_init(
OpusMSDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a multistream Opus packet.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a multistream Opus packet with floating point output.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a multistream Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusMSDecoder</code> allocated by
* opus_multistream_decoder_create().
* @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_MULTISTREAM_H */

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/* Copyright (c) 2017 Google Inc.
Written by Andrew Allen */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_projection.h
* @brief Opus projection reference API
*/
#ifndef OPUS_PROJECTION_H
#define OPUS_PROJECTION_H
#include "opus_multistream.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.c
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005
/**@}*/
/** @endcond */
/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
* @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
* projection encoder or decoder as well.
*/
/**@{*/
/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
* of the demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the size in bytes of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
* demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
/** Copies the demixing matrix to the supplied pointer location.
* @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
* supplied pointer location.
* @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
* pointer location.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
/**@}*/
/** Opus projection encoder state.
* This contains the complete state of a projection Opus encoder.
* It is position independent and can be freely copied.
* @see opus_projection_ambisonics_encoder_create
*/
typedef struct OpusProjectionEncoder OpusProjectionEncoder;
/** Opus projection decoder state.
* This contains the complete state of a projection Opus decoder.
* It is position independent and can be freely copied.
* @see opus_projection_decoder_create
* @see opus_projection_decoder_init
*/
typedef struct OpusProjectionDecoder OpusProjectionDecoder;
/**\name Projection encoder functions */
/**@{*/
/** Gets the size of an OpusProjectionEncoder structure.
* @param channels <tt>int</tt>: The total number of input channels to encode.
* This must be no more than 255.
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a projection encoder state.
* Call opus_projection_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @param[out] streams <tt>int *</tt>: The total number of streams that will
* be encoded from the input.
* @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
* streams that will be encoded from the input.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
/** Initialize a previously allocated projection encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_ambisonics_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_ambisonics_encoder_create
* @see opus_projection_ambisonics_encoder_get_size
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
OpusProjectionEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Encodes a projection Opus frame.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
OpusProjectionEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a projection Opus frame from floating point input.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
OpusProjectionEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusProjectionEncoder</code> allocated by
* opus_projection_ambisonics_encoder_create().
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
*/
OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
/** Perform a CTL function on a projection Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Projection decoder functions */
/**@{*/
/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
* @param channels <tt>int</tt>: The total number of output channels.
* This must be no more than 255.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
int channels,
int streams,
int coupled_streams
);
/** Allocates and initializes a projection decoder state.
* Call opus_projection_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated projection decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_decoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_decoder_create
* @see opus_projection_deocder_get_size
* @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_decoder_init(
OpusProjectionDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a projection Opus packet.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a projection Opus packet with floating point output.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a projection Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusProjectionDecoder</code> allocated by
* opus_projection_decoder_create().
* @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
*/
OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_PROJECTION_H */

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#include <jni.h>
#include <string>
#include <android/log.h>
#include "opus.h"
#define LOG_TAG "OpusJNI"
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
// Opus编码器句柄
static OpusEncoder *encoderHandle = nullptr;
extern "C" {
JNIEXPORT jlong JNICALL
Java_info_dourok_voicebot_OpusEncoder_nativeInitEncoder(JNIEnv *env, jobject thiz,
jint sample_rate, jint channels,
jint application) {
int error;
OpusEncoder *encoder = opus_encoder_create(sample_rate, channels, application, &error);
if (error != OPUS_OK || encoder == nullptr) {
LOGE("Failed to create encoder: %s", opus_strerror(error));
return 0;
}
opus_encoder_ctl(encoder, OPUS_SET_BITRATE(64000)); // 64 kbps
opus_encoder_ctl(encoder, OPUS_SET_COMPLEXITY(10)); // 0-10, 10是最高质量
LOGI("Opus encoder initialized: sample_rate=%d, channels=%d", sample_rate, channels);
return (jlong) (intptr_t) encoder;
}
JNIEXPORT jint JNICALL
Java_info_dourok_voicebot_OpusEncoder_nativeEncodeBytes(JNIEnv *env, jobject thiz,
jlong encoder_handle,
jbyteArray input_buffer,
jint input_size,
jbyteArray output_buffer,
jint max_output_size) {
OpusEncoder *encoder = (OpusEncoder *) (intptr_t) encoder_handle;
if (encoder == nullptr) {
LOGE("Encoder handle is null");
return -1;
}
jbyte *input = env->GetByteArrayElements(input_buffer, nullptr);
jbyte *output = env->GetByteArrayElements(output_buffer, nullptr);
opus_int16 *pcm = (opus_int16 *) input;
int frame_size = input_size / 2; // 16位samples的数量
int result = opus_encode(encoder, pcm, frame_size,
(unsigned char *) output, max_output_size);
env->ReleaseByteArrayElements(input_buffer, input, JNI_ABORT);
env->ReleaseByteArrayElements(output_buffer, output, 0);
if (result < 0) {
LOGE("Encoding failed: %s", opus_strerror(result));
return -1;
}
return result;
}
JNIEXPORT void JNICALL
Java_info_dourok_voicebot_OpusEncoder_nativeReleaseEncoder(JNIEnv *env, jobject thiz,
jlong encoder_handle) {
OpusEncoder *encoder = (OpusEncoder *) (intptr_t) encoder_handle;
if (encoder != nullptr) {
opus_encoder_destroy(encoder);
LOGI("Opus encoder released");
}
}
} // extern "C"

166
app/src/main/cpp/opus_types.h Normal file
View File

@ -0,0 +1,166 @@
/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
/* Modified by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/* opus_types.h based on ogg_types.h from libogg */
/**
@file opus_types.h
@brief Opus reference implementation types
*/
#ifndef OPUS_TYPES_H
#define OPUS_TYPES_H
#define opus_int int /* used for counters etc; at least 16 bits */
#define opus_int64 long long
#define opus_int8 signed char
#define opus_uint unsigned int /* used for counters etc; at least 16 bits */
#define opus_uint64 unsigned long long
#define opus_uint8 unsigned char
/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
#include <stdint.h>
# undef opus_int64
# undef opus_int8
# undef opus_uint64
# undef opus_uint8
typedef int8_t opus_int8;
typedef uint8_t opus_uint8;
typedef int16_t opus_int16;
typedef uint16_t opus_uint16;
typedef int32_t opus_int32;
typedef uint32_t opus_uint32;
typedef int64_t opus_int64;
typedef uint64_t opus_uint64;
#elif defined(_WIN32)
# if defined(__CYGWIN__)
# include <_G_config.h>
typedef _G_int32_t opus_int32;
typedef _G_uint32_t opus_uint32;
typedef _G_int16 opus_int16;
typedef _G_uint16 opus_uint16;
# elif defined(__MINGW32__)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
# elif defined(__MWERKS__)
typedef int opus_int32;
typedef unsigned int opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
# else
/* MSVC/Borland */
typedef __int32 opus_int32;
typedef unsigned __int32 opus_uint32;
typedef __int16 opus_int16;
typedef unsigned __int16 opus_uint16;
# endif
#elif defined(__MACOS__)
# include <sys/types.h>
typedef SInt16 opus_int16;
typedef UInt16 opus_uint16;
typedef SInt32 opus_int32;
typedef UInt32 opus_uint32;
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
typedef int16_t opus_int16;
typedef u_int16_t opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16 opus_int16;
typedef u_int16 opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined (DJGPP)
/* DJGPP */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(R5900)
/* PS2 EE */
typedef int opus_int32;
typedef unsigned opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short opus_int16;
typedef unsigned short opus_uint16;
typedef signed int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef long opus_int32;
typedef unsigned long opus_uint32;
#elif defined(CONFIG_TI_C6X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#else
/* Give up, take a reasonable guess */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#endif
#endif /* OPUS_TYPES_H */

40
app/src/main/java/com/zs/smarthuman/sherpa/VadManager.kt
View File

@ -25,11 +25,14 @@ class VadManager(
private var lastActiveMs = 0L
private var speechStartMs = 0L
// 配置参数
private val MIN_RMS = 0.003f
private val END_SILENCE_MS = 200L
private val SPEECH_START_PROTECT_MS = 90L
private val RESET_IDLE_MS = 3_000L
// 配置参数优化：适配预缓存逻辑，减少开头丢失概率
private val MIN_RMS = 0.002f // 降低能量阈值，避免弱语音开头被过滤
private val END_SILENCE_MS = 100L // 保持不变
private val SPEECH_START_PROTECT_MS = 30L // 缩短语音开始保护时间（从90→50ms），更快触发onSpeechStart
private val RESET_IDLE_MS = 3_000L // 保持不变
// 新增：标记是否已触发过语音开始回调，避免重复触发
private var speechStartTriggered = false
init {
try {
@ -38,14 +41,14 @@ class VadManager(
VadModelConfig(
sileroVadModelConfig = SileroVadModelConfig(
model = "silero_vad.onnx",
threshold = 0.45F,
threshold = 0.40F, // 降低VAD检测阈值（从0.45→0.40），更灵敏检测开头语音
minSilenceDuration = 0.05F,
minSpeechDuration = 0.1F,
minSpeechDuration = 0.08F, // 缩短最小语音时长（从0.1→0.08s），更快识别短开头
windowSize = 512
)
)
)
LogUtils.i(TAG, "✅ VAD 初始化完成（线程安全版）")
LogUtils.i(TAG, "✅ VAD 初始化完成（线程安全+开头优化版）")
} catch (e: Exception) {
LogUtils.e(TAG, "❌ VAD 初始化失败", e)
throw e
@ -63,7 +66,7 @@ class VadManager(
}
val now = System.currentTimeMillis()
// 快速能量检测
// 快速能量检测：优化采样步长，提高开头检测精度
val rms = fastRms(samples)
if (rms < MIN_RMS) {
handleSilence(now)
@ -82,11 +85,13 @@ class VadManager(
if (hasSpeech) {
lastSpeechMs = now
lastActiveMs = now
if (!isSpeaking) {
// 优化：避免重复触发onSpeechStart
if (!isSpeaking && !speechStartTriggered) {
isSpeaking = true
speechStartTriggered = true // 标记已触发
speechStartMs = now
onSpeechStart()
LogUtils.d(TAG, "🗣 语音开始 | RMS: $rms | 采样数: ${samples.size}")
onSpeechStart() // 立即触发，不延迟
LogUtils.d(TAG, "🗣 语音开始 | RMS: $rms | 采样数: ${samples.size} | 时间戳: $now")
}
} else {
handleSilence(now)
@ -115,6 +120,7 @@ class VadManager(
&& now - lastSpeechMs > END_SILENCE_MS
) {
isSpeaking = false
speechStartTriggered = false // 重置触发标记
onSpeechEnd()
LogUtils.d(TAG, "🔇 语音结束 | 总时长: ${now - speechStartMs}ms")
}
@ -123,6 +129,7 @@ class VadManager(
if (!isSpeaking && now - lastActiveMs > RESET_IDLE_MS) {
try {
vad.reset()
speechStartTriggered = false // 重置触发标记
lastActiveMs = now
LogUtils.d(TAG, "🔄 VAD 空闲自动重置")
} catch (e: Exception) {
@ -131,11 +138,11 @@ class VadManager(
}
}
// 快速RMS计算逻辑保持不变
// 优化：调整RMS计算步长，提高开头弱语音的检测精度（从4→2）
private fun fastRms(samples: FloatArray): Float {
var sum = 0f
var count = 0
val step = 4
val step = 2 // 缩小采样步长，更精准计算能量
var i = 0
while (i < samples.size) {
val v = samples[i]
@ -147,17 +154,18 @@ class VadManager(
}
/**
* 线程安全的重置方法
* 线程安全的重置方法：新增重置触发标记
*/
suspend fun reset() {
vadMutex.withLock {
isSpeaking = false
speechStartTriggered = false // 关键：重置触发标记
lastSpeechMs = 0L
lastActiveMs = 0L
speechStartMs = 0L
try {
vad.reset()
LogUtils.d(TAG, "🔄 VAD 手动重置完成")
LogUtils.d(TAG, "🔄 VAD 手动重置完成（含触发标记）")
} catch (e: Exception) {
LogUtils.e(TAG, "❌ VAD 手动重置异常", e)
}

210
app/src/main/java/com/zs/smarthuman/sherpa/VoiceController.kt
View File

@ -20,14 +20,6 @@ import kotlinx.coroutines.asCoroutineDispatcher
import kotlin.math.max
import kotlin.math.min
// 1. 封装声纹验证状态（替代零散的Boolean标记）
private data class SpeakerVerifyState(
var job: Job? = null,
var finished: Boolean = false,
var passed: Boolean = true, // fail-open 默认放行
var failed: Boolean = false
)
// 2. 封装超时相关状态
private data class TimeoutState(
var idleTimeoutMs: Long,
@ -54,16 +46,11 @@ class VoiceController(
private const val SAMPLE_RATE = 16000
private const val PRE_BUFFER_SIZE = SAMPLE_RATE * 2
private const val INVALID_RESET_DEBOUNCE_MS = 1500L
private const val SPEAKER_THRESHOLD = 0.35f
private const val MIN_VERIFY_MS = 600L
private const val SPEAKER_THRESHOLD = 0.4f
private const val MIN_VERIFY_MS = 650L
private const val MAX_VERIFY_MS = 1200L
private const val KWS_OBSERVE_MS = 500L
private const val SPEECH_COOLDOWN_MS = 680L
// 声纹验证常量集中管理
private const val SPEAKER_VERIFY_NEED_SAMPLES_MS = 600L
private const val SPEAKER_VERIFY_MAX_WAIT_MS = 800L
private const val SPEAKER_VERIFY_CHECK_INTERVAL_MS = 20L
}
// ================= 核心优化：自定义IO线程池（减少调度开销） =================
@ -85,9 +72,6 @@ class VoiceController(
onStateChanged?.invoke(value)
}
// 声纹验证状态封装（替代零散的Boolean）
private val speakerVerifyState = SpeakerVerifyState()
// 超时状态封装
private val timeoutState = TimeoutState(
idleTimeoutMs = idleTimeoutSeconds * 1000L,
@ -331,7 +315,6 @@ class VoiceController(
audioBufferSize += copySize
}
startSpeakerVerify()
state = VoiceState.RECORDING
}
@ -345,84 +328,36 @@ class VoiceController(
finishSentence()
}
/* ================= 声纹验证（统一Job管理 + 原生数组优化） ================= */
private fun startSpeakerVerify() {
// 修复：彻底取消旧任务，防止回调泄漏
speakerVerifyState.job?.cancel()
speakerVerifyState.job = null
// 修复：强制重置所有声纹状态，无残留
resetSpeakerVerifyState()
// 启动新的验证任务，绑定到统一作用域
speakerVerifyState.job = coroutineScope.launch(customIoDispatcher) {
val needSamples = (SAMPLE_RATE * SPEAKER_VERIFY_NEED_SAMPLES_MS / 1000).toInt()
var waited = 0L
// 等待足够的音频样本（使用audioBufferSize）
while (audioBufferSize < needSamples && waited < SPEAKER_VERIFY_MAX_WAIT_MS) {
delay(SPEAKER_VERIFY_CHECK_INTERVAL_MS)
waited += SPEAKER_VERIFY_CHECK_INTERVAL_MS
}
if (audioBufferSize < needSamples) {
speakerVerifyState.finished = true
return@launch
}
// 优化：直接从原生数组截取，无拷贝
val startIdx = maxOf(0, audioBufferSize - needSamples)
val input = audioBuffer.copyOfRange(startIdx, audioBufferSize)
val pass = verifySpeaker(input)
// 更新状态（线程安全）
withContext(Dispatchers.Main) {
speakerVerifyState.passed = pass
speakerVerifyState.failed = !pass
speakerVerifyState.finished = true
}
}
}
private suspend fun verifySpeaker(audio: FloatArray): Boolean {
if (audio.isEmpty()) return false
val audioMs = audio.size * 1000L / SAMPLE_RATE
// 修复：严格模式下，短音频直接判定失败，不允许跳过
// if (audioMs < MIN_VERIFY_MS) {
// return if (ENABLE_STRICT_SPEAKER_VERIFY) {
// LogUtils.w(TAG, "🔴 严格模式：短音频 $audioMs ms，声纹验证失败")
// false
// } else {
// LogUtils.d(TAG, "🟡 非严格模式：短音频 $audioMs ms，跳过声纹")
// true
// }
// }
val verifyStartMs = System.currentTimeMillis()
val maxSamples = (SAMPLE_RATE * MAX_VERIFY_MS / 1000).toInt()
val input = if (audio.size > maxSamples) {
audio.copyOfRange(audio.size - maxSamples, audio.size)
private fun verifySpeaker(audio: FloatArray): Boolean {
val verifyStartNs = System.nanoTime()
val totalSamples = audio.size
val tempTarget = min((SAMPLE_RATE * MAX_VERIFY_MS / 1000).toInt(), totalSamples)
val targetSamples = max(tempTarget, (SAMPLE_RATE * MIN_VERIFY_MS / 1000).toInt())
val midStartIndex = (totalSamples - targetSamples) / 2
val finalAudio = if (midStartIndex >= 0 && midStartIndex + targetSamples <= totalSamples) {
audio.copyOfRange(midStartIndex, midStartIndex + targetSamples)
} else {
audio
val verifyCostMs = (System.nanoTime() - verifyStartNs) / 1_000_000
LogUtils.d(TAG, "🔍 声纹验证耗时: ${verifyCostMs}ms | 结果: 失败（有效语音过短）")
return false
}
return withContext(customIoDispatcher) {
return try {
var stream: OnlineStream? = null
try {
stream = SpeakerRecognition.extractor.createStream()
stream.acceptWaveform(input, SAMPLE_RATE)
stream.acceptWaveform(finalAudio, SAMPLE_RATE) // 用中间段音频验证
stream.inputFinished()
if (!SpeakerRecognition.extractor.isReady(stream)) {
LogUtils.w(TAG, "⚠️ stream not ready，验证失败")
return@withContext false
val verifyCostMs = (System.nanoTime() - verifyStartNs) / 1_000_000
LogUtils.d(TAG, "🔍 声纹验证耗时: ${verifyCostMs}ms | 结果: 失败（Stream未就绪）")
return false
}
val embedding = SpeakerRecognition.extractor.compute(stream)
val pass = speakerManagerLock.withLock {
val result = speakerManagerLock.withLock {
SpeakerRecognition.manager.verify(
CURRENT_USER_ID,
embedding,
@ -430,76 +365,60 @@ class VoiceController(
)
}
val cost = System.currentTimeMillis() - verifyStartMs
LogUtils.d(
TAG,
"📊 声纹 | pass=$pass | 音频=${audioMs}ms | 输入=${input.size} | 耗时=${cost}ms"
)
// ================ 核心修复：移除状态判断，声纹失败强制触发终止流程 ================
if (!pass) {
withContext(Dispatchers.Main.immediate) {
// 标记失败状态
speakerVerifyState.failed = true
speakerVerifyState.finished = true
LogUtils.d(TAG, "🔴 声纹验证不通过，安全终止录音流程")
// 核心：主动结束录音，走标准finishSentence闭环，不暴力中断
finishSentence()
}
}
pass
} catch (e: Exception) {
LogUtils.e(TAG, "❌ 声纹异常，验证失败", e)
false
// 计算总耗时（毫秒级，易读）
val verifyCostMs = (System.nanoTime() - verifyStartNs) / 1_000_000
LogUtils.d(TAG, "🔍 声纹验证耗时: ${verifyCostMs}ms | 结果: ${if (result) "通过" else "拒绝"}")
return result
} finally {
stream?.release()
}
} catch (e: Exception) {
val verifyCostMs = (System.nanoTime() - verifyStartNs) / 1_000_000
LogUtils.d(TAG, "🔍 声纹验证耗时: ${verifyCostMs}ms | 结果: 失败（异常）")
return false
}
}
/* ================= 结束录音（优化：原生数组读取） ================= */
private fun finishSentence() {
// 取消旧任务
speakerVerifyState.job?.cancel()
speakerVerifyState.job = null
val now = cachedTimeMs
val duration = if (recordingStartMs != 0L) now - recordingStartMs else 0L
// 声纹失败优先拦截，直接重置，不上传音频
if (speakerVerifyState.failed) {
LogUtils.w(TAG, "❌ 声纹标记为失败，强制拒绝本次语音")
timeoutState.hasInvalidSpeech = true
// 重置到可交互状态，用户可以立即重新说话
resetToWaitSpeech()
return
}
recordingStartMs = 0L
vadStarted = false
// ============ 原有正常逻辑保留 ============
val isSpeakerVerifyFailed = (ENABLE_STRICT_SPEAKER_VERIFY && speakerVerifyState.finished && !speakerVerifyState.passed)
if (isSpeakerVerifyFailed) {
LogUtils.w(TAG, "❌ 声纹验证未通过，拒绝本次语音")
timeoutState.hasInvalidSpeech = true
resetToWaitSpeech()
return
}
// 无音频防护
// 无音频直接拒绝
if (audioBufferSize <= 0) {
LogUtils.w(TAG, "❌ 无有效音频数据，拒绝上传")
LogUtils.w(TAG, "❌ 无有效音频，丢弃")
resetToWaitSpeech()
return
}
// 正常上传逻辑
// 拷贝最终音频（此时 buffer 不再变化）
val audio = audioBuffer.copyOfRange(0, audioBufferSize)
clearAudioBuffer()
recordingStartMs = 0L
// ================= 声纹验证：只在这里做 =================
if (ENABLE_STRICT_SPEAKER_VERIFY) {
val pass = runCatching {
verifySpeaker(audio)
}.getOrElse {
LogUtils.e(TAG, "❌ 声纹异常，拒绝本次语音", it)
false
}
if (!pass) {
LogUtils.w(TAG, "🔴 声纹验证失败，拒绝上传")
timeoutState.hasInvalidSpeech = true
resetToWaitSpeech()
return
}
}
// ================= 通过后才上传 =================
timeoutState.hasInvalidSpeech = false
state = VoiceState.UPLOADING
onFinalAudio(audio)
timeoutState.hasInvalidSpeech = false
LogUtils.i(TAG, "✅ 语音通过 | 时长: $duration ms")
LogUtils.i(TAG, "✅ 语音通过 | 时长: ${duration}ms")
}
/* ================= 播放/上传回调（优化：精简状态判断） ================= */
@ -560,21 +479,11 @@ class VoiceController(
clearAudioBuffer()
synchronized(preBufferLock) { preBuffer.clear() }
// 声纹失败场景：全量重置
if (speakerVerifyState.failed) {
LogUtils.d(TAG, "🛡 声纹失败，跳过防抖，强制全量重置")
resetSpeakerVerifyState()
timeoutState.lastInvalidResetMs = 0
timeoutState.waitSpeechFailStartMs = now
return
}
// 原有防抖逻辑
if (now - timeoutState.lastInvalidResetMs < INVALID_RESET_DEBOUNCE_MS) {
return
}
resetSpeakerVerifyState()
timeoutState.lastInvalidResetMs = now
timeoutState.waitSpeechFailStartMs = now
}
@ -599,21 +508,9 @@ class VoiceController(
timeoutState.hasInvalidSpeech = false
timeoutState.currentType = TimeoutType.IDLE_TIMEOUT
// 重置声纹状态
resetSpeakerVerifyState()
state = VoiceState.WAIT_WAKEUP
}
// 声纹状态重置（统一方法，修复：彻底清空所有标记）
private fun resetSpeakerVerifyState() {
speakerVerifyState.job?.cancel()
speakerVerifyState.job = null
speakerVerifyState.finished = false
speakerVerifyState.passed = true
speakerVerifyState.failed = false
}
/* ================= 资源管理（优化：关闭自定义线程池） ================= */
fun release() {
LogUtils.d(TAG, "🔌 释放资源")
@ -634,7 +531,6 @@ class VoiceController(
// 重置状态
timeoutState.hasInvalidSpeech = false
timeoutState.currentType = TimeoutType.IDLE_TIMEOUT
resetSpeakerVerifyState()
try {
SpeakerRecognition.extractor.release()

19
app/src/main/java/com/zs/smarthuman/sherpa/WakeupManager.kt
View File

@ -35,7 +35,7 @@ class WakeupManager(assetManager: AssetManager, function: () -> Unit) {
featConfig = featConfig,
modelConfig = modelConfig,
keywordsFile = keywordsFile,
keywordsThreshold = 0.1f
keywordsThreshold = 0.06f
)
kws = KeywordSpotter(assetManager, config)
@ -49,11 +49,8 @@ class WakeupManager(assetManager: AssetManager, function: () -> Unit) {
/** ⭐ 永远喂 KWS */
fun acceptAudio(samples: FloatArray) {
val s = stream ?: return
for (i in samples.indices) {
samples[i] *= 2.5f
}
s.acceptWaveform(samples, sampleRate)
normalize(samples)
while (kws.isReady(s)) {
kws.decode(s)
val keyword = kws.getResult(s).keyword
@ -73,6 +70,18 @@ class WakeupManager(assetManager: AssetManager, function: () -> Unit) {
return r
}
private fun normalize(samples: FloatArray, targetRms: Float = 0.05f) {
var sum = 0.0
for (s in samples) sum += s * s
val rms = kotlin.math.sqrt(sum / samples.size)
if (rms < 1e-6) return
val gain = targetRms / rms
for (i in samples.indices) {
samples[i] = (samples[i] * gain).coerceIn(-1.0, 1.0).toFloat()
}
}
fun reset() {
stream?.let { kws.reset(it) }
justWokeUp = false

76
app/src/main/java/com/zs/smarthuman/utils/OpusDecoder.kt Normal file
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@ -0,0 +1,76 @@
package com.zs.smarthuman.utils;
import android.util.Log
import kotlinx.coroutines.Dispatchers
import kotlinx.coroutines.withContext
class OpusDecoder(
private val sampleRate: Int,
private val channels: Int,
frameSizeMs: Int
) {
companion object {
private const val TAG = "OpusDecoder"
init {
System.loadLibrary("app")
}
}
private var nativeDecoderHandle: Long = 0
private val frameSize: Int = (sampleRate * frameSizeMs) / 1000
init {
nativeDecoderHandle = nativeInitDecoder(sampleRate, channels)
if (nativeDecoderHandle == 0L) {
throw IllegalStateException("Failed to initialize Opus decoder")
}
}
// 使用协程进行解码，运行在 IO 线程
suspend fun decode(opusData: ByteArray): ByteArray? = withContext(Dispatchers.IO) {
val maxPcmSize = frameSize * channels * 2 // 16-bit PCM
val pcmBuffer = ByteArray(maxPcmSize)
val decodedBytes = nativeDecodeBytes(
nativeDecoderHandle,
opusData,
opusData.size,
pcmBuffer,
maxPcmSize
)
if (decodedBytes > 0) {
if (decodedBytes < pcmBuffer.size) {
pcmBuffer.copyOf(decodedBytes)
} else {
pcmBuffer
}
} else {
Log.e(TAG, "Failed to decode frame")
null
}
}
fun release() {
if (nativeDecoderHandle != 0L) {
nativeReleaseDecoder(nativeDecoderHandle)
nativeDecoderHandle = 0
}
}
protected fun finalize() {
release()
}
private external fun nativeInitDecoder(sampleRate: Int, channels: Int): Long
private external fun nativeDecodeBytes(
decoderHandle: Long,
inputBuffer: ByteArray,
inputSize: Int,
outputBuffer: ByteArray,
maxOutputSize: Int
): Int
private external fun nativeReleaseDecoder(decoderHandle: Long)
}

75
app/src/main/java/com/zs/smarthuman/utils/OpusEncoder.kt Normal file
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@ -0,0 +1,75 @@
package com.zs.smarthuman.utils;
import android.util.Log
import kotlinx.coroutines.Dispatchers
import kotlinx.coroutines.withContext
class OpusEncoder(
private val sampleRate: Int,
private val channels: Int,
frameSizeMs: Int
) {
companion object {
private const val TAG = "OpusEncoder"
init {
System.loadLibrary("app")
}
}
private var nativeEncoderHandle: Long = 0
private val frameSize: Int = (sampleRate * frameSizeMs) / 1000
init {
nativeEncoderHandle = nativeInitEncoder(sampleRate, channels, 2048) // OPUS_APPLICATION_VOIP
if (nativeEncoderHandle == 0L) {
throw IllegalStateException("Failed to initialize Opus encoder")
}
}
suspend fun encode(pcmData: ByteArray): ByteArray? = withContext(Dispatchers.IO) {
val frameBytes = frameSize * channels * 2 // 16-bit PCM
if (pcmData.size != frameBytes) {
Log.e(TAG, "Input buffer size must be $frameBytes bytes (got ${pcmData.size})")
return@withContext null
}
val outputBuffer = ByteArray(frameBytes) // 分配足够大的缓冲区
val encodedBytes = nativeEncodeBytes(
nativeEncoderHandle,
pcmData,
pcmData.size,
outputBuffer,
outputBuffer.size
)
if (encodedBytes > 0) {
outputBuffer.copyOf(encodedBytes)
} else {
Log.e(TAG, "Failed to encode frame")
null
}
}
fun release() {
if (nativeEncoderHandle != 0L) {
nativeReleaseEncoder(nativeEncoderHandle)
nativeEncoderHandle = 0
}
}
protected fun finalize() {
release()
}
private external fun nativeInitEncoder(sampleRate: Int, channels: Int, application: Int): Long
private external fun nativeEncodeBytes(
encoderHandle: Long,
inputBuffer: ByteArray,
inputSize: Int,
outputBuffer: ByteArray,
maxOutputSize: Int
): Int
private external fun nativeReleaseEncoder(encoderHandle: Long)
}

459
app/src/main/java/com/zs/smarthuman/websocket/WebsocketProtocol.kt Normal file
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@ -0,0 +1,459 @@
package com.zs.smarthuman.websocket
import com.blankj.utilcode.util.LogUtils
import com.blankj.utilcode.util.NetworkUtils
import com.zs.smarthuman.common.UserInfoManager
import com.zs.smarthuman.utils.AESUtils
import kotlinx.coroutines.*
import kotlinx.coroutines.flow.MutableSharedFlow
import okhttp3.*
import okio.ByteString
import org.json.JSONObject
import java.util.UUID
import java.util.concurrent.TimeUnit
import java.util.concurrent.atomic.AtomicBoolean
import java.util.concurrent.atomic.AtomicInteger
import java.util.concurrent.locks.ReentrantLock
import kotlin.concurrent.withLock
class WebsocketProtocol private constructor() {
companion object {
// 日志TAG
private const val TAG = "WS"
// WebSocket服务地址
private const val WS_URL = "ws://10.10.4.132:9001/aidialogue"
// 重连配置
private const val RECONNECT_DELAY = 3_000L // 重连基础延迟
private const val MAX_RECONNECT_COUNT = -1 // -1表示无限重连
private const val NETWORK_CHECK_INTERVAL = 5_000L // 网络检测间隔
// 线程安全的单例实现
val INSTANCE: WebsocketProtocol by lazy(mode = LazyThreadSafetyMode.SYNCHRONIZED) {
WebsocketProtocol()
}
}
// 公开Flow
val incomingJsonFlow = MutableSharedFlow<JSONObject>(replay = 0)
val incomingByteArrayFlow = MutableSharedFlow<ByteArray>(replay = 0)
val networkErrorFlow = MutableSharedFlow<String>(replay = 0)
// 公开属性
var sessionId: String = "your_session_id"
// 线程安全状态标记（核心：所有标记都要闭环管理）
private val isOpen = AtomicBoolean(false) // 连接是否已打开
private val isAuthSuccess = AtomicBoolean(false) // 认证是否成功
private val reconnectCount = AtomicInteger(0) // 重连次数
private val isReconnecting = AtomicBoolean(false) // 是否正在重连（核心标记）
private val isOpening = AtomicBoolean(false) // 是否正在执行打开操作
private val connectLock = ReentrantLock() // 连接操作锁
// 协程作用域
private val scope = CoroutineScope(Dispatchers.IO + SupervisorJob() + CoroutineName("WS-Scope"))
// WebSocket实例（volatile保证可见性）
@Volatile
private var websocket: WebSocket? = null
// OkHttp客户端
private val client by lazy {
OkHttpClient.Builder()
.connectTimeout(10, TimeUnit.SECONDS)
.readTimeout(10, TimeUnit.SECONDS)
.writeTimeout(10, TimeUnit.SECONDS)
.pingInterval(3, TimeUnit.SECONDS)
.retryOnConnectionFailure(false)
.build()
}
/**
* 发送音频数据
*/
fun sendAudio(data: ByteArray) {
scope.launch {
runCatching {
LogUtils.i(TAG, "Sending audio: ${data.size} bytes")
if (isOpen.get() && isAuthSuccess.get() && websocket != null) {
websocket?.send(ByteString.of(*data))
} else {
val errorMsg = "WebSocket not ready (open:${isOpen.get()}, auth:${isAuthSuccess.get()})"
LogUtils.eTag(TAG, errorMsg)
networkErrorFlow.emit(errorMsg)
launchReconnect()
}
}.onFailure { e ->
LogUtils.eTag(TAG, "Send audio failed: ${e.message}", e)
networkErrorFlow.emit("Send audio error: ${e.message}")
}
}
}
/**
* 发送文本数据
*/
fun sendText(text: String) {
scope.launch {
runCatching {
LogUtils.i(TAG, "Sending text: $text")
if (isOpen.get() && isAuthSuccess.get() && websocket != null) {
websocket?.send(text)
} else {
val errorMsg = "WebSocket not ready (open:${isOpen.get()}, auth:${isAuthSuccess.get()})"
LogUtils.eTag(TAG, errorMsg)
networkErrorFlow.emit(errorMsg)
launchReconnect()
}
}.onFailure { e ->
LogUtils.eTag(TAG, "Send text failed: ${e.message}", e)
networkErrorFlow.emit("Send text error: ${e.message}")
}
}
}
/**
* 判断通道是否就绪
*/
fun isAudioChannelOpened(): Boolean {
return websocket != null && isOpen.get() && isAuthSuccess.get()
}
/**
* 关闭通道
*/
fun closeAudioChannel() {
scope.launch {
connectLock.withLock {
LogUtils.iTag(TAG, "Close audio channel")
isOpen.set(false)
isAuthSuccess.set(false)
reconnectCount.set(0)
isReconnecting.set(false) // 关闭时重置重连标记
isOpening.set(false) // 关闭时重置打开标记
val ws = websocket
websocket = null
ws?.close(1000, "Normal closure")
}
}
}
/**
* 打开通道（核心修复：标记闭环）
*/
suspend fun openAudioChannel(): Boolean = withContext(Dispatchers.IO) {
// 1. 防并发：只有未打开时才执行
if (!isOpening.compareAndSet(false, true)) {
LogUtils.wTag(TAG, "openAudioChannel is running, skip")
return@withContext false
}
var result = false
connectLock.withLock {
try {
// 2. 如果已有连接，直接返回成功并重置标记
if (websocket != null && isOpen.get()) {
LogUtils.iTag(TAG, "WebSocket already connected, skip open")
result = true
isReconnecting.set(false) // 关键：重置重连标记
return@withContext result
}
// 3. 检查网络
if (!NetworkUtils.isConnected()) {
val errorMsg = "Network disconnected, can't connect WebSocket"
LogUtils.eTag(TAG, errorMsg)
scope.launch {
networkErrorFlow.emit(errorMsg)
}
result = false
return@withContext result
}
// 4. 创建新连接（核心：先创建再赋值）
val request = Request.Builder().url(WS_URL).build()
val newWs = client.newWebSocket(request, createWebSocketListener())
websocket = newWs
LogUtils.iTag(TAG, "WebSocket connecting to: $WS_URL")
result = true
isReconnecting.set(false) // 关键：发起连接后重置重连标记
} catch (e: Exception) {
val errorMsg = "Create WebSocket failed: ${e.message}"
LogUtils.eTag(TAG, errorMsg, e)
scope.launch {
networkErrorFlow.emit(errorMsg)
}
websocket = null
result = false
} finally {
isOpening.set(false) // 无论成败，最终重置打开标记
}
}
return@withContext result
}
/**
* 创建WebSocket监听器
*/
private fun createWebSocketListener(): WebSocketListener {
return object : WebSocketListener() {
override fun onOpen(webSocket: WebSocket, response: Response) {
connectLock.withLock {
if (websocket === webSocket) {
LogUtils.iTag(TAG, "✅ WebSocket connected, start auth")
isOpen.set(true)
isAuthSuccess.set(false)
reconnectCount.set(0) // 连接成功重置重连次数
isReconnecting.set(false) // 关键：连接成功重置重连标记
sendAuthMessage(webSocket)
} else {
LogUtils.wTag(TAG, "Ignore onOpen for old WebSocket")
webSocket.close(1001, "New connection exists")
}
}
}
override fun onMessage(webSocket: WebSocket, text: String) {
if (websocket !== webSocket) {
LogUtils.wTag(TAG, "Ignore text message for old WebSocket")
return
}
LogUtils.iTag(TAG, "📩 Receive text: $text")
scope.launch {
runCatching {
val json = JSONObject(text)
when (json.optInt("msgContentType")) {
WebsocketType.AUTH.code -> parseAuthResponse(json)
WebsocketType.HEARTBEAT.code -> sendHeartbeat()
else -> incomingJsonFlow.emit(json)
}
}.onFailure { e ->
val errorMsg = "Parse text message error: ${e.message}"
LogUtils.eTag(TAG, errorMsg, e)
networkErrorFlow.emit(errorMsg)
}
}
}
override fun onMessage(webSocket: WebSocket, bytes: ByteString) {
if (websocket !== webSocket) {
LogUtils.wTag(TAG, "Ignore binary message for old WebSocket")
return
}
val size = bytes.size
LogUtils.iTag(TAG, "📩 Receive binary: $size bytes")
if (isAuthSuccess.get()) {
scope.launch { incomingByteArrayFlow.emit(bytes.toByteArray()) }
}
}
override fun onClosed(webSocket: WebSocket, code: Int, reason: String) {
connectLock.withLock {
if (websocket === webSocket) {
LogUtils.wTag(TAG, "🔌 WebSocket closed: $code, reason: $reason")
isOpen.set(false)
isAuthSuccess.set(false)
websocket = null
isReconnecting.set(false) // 关闭时重置重连标记
} else {
LogUtils.wTag(TAG, "Ignore onClosed for old WebSocket")
}
}
// 非主动关闭才重连
val isNormalClose = code == 1000 && reason == "Normal closure"
if (!isNormalClose) {
launchReconnect()
}
}
override fun onFailure(webSocket: WebSocket, t: Throwable, response: Response?) {
connectLock.withLock {
if (websocket === webSocket) {
LogUtils.eTag(TAG, "❌ WebSocket failure: ${t.message}", t)
isOpen.set(false)
isAuthSuccess.set(false)
websocket = null
} else {
LogUtils.wTag(TAG, "Ignore onFailure for old WebSocket")
}
}
scope.launch {
networkErrorFlow.emit(t.message ?: "Unknown WebSocket error")
}
launchReconnect()
}
}
}
/**
* 发送认证消息
*/
private fun sendAuthMessage(webSocket: WebSocket) {
if (websocket !== webSocket) {
LogUtils.wTag(TAG, "Ignore auth for old WebSocket")
return
}
scope.launch {
runCatching {
val authData = JSONObject().apply {
put("unionId", AESUtils.encrypt(UserInfoManager.sn))
}
val authMsg = buildWsMessage(WebsocketType.AUTH, authData)
LogUtils.iTag(TAG, "🔑 Send auth: $authMsg")
webSocket.send(authMsg.toString())
}.onFailure { e ->
val errorMsg = "Send auth failed: ${e.message}"
LogUtils.eTag(TAG, errorMsg, e)
networkErrorFlow.emit(errorMsg)
}
}
}
/**
* 发送心跳
*/
private fun sendHeartbeat() {
scope.launch {
runCatching {
val heartbeatMsg = buildWsMessage(WebsocketType.HEARTBEAT, "pong")
LogUtils.iTag(TAG, "💓 Send heartbeat: $heartbeatMsg")
sendText(heartbeatMsg.toString())
}.onFailure { e ->
val errorMsg = "Send heartbeat failed: ${e.message}"
LogUtils.eTag(TAG, errorMsg, e)
networkErrorFlow.emit(errorMsg)
}
}
}
/**
* 构建通用消息
*/
private fun buildWsMessage(type: WebsocketType, data: Any): JSONObject {
return JSONObject().apply {
put("msgContentType", type.code)
put("timeStamp", System.currentTimeMillis())
put("msgId", UUID.randomUUID().toString())
put("data", data)
}
}
/**
* 解析认证响应
*/
private fun parseAuthResponse(root: JSONObject) {
scope.launch {
runCatching {
val dataObj = root.optJSONObject("data")
?: throw IllegalArgumentException("Auth response missing 'data' field")
val code = dataObj.optInt("code")
val msg = dataObj.optString("msg")
val authResult = dataObj.optString("data")
if (code == 200 && authResult == "WS_AUTH_SUCCESS") {
isAuthSuccess.set(true)
LogUtils.iTag(TAG, "✅ Auth success")
} else {
isAuthSuccess.set(false)
val errorMsg = "Auth failed: $msg (code:$code)"
LogUtils.eTag(TAG, errorMsg)
networkErrorFlow.emit(errorMsg)
launchReconnect()
}
}.onFailure { e ->
isAuthSuccess.set(false)
val errorMsg = "Parse auth response error: ${e.message}"
LogUtils.eTag(TAG, errorMsg, e)
networkErrorFlow.emit(errorMsg)
launchReconnect()
}
}
}
/**
* 核心修复：重连逻辑（标记闭环管理）
*/
private fun launchReconnect() {
// 1. 防止重复进入重连
if (!isReconnecting.compareAndSet(false, true)) {
LogUtils.wTag(TAG, "Already reconnecting, skip launchReconnect")
return
}
// 2. 协程执行重连
scope.launch(Dispatchers.IO) {
try {
// 循环重连直到成功/达到最大次数
while (true) {
// 检查最大重连次数
if (MAX_RECONNECT_COUNT != -1 && reconnectCount.get() >= MAX_RECONNECT_COUNT) {
val errorMsg = "Reach max reconnect count: $MAX_RECONNECT_COUNT"
LogUtils.eTag(TAG, errorMsg)
networkErrorFlow.emit(errorMsg)
break
}
val currentCount = reconnectCount.incrementAndGet()
LogUtils.wTag(TAG, "🔄 Reconnect attempt: $currentCount, delay: ${RECONNECT_DELAY}ms")
// 等待重连延迟
delay(RECONNECT_DELAY)
// 检查网络
if (!NetworkUtils.isConnected()) {
LogUtils.wTag(TAG, "🔌 Network disconnected, wait ${NETWORK_CHECK_INTERVAL}ms")
delay(NETWORK_CHECK_INTERVAL)
continue // 网络未恢复，继续循环
}
// 网络恢复，尝试连接
LogUtils.iTag(TAG, "🔌 Network available, try to connect")
val connectSuccess = openAudioChannel()
// 连接成功则退出循环
if (connectSuccess) {
LogUtils.iTag(TAG, "✅ Reconnect success")
break
}
// 连接失败，继续循环
LogUtils.wTag(TAG, "❌ Reconnect attempt $currentCount failed")
}
} catch (e: Exception) {
LogUtils.eTag(TAG, "Reconnect loop error: ${e.message}", e)
networkErrorFlow.emit("Reconnect error: ${e.message}")
} finally {
// 最终必须重置重连标记（核心修复）
isReconnecting.set(false)
LogUtils.iTag(TAG, "Reconnect loop finished, reset isReconnecting to false")
}
}
}
/**
* 释放资源
*/
fun dispose() {
LogUtils.iTag(TAG, "🗑️ Dispose WebSocketProtocol")
scope.cancel("Dispose called")
connectLock.withLock {
closeAudioChannel()
websocket = null
isReconnecting.set(false)
isOpening.set(false)
}
client.dispatcher.executorService.shutdown()
client.connectionPool.evictAll()
}
}

11
app/src/main/java/com/zs/smarthuman/websocket/WebsocketType.kt Normal file
View File

@ -0,0 +1,11 @@
package com.zs.smarthuman.websocket
/**
* @description:
* @author: lrs
* @date: 2026/2/10 18:59
*/
enum class WebsocketType(val code: Int) {
AUTH(1001),
HEARTBEAT(1002);
}

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app/src/main/jniLibs/arm64-v8a/libopus.so Normal file
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app/src/main/jniLibs/armeabi-v7a/libopus.so Normal file
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4
gradle/libs.versions.toml
View File

@ -35,7 +35,7 @@ immersionbar = "3.2.2"
immersionbarKtx = "3.2.2"
immersionbarComponents = "3.2.2"
lifecycleRuntimeAndroid = "2.9.1"
opus = "1.3.1"
[libraries]
android-spinkit = { module = "com.github.ybq:Android-SpinKit", version.ref = "androidSpinkit" }
androidautosize = { module = "com.github.JessYanCoding:AndroidAutoSize", version.ref = "androidautosize" }
@ -73,7 +73,7 @@ immersionbar-components = { module = "com.geyifeng.immersionbar:immersionbar-com
immersionbar-ktx = { module = "com.geyifeng.immersionbar:immersionbar-ktx", version.ref = "immersionbarKtx" }
immersionbar = { module = "com.geyifeng.immersionbar:immersionbar", version.ref = "immersionbar" }
androidx-lifecycle-runtime-android = { group = "androidx.lifecycle", name = "lifecycle-runtime-android", version.ref = "lifecycleRuntimeAndroid" }
opus-v131 = { module = "com.fpliu.ndk.pkg.prefab.android.21:opus", version.ref = "opus" }
[plugins]
android-application = { id = "com.android.application", version.ref = "agp" }
kotlin-android = { id = "org.jetbrains.kotlin.android", version.ref = "kotlin" }

7
settings.gradle
View File

@ -16,6 +16,10 @@ pluginManagement {
maven { setUrl("https://maven.aliyun.com/repository/public") }
maven { url "https://jitpack.io" }
maven { url "https://s01.oss.sonatype.org/content/groups/public" }
maven {
url = uri("https://raw.githubusercontent.com/leleliu008/ndk-pkg-prefab-aar-maven-repo/master")
}
}
resolutionStrategy {
@ -39,6 +43,9 @@ dependencyResolutionManagement {
mavenCentral()
maven { url "https://jitpack.io" }
maven { url "https://s01.oss.sonatype.org/content/groups/public" }
maven {
url = uri("https://raw.githubusercontent.com/leleliu008/ndk-pkg-prefab-aar-maven-repo/master")
}
}
}