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added files from limelight from android

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Diego Waxemberg
2013-12-03 23:53:12 -05:00
parent e4894b71f4
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limelight-pc/.classpath Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<classpath>
<classpathentry kind="src" path="src"/>
<classpathentry kind="con" path="org.eclipse.jdt.launching.JRE_CONTAINER/org.eclipse.jdt.internal.debug.ui.launcher.StandardVMType/JavaSE-1.6"/>
<classpathentry kind="output" path="bin"/>
</classpath>

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<?xml version="1.0" encoding="UTF-8"?>
<projectDescription>
<name>limelight-pc</name>
<comment></comment>
<projects>
</projects>
<buildSpec>
<buildCommand>
<name>org.eclipse.jdt.core.javabuilder</name>
<arguments>
</arguments>
</buildCommand>
</buildSpec>
<natures>
<nature>org.eclipse.jdt.core.javanature</nature>
</natures>
</projectDescription>

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limelight-pc/.settings/org.eclipse.jdt.core.prefs Normal file
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eclipse.preferences.version=1
org.eclipse.jdt.core.compiler.codegen.inlineJsrBytecode=enabled
org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.6
org.eclipse.jdt.core.compiler.compliance=1.6
org.eclipse.jdt.core.compiler.problem.assertIdentifier=error
org.eclipse.jdt.core.compiler.problem.enumIdentifier=error
org.eclipse.jdt.core.compiler.source=1.6

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limelight-pc/jni/nv_opus_dec/libopus/inc/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 1 byte, 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(4);
/** 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);
/**@}*/
/** @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 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);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_H */

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@@ -0,0 +1,329 @@
/* 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 CELT_C
#define OPUS_CUSTOM_EXPORT_STATIC static 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);
/* 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);
/** 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);
/** 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_STATIC int opus_custom_encoder_init(
OpusCustomEncoder *st,
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** 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);
/* 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);
/** 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);
/** 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);
/** 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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/* 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
/** The mode struct passed is invalid @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)
# ifdef OPUS_BUILD
# 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
/**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
/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
/* 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)))
/** @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*/
/**@}*/
/** @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.
* @warning Only the MDCT mode of Opus can provide hard CBR behavior.
* @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 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)
/** 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 with trigger progressively more loss resistant behavior in the encoder
* at the expense of quality at a given bitrate in the lossless case, 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.
* @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)
/** 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)
/**@}*/
/** @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 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)
/** 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)
/**@}*/
/** @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)
/**@}*/
/** @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.
*
* @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 the
* <a href="http://tools.ietf.org/html/draft-terriberry-oggopus">Ogg
* encapsulation specification</a> and is based on the self-delimited Opus
* framing described in Appendix B of <a href="http://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="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">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(5);
/** 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(6);
/** 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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limelight-pc/jni/nv_opus_dec/libopus/inc/opus_types.h Normal file
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/* (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
/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
#include <stdint.h>
typedef int16_t opus_int16;
typedef uint16_t opus_uint16;
typedef int32_t opus_int32;
typedef uint32_t opus_uint32;
#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
#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
#endif /* OPUS_TYPES_H */

54
limelight-pc/jni/nv_opus_dec/libopus/nv_opus_dec.c Normal file
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#include <stdlib.h>
#include <opus.h>
#include "nv_opus_dec.h"
OpusDecoder* decoder;
// This function must be called before
// any other decoding functions
int nv_opus_init(void) {
int err;
decoder = opus_decoder_create(
nv_opus_get_sample_rate(),
nv_opus_get_channel_count(),
&err);
return err;
}
// This function must be called after
// decoding is finished
void nv_opus_destroy(void) {
if (decoder != NULL) {
opus_decoder_destroy(decoder);
}
}
// The Opus stream is stereo
int nv_opus_get_channel_count(void) {
return 2;
}
// This number assumes 2 channels at 48 KHz
int nv_opus_get_max_out_shorts(void) {
return 512*nv_opus_get_channel_count();
}
// The Opus stream is 48 KHz
int nv_opus_get_sample_rate(void) {
return 48000;
}
// outpcmdata must be 5760*2 shorts in length
// packets must be decoded in order
// a packet loss must call this function with NULL indata and 0 inlen
// returns the number of decoded samples
int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata) {
int err;
// Decoding to 16-bit PCM with FEC off
// Maximum length assuming 48KHz sample rate
err = opus_decode(decoder, indata, inlen,
outpcmdata, 512, 0);
return err;
}

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limelight-pc/jni/nv_opus_dec/libopus/nv_opus_dec.h Normal file
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int nv_opus_init(void);
void nv_opus_destroy(void);
int nv_opus_get_channel_count(void);
int nv_opus_get_max_out_shorts(void);
int nv_opus_get_sample_rate(void);
int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata);

68
limelight-pc/jni/nv_opus_dec/libopus/nv_opus_dec_jni.c Normal file
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#include "nv_opus_dec.h"
#include <stdlib.h>
#include <jni.h>
// This function must be called before
// any other decoding functions
JNIEXPORT jint JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_init(JNIEnv *env, jobject this) {
return nv_opus_init();
}
// This function must be called after
// decoding is finished
JNIEXPORT void JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_destroy(JNIEnv *env, jobject this) {
nv_opus_destroy();
}
// The Opus stream is stereo
JNIEXPORT jint JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_getChannelCount(JNIEnv *env, jobject this) {
return nv_opus_get_channel_count();
}
// This number assumes 2 channels at 48 KHz
JNIEXPORT jint JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_getMaxOutputShorts(JNIEnv *env, jobject this) {
return nv_opus_get_max_out_shorts();
}
// The Opus stream is 48 KHz
JNIEXPORT jint JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_getSampleRate(JNIEnv *env, jobject this) {
return nv_opus_get_sample_rate();
}
// outpcmdata must be 5760*2 shorts in length
// packets must be decoded in order
// a packet loss must call this function with NULL indata and 0 inlen
// returns the number of decoded samples
JNIEXPORT jint JNICALL
Java_com_limelight_nvstream_av_audio_OpusDecoder_decode(
JNIEnv *env, jobject this, // JNI parameters
jbyteArray indata, jint inoff, jint inlen, // Input parameters
jshortArray outpcmdata) // Output parameter
{
jint ret;
jbyte* jni_input_data;
jshort* jni_pcm_data;
jni_pcm_data = (*env)->GetShortArrayElements(env, outpcmdata, 0);
if (indata != NULL) {
jni_input_data = (*env)->GetByteArrayElements(env, indata, 0);
ret = nv_opus_decode(&jni_input_data[inoff], inlen, jni_pcm_data);
// The input data isn't changed so it can be safely aborted
(*env)->ReleaseByteArrayElements(env, indata, jni_input_data, JNI_ABORT);
}
else {
ret = nv_opus_decode(NULL, 0, jni_pcm_data);
}
(*env)->ReleaseShortArrayElements(env, outpcmdata, jni_pcm_data, 0);
return ret;
}

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limelight-pc/src/com/limelight/#Limelight.java# Normal file
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package com.limelight;
import com.limelight.gui.MainFrame;
import com.limelight.gui.StreamFrame;
public class Limelight {
public static final double VERSION = 1.0;
private final String HOST;
public Limelight(String host) {
this.HOST = host;
}
private void startUp() {
StreamFrame streamFrame = new StreamFrame();
streamFrame.build();
}
public static void createInstance(String host) {
Limelight limelight = new Limelight(host);
limelight.startUp();
}
public static void main(String args[]) {
MainFrame limeFrame = new MainFrame();
limeFrame.build();
}
}

66
limelight-pc/src/com/limelight/Limelight.java Normal file
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package com.limelight;
import javax.swing.JOptionPane;
import com.limelight.gui.MainFrame;
import com.limelight.gui.StreamFrame;
import com.limelight.nvstream.NvConnection;
import com.limelight.nvstream.NvConnectionListener;
public class Limelight implements NvConnectionListener {
public static final double VERSION = 1.0;
private String host;
private StreamFrame streamFrame;
private NvConnection conn;
public Limelight(String host) {
this.host = host;
}
private void startUp() {
streamFrame = new StreamFrame();
streamFrame.build();
conn = new NvConnection(host, streamFrame, this);
conn.start();
}
public static void createInstance(String host) {
Limelight limelight = new Limelight(host);
limelight.startUp();
}
public static void main(String args[]) {
MainFrame limeFrame = new MainFrame();
limeFrame.build();
}
@Override
public void stageStarting(Stage stage) {
System.out.println("Starting "+stage.getName());
}
@Override
public void stageComplete(Stage stage) {
}
@Override
public void stageFailed(Stage stage) {
JOptionPane.showMessageDialog(streamFrame, "Starting "+stage.getName()+" failed", "Connection Error", JOptionPane.ERROR_MESSAGE);
conn.stop();
}
@Override
public void connectionStarted() {
}
@Override
public void connectionTerminated(Exception e) {
e.printStackTrace();
JOptionPane.showMessageDialog(streamFrame, "The connection failed unexpectedly", "Connection Terminated", JOptionPane.ERROR_MESSAGE);
conn.stop();
}
}

90
limelight-pc/src/com/limelight/gui/MainFrame.java Normal file
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package com.limelight.gui;
import java.awt.BorderLayout;
import java.awt.Container;
import java.awt.Dimension;
import java.awt.event.ActionEvent;
import java.awt.event.ActionListener;
import javax.swing.Box;
import javax.swing.BoxLayout;
import javax.swing.JButton;
import javax.swing.JFrame;
import javax.swing.JPanel;
import javax.swing.JTextField;
import com.limelight.Limelight;
public class MainFrame {
private JTextField hostField;
private JButton pair;
private JButton stream;
public MainFrame() {
}
public void build() {
JFrame limeFrame = new JFrame("Limelight V" + Limelight.VERSION);
limeFrame.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
Container mainPane = limeFrame.getContentPane();
mainPane.setLayout(new BorderLayout());
JPanel centerPane = new JPanel();
centerPane.setLayout(new BoxLayout(centerPane, BoxLayout.Y_AXIS));
hostField = new JTextField();
hostField.setMaximumSize(new Dimension(Integer.MAX_VALUE, 24));
hostField.setToolTipText("Enter host name or IP address");
hostField.setText("GeForce PC host");
stream = new JButton("Start Streaming");
stream.addActionListener(createStreamButtonListener());
stream.setToolTipText("Start the GeForce stream");
pair = new JButton("Pair");
pair.addActionListener(createPairButtonListener());
pair.setToolTipText("Send pair request to GeForce PC");
Box streamBox = Box.createHorizontalBox();
streamBox.add(Box.createHorizontalGlue());
streamBox.add(stream);
streamBox.add(Box.createHorizontalGlue());
Box pairBox = Box.createHorizontalBox();
pairBox.add(Box.createHorizontalGlue());
pairBox.add(pair);
pairBox.add(Box.createHorizontalGlue());
Box contentBox = Box.createVerticalBox();
contentBox.add(Box.createVerticalStrut(20));
contentBox.add(hostField);
contentBox.add(Box.createVerticalStrut(10));
contentBox.add(streamBox);
contentBox.add(Box.createVerticalStrut(10));
contentBox.add(pairBox);
contentBox.add(Box.createVerticalGlue());
centerPane.add(contentBox);
mainPane.add(centerPane, "Center");
limeFrame.setSize(1000, 800);
limeFrame.setVisible(true);
}
private ActionListener createStreamButtonListener() {
return new ActionListener() {
@Override
public void actionPerformed(ActionEvent e) {
Limelight.createInstance(hostField.getText());
}
};
}
private ActionListener createPairButtonListener() {
return null;
}
}

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limelight-pc/src/com/limelight/gui/StreamFrame.java Normal file
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package com.limelight.gui;
import javax.swing.JFrame;
public class StreamFrame extends JFrame {
private static final long serialVersionUID = 1L;
public void build() {
}
}

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limelight-pc/src/com/limelight/input/KeyboardHandler.java Normal file
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package com.limelight.input;
import java.awt.event.KeyEvent;
import java.awt.event.KeyListener;
public class KeyboardHandler implements KeyListener {
@Override
public void keyPressed(KeyEvent event) {
}
@Override
public void keyReleased(KeyEvent event) {
// TODO Auto-generated method stub
}
@Override
public void keyTyped(KeyEvent event) {
}
}

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limelight-pc/src/com/limelight/nvstream/NvApp.java Normal file
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package com.limelight.nvstream;
public class NvApp {
private String appName;
private int appId;
private boolean isRunning;
public void setAppName(String appName) {
this.appName = appName;
}
public void setAppId(String appId) {
this.appId = Integer.parseInt(appId);
}
public void setIsRunning(String isRunning) {
this.isRunning = isRunning.equals("1");
}
public String getAppName() {
return this.appName;
}
public int getAppId() {
return this.appId;
}
public boolean getIsRunning() {
return this.isRunning;
}
}

250
limelight-pc/src/com/limelight/nvstream/NvAudioStream.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.SocketException;
import java.util.LinkedList;
import java.util.concurrent.LinkedBlockingQueue;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.SourceDataLine;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
import com.limelight.nvstream.av.AvRtpPacket;
import com.limelight.nvstream.av.AvShortBufferDescriptor;
import com.limelight.nvstream.av.audio.AvAudioDepacketizer;
import com.limelight.nvstream.av.audio.OpusDecoder;
public class NvAudioStream {
public static final int RTP_PORT = 48000;
public static final int RTCP_PORT = 47999;
private LinkedBlockingQueue<AvRtpPacket> packets = new LinkedBlockingQueue<AvRtpPacket>(100);
private SourceDataLine track;
private DatagramSocket rtp;
private AvAudioDepacketizer depacketizer = new AvAudioDepacketizer();
private LinkedList<Thread> threads = new LinkedList<Thread>();
private boolean aborting = false;
private InetAddress host;
private NvConnectionListener listener;
public NvAudioStream(InetAddress host, NvConnectionListener listener)
{
this.host = host;
this.listener = listener;
}
public void abort()
{
if (aborting) {
return;
}
aborting = true;
for (Thread t : threads) {
t.interrupt();
}
// Close the socket to interrupt the receive thread
if (rtp != null) {
rtp.close();
}
// Wait for threads to terminate
for (Thread t : threads) {
try {
t.join();
} catch (InterruptedException e) { }
}
if (track != null) {
track.close();
}
threads.clear();
}
public void startAudioStream() throws SocketException
{
setupRtpSession();
setupAudio();
startReceiveThread();
startDepacketizerThread();
startDecoderThread();
startUdpPingThread();
}
private void setupRtpSession() throws SocketException
{
rtp = new DatagramSocket(RTP_PORT);
}
private void setupAudio()
{
int channelConfig;
int err;
err = OpusDecoder.init();
if (err != 0) {
throw new IllegalStateException("Opus decoder failed to initialize");
}
switch (OpusDecoder.getChannelCount())
{
case 1:
channelConfig = 1;
break;
case 2:
channelConfig = 2;
break;
default:
throw new IllegalStateException("Opus decoder returned unhandled channel count");
}
/*
track = new AudioTrack(AudioManager.STREAM_MUSIC,
OpusDecoder.getSampleRate(),
channelConfig,
AudioFormat.ENCODING_PCM_16BIT,
1024, // 1KB buffer
AudioTrack.MODE_STREAM);
track.play();*/
}
private void startDepacketizerThread()
{
// This thread lessens the work on the receive thread
// so it can spend more time waiting for data
Thread t = new Thread() {
@Override
public void run() {
while (!isInterrupted())
{
AvRtpPacket packet;
try {
packet = packets.take();
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
depacketizer.decodeInputData(packet);
}
}
};
threads.add(t);
t.setName("Audio - Depacketizer");
t.start();
}
private void startDecoderThread()
{
// Decoder thread
Thread t = new Thread() {
@Override
public void run() {
while (!isInterrupted())
{
AvShortBufferDescriptor samples;
try {
samples = depacketizer.getNextDecodedData();
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
//track.write(samples.data, samples.offset, samples.length);
}
}
};
threads.add(t);
t.setName("Audio - Player");
t.start();
}
private void startReceiveThread()
{
// Receive thread
Thread t = new Thread() {
@Override
public void run() {
AvByteBufferDescriptor desc = new AvByteBufferDescriptor(new byte[1500], 0, 1500);
DatagramPacket packet = new DatagramPacket(desc.data, desc.length);
while (!isInterrupted())
{
try {
rtp.receive(packet);
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
// Give the packet to the depacketizer thread
desc.length = packet.getLength();
if (packets.offer(new AvRtpPacket(desc))) {
desc.reinitialize(new byte[1500], 0, 1500);
packet.setData(desc.data, desc.offset, desc.length);
}
}
}
};
threads.add(t);
t.setName("Audio - Receive");
t.start();
}
private void startUdpPingThread()
{
// Ping thread
Thread t = new Thread() {
@Override
public void run() {
// PING in ASCII
final byte[] pingPacketData = new byte[] {0x50, 0x49, 0x4E, 0x47};
DatagramPacket pingPacket = new DatagramPacket(pingPacketData, pingPacketData.length);
pingPacket.setSocketAddress(new InetSocketAddress(host, RTP_PORT));
// Send PING every 100 ms
while (!isInterrupted())
{
try {
rtp.send(pingPacket);
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
try {
Thread.sleep(100);
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
}
}
};
threads.add(t);
t.setPriority(Thread.MIN_PRIORITY);
t.setName("Audio - Ping");
t.start();
}
}

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limelight-pc/src/com/limelight/nvstream/NvConnection.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.net.InetAddress;
import java.net.NetworkInterface;
import java.net.SocketException;
import java.net.UnknownHostException;
import java.util.Enumeration;
import java.util.concurrent.ThreadPoolExecutor;
import javax.swing.JFrame;
import javax.swing.JOptionPane;
import javax.swing.SwingUtilities;
import javax.xml.stream.XMLStreamException;
import com.limelight.nvstream.input.NvController;
public class NvConnection {
private String host;
private JFrame parent;
private NvConnectionListener listener;
private int drFlags;
private InetAddress hostAddr;
private NvControl controlStream;
private NvController inputStream;
private NvVideoStream videoStream;
private NvAudioStream audioStream;
private ThreadPoolExecutor threadPool;
public NvConnection(String host, JFrame parent, NvConnectionListener listener) {
this.host = host;
this.parent = parent;
this.listener = listener;
}
public static String getMacAddressString() throws SocketException {
Enumeration<NetworkInterface> ifaceList;
NetworkInterface selectedIface = null;
// First look for a WLAN interface (since those generally aren't removable)
ifaceList = NetworkInterface.getNetworkInterfaces();
while (selectedIface == null && ifaceList.hasMoreElements()) {
NetworkInterface iface = ifaceList.nextElement();
if (iface.getName().startsWith("wlan") &&
iface.getHardwareAddress() != null) {
selectedIface = iface;
}
}
// If we didn't find that, look for an Ethernet interface
ifaceList = NetworkInterface.getNetworkInterfaces();
while (selectedIface == null && ifaceList.hasMoreElements()) {
NetworkInterface iface = ifaceList.nextElement();
if (iface.getName().startsWith("eth") &&
iface.getHardwareAddress() != null) {
selectedIface = iface;
}
}
// Now just find something with a MAC address
ifaceList = NetworkInterface.getNetworkInterfaces();
while (selectedIface == null && ifaceList.hasMoreElements()) {
NetworkInterface iface = ifaceList.nextElement();
if (iface.getHardwareAddress() != null) {
selectedIface = ifaceList.nextElement();
break;
}
}
if (selectedIface == null) {
return null;
}
byte[] macAddress = selectedIface.getHardwareAddress();
if (macAddress != null) {
StringBuilder addrStr = new StringBuilder();
for (int i = 0; i < macAddress.length; i++) {
addrStr.append(String.format("%02x", macAddress[i]));
if (i != macAddress.length - 1) {
addrStr.append(':');
}
}
return addrStr.toString();
}
return null;
}
public void start() {
new Thread(new Runnable() {
@Override
public void run() {
try {
hostAddr = InetAddress.getByName(host);
} catch (UnknownHostException e) {
e.printStackTrace();
displayMessage(e.getMessage());
listener.connectionTerminated(e);
return;
}
establishConnection();
}
}).start();
}
private void establishConnection() {
for (NvConnectionListener.Stage currentStage : NvConnectionListener.Stage.values())
{
boolean success = false;
listener.stageStarting(currentStage);
try {
switch (currentStage)
{
case LAUNCH_APP:
success = startSteamBigPicture();
break;
case HANDSHAKE:
success = NvHandshake.performHandshake(hostAddr);
break;
case CONTROL_START:
success = startControlStream();
break;
case VIDEO_START:
success = startVideoStream();
break;
case AUDIO_START:
success = startAudioStream();
break;
case CONTROL_START2:
controlStream.startJitterPackets();
success = true;
break;
case INPUT_START:
success = startInputConnection();
break;
}
} catch (Exception e) {
e.printStackTrace();
success = false;
}
if (success) {
listener.stageComplete(currentStage);
}
else {
listener.stageFailed(currentStage);
return;
}
}
listener.connectionStarted();
}
private boolean startSteamBigPicture() throws XMLStreamException, IOException
{
System.out.println(hostAddr.toString() + "\t" + getMacAddressString());
NvHTTP h = new NvHTTP(hostAddr.toString(), "");//getMacAddressString());
if (!h.getPairState()) {
displayMessage("Device not paired with computer");
return false;
}
int sessionId = h.getSessionId();
int appId = h.getSteamAppId(sessionId);
h.launchApp(sessionId, appId);
return true;
}
private boolean startControlStream() throws IOException
{
controlStream = new NvControl(hostAddr, listener);
controlStream.initialize();
controlStream.start();
return true;
}
private boolean startVideoStream() throws IOException
{
videoStream = new NvVideoStream(hostAddr, listener, controlStream);
//videoStream.startVideoStream(video, drFlags);
return true;
}
private boolean startAudioStream() throws IOException
{
audioStream = new NvAudioStream(hostAddr, listener);
audioStream.startAudioStream();
return true;
}
private boolean startInputConnection() throws IOException
{
inputStream = new NvController(hostAddr);
inputStream.initialize();
return true;
}
public void stop()
{
threadPool.shutdownNow();
if (videoStream != null) {
videoStream.abort();
}
if (audioStream != null) {
audioStream.abort();
}
if (controlStream != null) {
controlStream.abort();
}
if (inputStream != null) {
inputStream.close();
inputStream = null;
}
}
private void displayMessage(final String text) {
SwingUtilities.invokeLater(new Runnable() {
@Override
public void run() {
JOptionPane.showMessageDialog(parent, text);
}
});
}
}

30
limelight-pc/src/com/limelight/nvstream/NvConnectionListener.java Normal file
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package com.limelight.nvstream;
public interface NvConnectionListener {
public enum Stage {
LAUNCH_APP("app"),
HANDSHAKE("handshake"),
CONTROL_START("control connection"),
VIDEO_START("video stream"),
AUDIO_START("audio stream"),
CONTROL_START2("control connection"),
INPUT_START("input connection");
private String name;
private Stage(String name) {
this.name = name;
}
public String getName() {
return name;
}
};
public void stageStarting(Stage stage);
public void stageComplete(Stage stage);
public void stageFailed(Stage stage);
public void connectionStarted();
public void connectionTerminated(Exception e);
}

483
limelight-pc/src/com/limelight/nvstream/NvControl.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.io.InputStream;
import java.io.OutputStream;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.Socket;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import com.limelight.nvstream.av.ConnectionStatusListener;
public class NvControl implements ConnectionStatusListener {
public static final int PORT = 47995;
public static final int CONTROL_TIMEOUT = 5000;
public static final short PTYPE_HELLO = 0x1204;
public static final short PPAYLEN_HELLO = 0x0004;
public static final byte[] PPAYLOAD_HELLO =
{
(byte)0x00,
(byte)0x05,
(byte)0x00,
(byte)0x00
};
public static final short PTYPE_KEEPALIVE = 0x13ff;
public static final short PPAYLEN_KEEPALIVE = 0x0000;
public static final short PTYPE_HEARTBEAT = 0x1401;
public static final short PPAYLEN_HEARTBEAT = 0x0000;
public static final short PTYPE_1405 = 0x1405;
public static final short PPAYLEN_1405 = 0x0000;
public static final short PTYPE_RESYNC = 0x1404;
public static final short PPAYLEN_RESYNC = 16;
public static final short PTYPE_CONFIG = 0x1205;
public static final short PPAYLEN_CONFIG = 0x0004;
public static final int[] PPAYLOAD_CONFIG =
{
720,
266758,
1,
266762,
30,
70151,
68291329,
1280,
68291584,
1280,
68291840,
15360,
68292096,
25600,
68292352,
2048,
68292608,
1024,
68289024,
262144,
17957632,
302055424,
134217729,
16777490,
70153,
68293120,
768000,
17961216,
303235072,
335609857,
838861842,
352321536,
1006634002,
369098752,
335545362,
385875968,
1042,
402653184,
134218770,
419430400,
167773202,
436207616,
855638290,
266779,
7000,
266780,
2000,
266781,
50,
266782,
3000,
266783,
2,
266794,
5000,
266795,
500,
266784,
75,
266785,
25,
266786,
10,
266787,
60,
266788,
30,
266789,
3,
266790,
1000,
266791,
5000,
266792,
5000,
266793,
5000,
70190,
68301063,
10240,
68301312,
6400,
68301568,
768000,
68299776,
768,
68300032,
2560,
68300544,
0,
34746368,
(int)0xFE000000
};
public static final short PTYPE_JITTER = 0x140c;
public static final short PPAYLEN_JITTER = 0x10;
private int seqNum;
private NvConnectionListener listener;
private InetAddress host;
private Socket s;
private InputStream in;
private OutputStream out;
private Thread heartbeatThread;
private Thread jitterThread;
private boolean aborting = false;
public NvControl(InetAddress host, NvConnectionListener listener)
{
this.listener = listener;
this.host = host;
}
public void initialize() throws IOException
{
s = new Socket();
s.setSoTimeout(CONTROL_TIMEOUT);
s.connect(new InetSocketAddress(host, PORT), CONTROL_TIMEOUT);
in = s.getInputStream();
out = s.getOutputStream();
}
private void sendPacket(NvCtlPacket packet) throws IOException
{
out.write(packet.toWire());
out.flush();
}
private NvControl.NvCtlResponse sendAndGetReply(NvCtlPacket packet) throws IOException
{
sendPacket(packet);
return new NvCtlResponse(in);
}
private void sendJitter() throws IOException
{
ByteBuffer bb = ByteBuffer.allocate(16).order(ByteOrder.LITTLE_ENDIAN);
bb.putInt(0);
bb.putInt(77);
bb.putInt(888);
bb.putInt(seqNum += 2);
sendPacket(new NvCtlPacket(PTYPE_JITTER, PPAYLEN_JITTER, bb.array()));
}
public void abort()
{
if (aborting) {
return;
}
aborting = true;
if (jitterThread != null) {
jitterThread.interrupt();
}
if (heartbeatThread != null) {
heartbeatThread.interrupt();
}
try {
s.close();
} catch (IOException e) {}
}
public void requestResync() throws IOException
{
System.out.println("CTL: Requesting IDR frame");
sendResync();
}
public void start() throws IOException
{
sendHello();
sendConfig();
pingPong();
send1405AndGetResponse();
heartbeatThread = new Thread() {
@Override
public void run() {
while (!isInterrupted())
{
try {
sendHeartbeat();
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
try {
Thread.sleep(3000);
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
}
}
};
heartbeatThread.start();
}
public void startJitterPackets()
{
jitterThread = new Thread() {
@Override
public void run() {
while (!isInterrupted())
{
try {
sendJitter();
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
try {
Thread.sleep(100);
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
}
}
};
jitterThread.start();
}
private NvControl.NvCtlResponse send1405AndGetResponse() throws IOException
{
return sendAndGetReply(new NvCtlPacket(PTYPE_1405, PPAYLEN_1405));
}
private void sendHello() throws IOException
{
sendPacket(new NvCtlPacket(PTYPE_HELLO, PPAYLEN_HELLO, PPAYLOAD_HELLO));
}
private void sendResync() throws IOException
{
ByteBuffer conf = ByteBuffer.wrap(new byte[PPAYLEN_RESYNC]).order(ByteOrder.LITTLE_ENDIAN);
conf.putLong(0);
conf.putLong(0xFFFF);
sendAndGetReply(new NvCtlPacket(PTYPE_RESYNC, PPAYLEN_RESYNC, conf.array()));
}
private void sendConfig() throws IOException
{
ByteBuffer conf = ByteBuffer.wrap(new byte[PPAYLOAD_CONFIG.length * 4 + 3]).order(ByteOrder.LITTLE_ENDIAN);
for (int i : PPAYLOAD_CONFIG)
conf.putInt(i);
conf.putShort((short)0x0013);
conf.put((byte) 0x00);
sendPacket(new NvCtlPacket(PTYPE_CONFIG, PPAYLEN_CONFIG, conf.array()));
}
private void sendHeartbeat() throws IOException
{
sendPacket(new NvCtlPacket(PTYPE_HEARTBEAT, PPAYLEN_HEARTBEAT));
}
private NvControl.NvCtlResponse pingPong() throws IOException
{
sendPacket(new NvCtlPacket(PTYPE_KEEPALIVE, PPAYLEN_KEEPALIVE));
return new NvControl.NvCtlResponse(in);
}
class NvCtlPacket {
public short type;
public short paylen;
public byte[] payload;
public NvCtlPacket(InputStream in) throws IOException
{
byte[] header = new byte[4];
int offset = 0;
do
{
int bytesRead = in.read(header, offset, header.length - offset);
if (bytesRead < 0) {
break;
}
offset += bytesRead;
} while (offset != header.length);
if (offset != header.length) {
throw new IOException("Socket closed prematurely");
}
ByteBuffer bb = ByteBuffer.wrap(header).order(ByteOrder.LITTLE_ENDIAN);
type = bb.getShort();
paylen = bb.getShort();
if (paylen != 0)
{
payload = new byte[paylen];
offset = 0;
do
{
int bytesRead = in.read(payload, offset, payload.length - offset);
if (bytesRead < 0) {
break;
}
offset += bytesRead;
} while (offset != payload.length);
if (offset != payload.length) {
throw new IOException("Socket closed prematurely");
}
}
}
public NvCtlPacket(byte[] payload)
{
ByteBuffer bb = ByteBuffer.wrap(payload).order(ByteOrder.LITTLE_ENDIAN);
type = bb.getShort();
paylen = bb.getShort();
if (bb.hasRemaining())
{
payload = new byte[bb.remaining()];
bb.get(payload);
}
}
public NvCtlPacket(short type, short paylen)
{
this.type = type;
this.paylen = paylen;
}
public NvCtlPacket(short type, short paylen, byte[] payload)
{
this.type = type;
this.paylen = paylen;
this.payload = payload;
}
public short getType()
{
return type;
}
public short getPaylen()
{
return paylen;
}
public void setType(short type)
{
this.type = type;
}
public void setPaylen(short paylen)
{
this.paylen = paylen;
}
public byte[] toWire()
{
ByteBuffer bb = ByteBuffer.allocate(4 + (payload != null ? payload.length : 0)).order(ByteOrder.LITTLE_ENDIAN);
bb.putShort(type);
bb.putShort(paylen);
if (payload != null)
bb.put(payload);
return bb.array();
}
}
class NvCtlResponse extends NvCtlPacket {
public short status;
public NvCtlResponse(InputStream in) throws IOException {
super(in);
}
public NvCtlResponse(short type, short paylen) {
super(type, paylen);
}
public NvCtlResponse(short type, short paylen, byte[] payload) {
super(type, paylen, payload);
}
public NvCtlResponse(byte[] payload) {
super(payload);
}
public void setStatusCode(short status)
{
this.status = status;
}
public short getStatusCode()
{
return status;
}
}
@Override
public void connectionTerminated() {
abort();
}
@Override
public void connectionNeedsResync() {
new Thread(new Runnable() {
@Override
public void run() {
try {
requestResync();
} catch (IOException e1) {
abort();
return;
}
}
}).start();
}
}

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limelight-pc/src/com/limelight/nvstream/NvHTTP.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.io.InputStream;
import java.net.InetAddress;
import java.net.URL;
import java.net.UnknownHostException;
import java.util.LinkedList;
import java.util.Stack;
import javax.xml.stream.XMLInputFactory;
import javax.xml.stream.XMLStreamException;
import javax.xml.stream.XMLStreamReader;
public class NvHTTP {
private String macAddress;
public static final int PORT = 47989;
public String baseUrl;
public NvHTTP(String host, String macAddress) {
this.macAddress = macAddress;
this.baseUrl = "http://" + host + ":" + PORT;
}
private String getXmlString(InputStream in, String tagname)
throws IOException, XMLStreamException {
XMLInputFactory factory = XMLInputFactory.newFactory();
XMLStreamReader xReader = factory.createXMLStreamReader(in);
int eventType = xReader.getEventType();
Stack<String> currentTag = new Stack<String>();
while (eventType != XMLStreamReader.END_DOCUMENT) {
switch (eventType) {
case (XMLStreamReader.START_ELEMENT):
currentTag.push(xReader.getElementText());
break;
case (XMLStreamReader.END_ELEMENT):
currentTag.pop();
break;
case (XMLStreamReader.CHARACTERS):
if (currentTag.peek().equals(tagname)) {
return xReader.getElementText();
}
break;
}
eventType = xReader.next();
}
return null;
}
private InputStream openHttpConnection(String url) throws IOException {
return new URL(url).openConnection().getInputStream();
}
public String getAppVersion() throws XMLStreamException, IOException {
InputStream in = openHttpConnection(baseUrl + "/appversion");
return getXmlString(in, "appversion");
}
public boolean getPairState() throws IOException, XMLStreamException {
InputStream in = openHttpConnection(baseUrl + "/pairstate?mac=" + macAddress);
String paired = getXmlString(in, "paired");
return Integer.valueOf(paired) != 0;
}
public int getSessionId() throws IOException, XMLStreamException {
/* Pass the model (minus spaces) as the device name */
String deviceName = "Unknown";
try
{
InetAddress addr;
addr = InetAddress.getLocalHost();
deviceName = addr.getHostName();
}
catch (UnknownHostException ex)
{
System.out.println("Hostname can not be resolved");
}
InputStream in = openHttpConnection(baseUrl + "/pair?mac=" + macAddress
+ "&devicename=" + deviceName);
String sessionId = getXmlString(in, "sessionid");
return Integer.parseInt(sessionId);
}
public int getSteamAppId(int sessionId) throws IOException,
XMLStreamException {
LinkedList<NvApp> appList = getAppList(sessionId);
for (NvApp app : appList) {
if (app.getAppName().equals("Steam")) {
return app.getAppId();
}
}
return 0;
}
public LinkedList<NvApp> getAppList(int sessionId) throws IOException, XMLStreamException {
InputStream in = openHttpConnection(baseUrl + "/applist?session=" + sessionId);
XMLInputFactory factory = XMLInputFactory.newFactory();
XMLStreamReader xReader = factory.createXMLStreamReader(in);
int eventType = xReader.getEventType();
LinkedList<NvApp> appList = new LinkedList<NvApp>();
Stack<String> currentTag = new Stack<String>();
while (eventType != XMLStreamReader.END_DOCUMENT) {
switch (eventType) {
case (XMLStreamReader.START_ELEMENT):
currentTag.push(xReader.getName().toString());
if (xReader.getName().toString().equals("App")) {
appList.addLast(new NvApp());
}
break;
case (XMLStreamReader.END_DOCUMENT):
currentTag.pop();
break;
case (XMLStreamReader.CHARACTERS):
NvApp app = appList.getLast();
if (currentTag.peek().equals("AppTitle")) {
app.setAppName(xReader.getText());
} else if (currentTag.peek().equals("ID")) {
app.setAppId(xReader.getText());
} else if (currentTag.peek().equals("IsRunning")) {
app.setIsRunning(xReader.getText());
}
break;
}
eventType = xReader.next();
}
return appList;
}
// Returns gameSession XML attribute
public int launchApp(int sessionId, int appId) throws IOException,
XMLStreamException {
InputStream in = openHttpConnection(baseUrl + "/launch?session="
+ sessionId + "&appid=" + appId);
String gameSession = getXmlString(in, "gamesession");
return Integer.parseInt(gameSession);
}
}

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limelight-pc/src/com/limelight/nvstream/NvHandshake.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.io.InputStream;
import java.io.OutputStream;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.Socket;
public class NvHandshake {
public static final int PORT = 47991;
public static final int HANDSHAKE_TIMEOUT = 5000;
public static final byte[] PLATFORM_HELLO =
{
(byte)0x07,
(byte)0x00,
(byte)0x00,
(byte)0x00,
// android in ASCII
(byte)0x61,
(byte)0x6e,
(byte)0x64,
(byte)0x72,
(byte)0x6f,
(byte)0x69,
(byte)0x64,
(byte)0x03,
(byte)0x01,
(byte)0x00,
(byte)0x00
};
public static final byte[] PACKET_2 =
{
(byte)0x01,
(byte)0x03,
(byte)0x02,
(byte)0x00,
(byte)0x08,
(byte)0x00
};
public static final byte[] PACKET_3 =
{
(byte)0x04,
(byte)0x01,
(byte)0x00,
(byte)0x00,
(byte)0x00,
(byte)0x00,
(byte)0x00,
(byte)0x00
};
public static final byte[] PACKET_4 =
{
(byte)0x01,
(byte)0x01,
(byte)0x00,
(byte)0x00
};
private static boolean waitAndDiscardResponse(InputStream in)
{
// Wait for response and discard response
try {
in.read();
// Wait for the full response to come in
Thread.sleep(250);
for (int i = 0; i < in.available(); i++)
in.read();
} catch (IOException e1) {
return false;
} catch (InterruptedException e) {
return false;
}
return true;
}
public static boolean performHandshake(InetAddress host) throws IOException
{
Socket s = new Socket();
s.connect(new InetSocketAddress(host, PORT), HANDSHAKE_TIMEOUT);
s.setSoTimeout(HANDSHAKE_TIMEOUT);
OutputStream out = s.getOutputStream();
InputStream in = s.getInputStream();
// First packet
out.write(PLATFORM_HELLO);
out.flush();
if (!waitAndDiscardResponse(in)) {
s.close();
return false;
}
// Second packet
out.write(PACKET_2);
out.flush();
if (!waitAndDiscardResponse(in)) {
s.close();
return false;
}
// Third packet
out.write(PACKET_3);
out.flush();
if (!waitAndDiscardResponse(in)) {
s.close();
return false;
}
// Fourth packet
out.write(PACKET_4);
out.flush();
// Done
s.close();
return true;
}
}

307
limelight-pc/src/com/limelight/nvstream/NvVideoStream.java Normal file
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package com.limelight.nvstream;
import java.io.IOException;
import java.io.InputStream;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.Socket;
import java.net.SocketException;
import java.util.LinkedList;
import java.util.concurrent.LinkedBlockingQueue;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
import com.limelight.nvstream.av.AvDecodeUnit;
import com.limelight.nvstream.av.AvRtpPacket;
import com.limelight.nvstream.av.ConnectionStatusListener;
import com.limelight.nvstream.av.video.AvVideoDepacketizer;
import com.limelight.nvstream.av.video.AvVideoPacket;
import com.limelight.nvstream.av.video.CpuDecoderRenderer;
import com.limelight.nvstream.av.video.DecoderRenderer;
public class NvVideoStream {
public static final int RTP_PORT = 47998;
public static final int RTCP_PORT = 47999;
public static final int FIRST_FRAME_PORT = 47996;
public static final int FIRST_FRAME_TIMEOUT = 5000;
private LinkedBlockingQueue<AvRtpPacket> packets = new LinkedBlockingQueue<AvRtpPacket>(100);
private InetAddress host;
private DatagramSocket rtp;
private Socket firstFrameSocket;
private LinkedList<Thread> threads = new LinkedList<Thread>();
private NvConnectionListener listener;
private AvVideoDepacketizer depacketizer;
private DecoderRenderer decrend;
private boolean startedRendering;
private boolean aborting = false;
public NvVideoStream(InetAddress host, NvConnectionListener listener, ConnectionStatusListener avConnListener)
{
this.host = host;
this.listener = listener;
this.depacketizer = new AvVideoDepacketizer(avConnListener);
}
public void abort()
{
if (aborting) {
return;
}
aborting = true;
// Interrupt threads
for (Thread t : threads) {
t.interrupt();
}
// Close the socket to interrupt the receive thread
if (rtp != null) {
rtp.close();
}
if (firstFrameSocket != null) {
try {
firstFrameSocket.close();
} catch (IOException e) {}
}
// Wait for threads to terminate
for (Thread t : threads) {
try {
t.join();
} catch (InterruptedException e) { }
}
if (startedRendering) {
decrend.stop();
}
if (decrend != null) {
decrend.release();
}
threads.clear();
}
private void readFirstFrame() throws IOException
{
byte[] firstFrame = new byte[1500];
firstFrameSocket = new Socket();
firstFrameSocket.setSoTimeout(FIRST_FRAME_TIMEOUT);
try {
firstFrameSocket.connect(new InetSocketAddress(host, FIRST_FRAME_PORT), FIRST_FRAME_TIMEOUT);
InputStream firstFrameStream = firstFrameSocket.getInputStream();
int offset = 0;
for (;;)
{
int bytesRead = firstFrameStream.read(firstFrame, offset, firstFrame.length-offset);
if (bytesRead == -1)
break;
offset += bytesRead;
}
depacketizer.addInputData(new AvVideoPacket(new AvByteBufferDescriptor(firstFrame, 0, offset)));
} finally {
firstFrameSocket.close();
firstFrameSocket = null;
}
}
public void setupRtpSession() throws SocketException
{
rtp = new DatagramSocket(RTP_PORT);
}
/*
public void setupDecoderRenderer(SurfaceHolder renderTarget, int drFlags) {
if (Build.HARDWARE.equals("goldfish")) {
// Emulator - don't render video (it's slow!)
decrend = null;
}
else if (MediaCodecDecoderRenderer.findSafeDecoder() != null) {
// Hardware decoding
decrend = new MediaCodecDecoderRenderer();
}
else {
// Software decoding
decrend = new CpuDecoderRenderer();
}
if (decrend != null) {
decrend.setup(1280, 720, renderTarget, drFlags);
}
}*/
/*
public void startVideoStream(final SurfaceHolder surface, int drFlags) throws IOException
{
// Setup the decoder and renderer
setupDecoderRenderer(surface, drFlags);
// Open RTP sockets and start session
setupRtpSession();
// Start pinging before reading the first frame
// so Shield Proxy knows we're here and sends us
// the reference frame
startUdpPingThread();
// Read the first frame to start the UDP video stream
// This MUST be called before the normal UDP receive thread
// starts in order to avoid state corruption caused by two
// threads simultaneously adding input data.
readFirstFrame();
if (decrend != null) {
// Start the receive thread early to avoid missing
// early packets
startReceiveThread();
// Start the depacketizer thread to deal with the RTP data
startDepacketizerThread();
// Start decoding the data we're receiving
startDecoderThread();
// Start the renderer
decrend.start();
startedRendering = true;
}
}
*/
private void startDecoderThread()
{
Thread t = new Thread() {
@Override
public void run() {
// Read the decode units generated from the RTP stream
while (!isInterrupted())
{
AvDecodeUnit du;
try {
du = depacketizer.getNextDecodeUnit();
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
decrend.submitDecodeUnit(du);
}
}
};
threads.add(t);
t.setName("Video - Decoder");
t.setPriority(Thread.MAX_PRIORITY);
t.start();
}
private void startDepacketizerThread()
{
// This thread lessens the work on the receive thread
// so it can spend more time waiting for data
Thread t = new Thread() {
@Override
public void run() {
while (!isInterrupted())
{
AvRtpPacket packet;
try {
packet = packets.take();
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
// !!! We no longer own the data buffer at this point !!!
depacketizer.addInputData(packet);
}
}
};
threads.add(t);
t.setName("Video - Depacketizer");
t.start();
}
private void startReceiveThread()
{
// Receive thread
Thread t = new Thread() {
@Override
public void run() {
AvByteBufferDescriptor desc = new AvByteBufferDescriptor(new byte[1500], 0, 1500);
DatagramPacket packet = new DatagramPacket(desc.data, desc.length);
while (!isInterrupted())
{
try {
rtp.receive(packet);
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
// Give the packet to the depacketizer thread
desc.length = packet.getLength();
if (packets.offer(new AvRtpPacket(desc))) {
desc.reinitialize(new byte[1500], 0, 1500);
packet.setData(desc.data, desc.offset, desc.length);
}
}
}
};
threads.add(t);
t.setName("Video - Receive");
t.start();
}
private void startUdpPingThread()
{
// Ping thread
Thread t = new Thread() {
@Override
public void run() {
// PING in ASCII
final byte[] pingPacketData = new byte[] {0x50, 0x49, 0x4E, 0x47};
DatagramPacket pingPacket = new DatagramPacket(pingPacketData, pingPacketData.length);
pingPacket.setSocketAddress(new InetSocketAddress(host, RTP_PORT));
// Send PING every 100 ms
while (!isInterrupted())
{
try {
rtp.send(pingPacket);
} catch (IOException e) {
listener.connectionTerminated(e);
return;
}
try {
Thread.sleep(100);
} catch (InterruptedException e) {
listener.connectionTerminated(e);
return;
}
}
}
};
threads.add(t);
t.setName("Video - Ping");
t.setPriority(Thread.MIN_PRIORITY);
t.start();
}
}

46
limelight-pc/src/com/limelight/nvstream/av/AvByteBufferDescriptor.java Normal file
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package com.limelight.nvstream.av;
public class AvByteBufferDescriptor {
public byte[] data;
public int offset;
public int length;
public AvByteBufferDescriptor(byte[] data, int offset, int length)
{
this.data = data;
this.offset = offset;
this.length = length;
}
public AvByteBufferDescriptor(AvByteBufferDescriptor desc)
{
this.data = desc.data;
this.offset = desc.offset;
this.length = desc.length;
}
public void reinitialize(byte[] data, int offset, int length)
{
this.data = data;
this.offset = offset;
this.length = length;
}
public void print()
{
print(offset, length);
}
public void print(int length)
{
print(this.offset, length);
}
public void print(int offset, int length)
{
for (int i = offset; i < offset+length; i++) {
System.out.printf("%d: %02x \n", i, data[i]);
}
System.out.println();
}
}

42
limelight-pc/src/com/limelight/nvstream/av/AvDecodeUnit.java Normal file
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package com.limelight.nvstream.av;
import java.util.List;
public class AvDecodeUnit {
public static final int TYPE_UNKNOWN = 0;
public static final int TYPE_H264 = 1;
public static final int TYPE_OPUS = 2;
private int type;
private List<AvByteBufferDescriptor> bufferList;
private int dataLength;
private int flags;
public AvDecodeUnit(int type, List<AvByteBufferDescriptor> bufferList, int dataLength, int flags)
{
this.type = type;
this.bufferList = bufferList;
this.dataLength = dataLength;
this.flags = flags;
}
public int getType()
{
return type;
}
public int getFlags()
{
return flags;
}
public List<AvByteBufferDescriptor> getBufferList()
{
return bufferList;
}
public int getDataLength()
{
return dataLength;
}
}

46
limelight-pc/src/com/limelight/nvstream/av/AvRtpPacket.java Normal file
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package com.limelight.nvstream.av;
import java.nio.ByteBuffer;
public class AvRtpPacket {
private byte packetType;
private short seqNum;
private AvByteBufferDescriptor buffer;
public AvRtpPacket(AvByteBufferDescriptor buffer)
{
this.buffer = new AvByteBufferDescriptor(buffer);
ByteBuffer bb = ByteBuffer.wrap(buffer.data, buffer.offset, buffer.length);
// Discard the first byte
bb.position(bb.position()+1);
// Get the packet type
packetType = bb.get();
// Get the sequence number
seqNum = bb.getShort();
}
public byte getPacketType()
{
return packetType;
}
public short getSequenceNumber()
{
return seqNum;
}
public byte[] getBackingBuffer()
{
return buffer.data;
}
public AvByteBufferDescriptor getNewPayloadDescriptor()
{
return new AvByteBufferDescriptor(buffer.data, buffer.offset+12, buffer.length-12);
}
}

28
limelight-pc/src/com/limelight/nvstream/av/AvShortBufferDescriptor.java Normal file
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package com.limelight.nvstream.av;
public class AvShortBufferDescriptor {
public short[] data;
public int offset;
public int length;
public AvShortBufferDescriptor(short[] data, int offset, int length)
{
this.data = data;
this.offset = offset;
this.length = length;
}
public AvShortBufferDescriptor(AvShortBufferDescriptor desc)
{
this.data = desc.data;
this.offset = desc.offset;
this.length = desc.length;
}
public void reinitialize(short[] data, int offset, int length)
{
this.data = data;
this.offset = offset;
this.length = length;
}
}

7
limelight-pc/src/com/limelight/nvstream/av/ConnectionStatusListener.java Normal file
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package com.limelight.nvstream.av;
public interface ConnectionStatusListener {
public void connectionTerminated();
public void connectionNeedsResync();
}

65
limelight-pc/src/com/limelight/nvstream/av/audio/AvAudioDepacketizer.java Normal file
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package com.limelight.nvstream.av.audio;
import java.util.concurrent.LinkedBlockingQueue;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
import com.limelight.nvstream.av.AvRtpPacket;
import com.limelight.nvstream.av.AvShortBufferDescriptor;
public class AvAudioDepacketizer {
private static final int DU_LIMIT = 15;
private LinkedBlockingQueue<AvShortBufferDescriptor> decodedUnits =
new LinkedBlockingQueue<AvShortBufferDescriptor>(DU_LIMIT);
// Sequencing state
private short lastSequenceNumber;
private void decodeData(byte[] data, int off, int len)
{
// Submit this data to the decoder
short[] pcmData = new short[OpusDecoder.getMaxOutputShorts()];
int decodeLen = OpusDecoder.decode(data, off, len, pcmData);
if (decodeLen > 0) {
// Return value of decode is frames decoded per channel
decodeLen *= OpusDecoder.getChannelCount();
// Put it on the decoded queue
if (!decodedUnits.offer(new AvShortBufferDescriptor(pcmData, 0, decodeLen))) {
// Clear out the queue
decodedUnits.clear();
}
}
}
public void decodeInputData(AvRtpPacket packet)
{
short seq = packet.getSequenceNumber();
if (packet.getPacketType() != 97) {
// Only type 97 is audio
return;
}
// Toss out the current NAL if we receive a packet that is
// out of sequence
if (lastSequenceNumber != 0 &&
(short)(lastSequenceNumber + 1) != seq)
{
System.out.println("Received OOS audio data (expected "+(lastSequenceNumber + 1)+", got "+seq+")");
decodeData(null, 0, 0);
}
lastSequenceNumber = seq;
// This is all the depacketizing we need to do
AvByteBufferDescriptor rtpPayload = packet.getNewPayloadDescriptor();
decodeData(rtpPayload.data, rtpPayload.offset, rtpPayload.length);
}
public AvShortBufferDescriptor getNextDecodedData() throws InterruptedException
{
return decodedUnits.take();
}
}

14
limelight-pc/src/com/limelight/nvstream/av/audio/OpusDecoder.java Normal file
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package com.limelight.nvstream.av.audio;
public class OpusDecoder {
static {
System.loadLibrary("nv_opus_dec");
}
public static native int init();
public static native void destroy();
public static native int getChannelCount();
public static native int getMaxOutputShorts();
public static native int getSampleRate();
public static native int decode(byte[] indata, int inoff, int inlen, short[] outpcmdata);
}

313
limelight-pc/src/com/limelight/nvstream/av/video/AvVideoDepacketizer.java Normal file
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package com.limelight.nvstream.av.video;
import java.util.LinkedList;
import java.util.concurrent.LinkedBlockingQueue;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
import com.limelight.nvstream.av.AvDecodeUnit;
import com.limelight.nvstream.av.AvRtpPacket;
import com.limelight.nvstream.av.ConnectionStatusListener;
public class AvVideoDepacketizer {
// Current NAL state
private LinkedList<AvByteBufferDescriptor> avcNalDataChain = null;
private int avcNalDataLength = 0;
private int currentlyDecoding;
// Cached buffer descriptor to save on allocations
// Only safe to use in decode thread!!!!
private AvByteBufferDescriptor cachedDesc;
// Sequencing state
private short lastSequenceNumber;
private ConnectionStatusListener controlListener;
private static final int DU_LIMIT = 15;
private LinkedBlockingQueue<AvDecodeUnit> decodedUnits = new LinkedBlockingQueue<AvDecodeUnit>(DU_LIMIT);
public AvVideoDepacketizer(ConnectionStatusListener controlListener)
{
this.controlListener = controlListener;
this.cachedDesc = new AvByteBufferDescriptor(null, 0, 0);
}
private void clearAvcNalState()
{
avcNalDataChain = null;
avcNalDataLength = 0;
}
private void reassembleAvcNal()
{
// This is the start of a new NAL
if (avcNalDataChain != null && avcNalDataLength != 0)
{
int flags = 0;
// Check if this is a special NAL unit
AvByteBufferDescriptor header = avcNalDataChain.getFirst();
if (NAL.getSpecialSequenceDescriptor(header, cachedDesc))
{
// The next byte after the special sequence is the NAL header
byte nalHeader = cachedDesc.data[cachedDesc.offset+cachedDesc.length];
switch (nalHeader)
{
// SPS and PPS
case 0x67:
case 0x68:
System.out.println("Codec config");
//flags |= MediaCodec.BUFFER_FLAG_CODEC_CONFIG;
break;
// IDR
case 0x65:
System.out.println("Reference frame");
//flags |= MediaCodec.BUFFER_FLAG_SYNC_FRAME;
break;
// non-IDR frame
case 0x61:
break;
// Unknown type
default:
System.out.printf("Unknown NAL header: %02x %02x %02x %02x %02x\n",
header.data[header.offset], header.data[header.offset+1],
header.data[header.offset+2], header.data[header.offset+3],
header.data[header.offset+4]);
break;
}
}
else
{
System.out.printf("Invalid NAL: %02x %02x %02x %02x %02x\n",
header.data[header.offset], header.data[header.offset+1],
header.data[header.offset+2], header.data[header.offset+3],
header.data[header.offset+4]);
}
// Construct the H264 decode unit
AvDecodeUnit du = new AvDecodeUnit(AvDecodeUnit.TYPE_H264, avcNalDataChain, avcNalDataLength, flags);
if (!decodedUnits.offer(du)) {
// We need a new IDR frame since we're discarding data now
decodedUnits.clear();
controlListener.connectionNeedsResync();
}
// Clear old state
avcNalDataChain = null;
avcNalDataLength = 0;
}
}
public void addInputData(AvVideoPacket packet)
{
AvByteBufferDescriptor location = packet.getNewPayloadDescriptor();
while (location.length != 0)
{
// Remember the start of the NAL data in this packet
int start = location.offset;
// Check for a special sequence
if (NAL.getSpecialSequenceDescriptor(location, cachedDesc))
{
if (NAL.isAvcStartSequence(cachedDesc))
{
// We're decoding H264 now
currentlyDecoding = AvDecodeUnit.TYPE_H264;
// Check if it's the end of the last frame
if (NAL.isAvcFrameStart(cachedDesc))
{
// Reassemble any pending AVC NAL
reassembleAvcNal();
// Setup state for the new NAL
avcNalDataChain = new LinkedList<AvByteBufferDescriptor>();
avcNalDataLength = 0;
}
// Skip the start sequence
location.length -= cachedDesc.length;
location.offset += cachedDesc.length;
}
else
{
// Check if this is padding after a full AVC frame
if (currentlyDecoding == AvDecodeUnit.TYPE_H264 &&
NAL.isPadding(cachedDesc)) {
// The decode unit is complete
reassembleAvcNal();
}
// Not decoding AVC
currentlyDecoding = AvDecodeUnit.TYPE_UNKNOWN;
// Just skip this byte
location.length--;
location.offset++;
}
}
// Move to the next special sequence
while (location.length != 0)
{
// Catch the easy case first where byte 0 != 0x00
if (location.data[location.offset] == 0x00)
{
// Check if this should end the current NAL
if (NAL.getSpecialSequenceDescriptor(location, cachedDesc))
{
// Only stop if we're decoding something or this
// isn't padding
if (currentlyDecoding != AvDecodeUnit.TYPE_UNKNOWN ||
!NAL.isPadding(cachedDesc))
{
break;
}
}
}
// This byte is part of the NAL data
location.offset++;
location.length--;
}
if (currentlyDecoding == AvDecodeUnit.TYPE_H264 && avcNalDataChain != null)
{
AvByteBufferDescriptor data = new AvByteBufferDescriptor(location.data, start, location.offset-start);
// Add a buffer descriptor describing the NAL data in this packet
avcNalDataChain.add(data);
avcNalDataLength += location.offset-start;
}
}
}
public void addInputData(AvRtpPacket packet)
{
short seq = packet.getSequenceNumber();
// Toss out the current NAL if we receive a packet that is
// out of sequence
if (lastSequenceNumber != 0 &&
(short)(lastSequenceNumber + 1) != seq)
{
System.out.println("Received OOS video data (expected "+(lastSequenceNumber + 1)+", got "+seq+")");
// Reset the depacketizer state
currentlyDecoding = AvDecodeUnit.TYPE_UNKNOWN;
clearAvcNalState();
// Request an IDR frame
controlListener.connectionNeedsResync();
}
lastSequenceNumber = seq;
// Pass the payload to the non-sequencing parser
AvByteBufferDescriptor rtpPayload = packet.getNewPayloadDescriptor();
addInputData(new AvVideoPacket(rtpPayload));
}
public AvDecodeUnit getNextDecodeUnit() throws InterruptedException
{
return decodedUnits.take();
}
}
class NAL {
// This assumes that the buffer passed in is already a special sequence
public static boolean isAvcStartSequence(AvByteBufferDescriptor specialSeq)
{
// The start sequence is 00 00 01 or 00 00 00 01
return (specialSeq.data[specialSeq.offset+specialSeq.length-1] == 0x01);
}
// This assumes that the buffer passed in is already a special sequence
public static boolean isPadding(AvByteBufferDescriptor specialSeq)
{
// The padding sequence is 00 00 00
return (specialSeq.data[specialSeq.offset+specialSeq.length-1] == 0x00);
}
// This assumes that the buffer passed in is already a special sequence
public static boolean isAvcFrameStart(AvByteBufferDescriptor specialSeq)
{
if (specialSeq.length != 4)
return false;
// The frame start sequence is 00 00 00 01
return (specialSeq.data[specialSeq.offset+specialSeq.length-1] == 0x01);
}
// Returns a buffer descriptor describing the start sequence
public static boolean getSpecialSequenceDescriptor(AvByteBufferDescriptor buffer, AvByteBufferDescriptor outputDesc)
{
// NAL start sequence is 00 00 00 01 or 00 00 01
if (buffer.length < 3)
return false;
// 00 00 is magic
if (buffer.data[buffer.offset] == 0x00 &&
buffer.data[buffer.offset+1] == 0x00)
{
// Another 00 could be the end of the special sequence
// 00 00 00 or the middle of 00 00 00 01
if (buffer.data[buffer.offset+2] == 0x00)
{
if (buffer.length >= 4 &&
buffer.data[buffer.offset+3] == 0x01)
{
// It's the AVC start sequence 00 00 00 01
outputDesc.reinitialize(buffer.data, buffer.offset, 4);
}
else
{
// It's 00 00 00
outputDesc.reinitialize(buffer.data, buffer.offset, 3);
}
return true;
}
else if (buffer.data[buffer.offset+2] == 0x01 ||
buffer.data[buffer.offset+2] == 0x02)
{
// These are easy: 00 00 01 or 00 00 02
outputDesc.reinitialize(buffer.data, buffer.offset, 3);
return true;
}
else if (buffer.data[buffer.offset+2] == 0x03)
{
// 00 00 03 is special because it's a subsequence of the
// NAL wrapping substitute for 00 00 00, 00 00 01, 00 00 02,
// or 00 00 03 in the RBSP sequence. We need to check the next
// byte to see whether it's 00, 01, 02, or 03 (a valid RBSP substitution)
// or whether it's something else
if (buffer.length < 4)
return false;
if (buffer.data[buffer.offset+3] >= 0x00 &&
buffer.data[buffer.offset+3] <= 0x03)
{
// It's not really a special sequence after all
return false;
}
else
{
// It's not a standard replacement so it's a special sequence
outputDesc.reinitialize(buffer.data, buffer.offset, 3);
return true;
}
}
}
return false;
}
}

17
limelight-pc/src/com/limelight/nvstream/av/video/AvVideoPacket.java Normal file
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package com.limelight.nvstream.av.video;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
public class AvVideoPacket {
private AvByteBufferDescriptor buffer;
public AvVideoPacket(AvByteBufferDescriptor rtpPayload)
{
buffer = new AvByteBufferDescriptor(rtpPayload);
}
public AvByteBufferDescriptor getNewPayloadDescriptor()
{
return new AvByteBufferDescriptor(buffer.data, buffer.offset+56, buffer.length-56);
}
}

33
limelight-pc/src/com/limelight/nvstream/av/video/AvcDecoder.java Normal file
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package com.limelight.nvstream.av.video;
public class AvcDecoder {
static {
// FFMPEG dependencies
System.loadLibrary("avutil-52");
System.loadLibrary("swresample-0");
System.loadLibrary("swscale-2");
System.loadLibrary("avcodec-55");
System.loadLibrary("avformat-55");
System.loadLibrary("avfilter-3");
System.loadLibrary("nv_avc_dec");
}
/** Disables the deblocking filter at the cost of image quality */
public static final int DISABLE_LOOP_FILTER = 0x1;
/** Uses the low latency decode flag (disables multithreading) */
public static final int LOW_LATENCY_DECODE = 0x2;
/** Threads process each slice, rather than each frame */
public static final int SLICE_THREADING = 0x4;
/** Uses nonstandard speedup tricks */
public static final int FAST_DECODE = 0x8;
/** Uses bilinear filtering instead of bicubic */
public static final int BILINEAR_FILTERING = 0x10;
/** Uses a faster bilinear filtering with lower image quality */
public static final int FAST_BILINEAR_FILTERING = 0x20;
public static native int init(int width, int height, int perflvl, int threadcount);
public static native void destroy();
//public static native void redraw(Surface surface);
public static native int decode(byte[] indata, int inoff, int inlen);
}

203
limelight-pc/src/com/limelight/nvstream/av/video/CpuDecoderRenderer.java Normal file
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package com.limelight.nvstream.av.video;
import java.io.BufferedReader;
import java.io.File;
import java.io.FileReader;
import java.io.IOException;
import java.nio.ByteBuffer;
import com.limelight.nvstream.av.AvByteBufferDescriptor;
import com.limelight.nvstream.av.AvDecodeUnit;
public class CpuDecoderRenderer/* implements DecoderRenderer */{
private ByteBuffer decoderBuffer;
private Thread rendererThread;
private int targetFps;
// Only sleep if the difference is above this value
private static final int WAIT_CEILING_MS = 8;
private static final int LOW_PERF = 1;
private static final int MED_PERF = 2;
private static final int HIGH_PERF = 3;
private int cpuCount = Runtime.getRuntime().availableProcessors();
private int findOptimalPerformanceLevel() {
StringBuilder cpuInfo = new StringBuilder();
BufferedReader br = null;
try {
br = new BufferedReader(new FileReader(new File("/proc/cpuinfo")));
for (;;) {
int ch = br.read();
if (ch == -1)
break;
cpuInfo.append((char)ch);
}
// Here we're doing very simple heuristics based on CPU model
String cpuInfoStr = cpuInfo.toString();
// We order them from greatest to least for proper detection
// of devices with multiple sets of cores (like Exynos 5 Octa)
// TODO Make this better
if (cpuInfoStr.contains("0xc0f")) {
// Cortex-A15
return MED_PERF;
}
else if (cpuInfoStr.contains("0xc09")) {
// Cortex-A9
return LOW_PERF;
}
else if (cpuInfoStr.contains("0xc07")) {
// Cortex-A7
return LOW_PERF;
}
else {
// Didn't have anything we're looking for
return MED_PERF;
}
} catch (IOException e) {
} finally {
if (br != null) {
try {
br.close();
} catch (IOException e) {}
}
}
// Couldn't read cpuinfo, so assume medium
return MED_PERF;
}
/*@Override
public void setup(int width, int height, SurfaceHolder renderTarget, int drFlags) {
this.renderTarget = renderTarget.getSurface();
this.targetFps = 30;
int perfLevel = findOptimalPerformanceLevel();
int threadCount;
int avcFlags = 0;
switch (perfLevel) {
case HIGH_PERF:
// Single threaded low latency decode is ideal but hard to acheive
avcFlags = AvcDecoder.LOW_LATENCY_DECODE;
threadCount = 1;
break;
case LOW_PERF:
// Disable the loop filter for performance reasons
avcFlags = AvcDecoder.DISABLE_LOOP_FILTER |
AvcDecoder.FAST_BILINEAR_FILTERING |
AvcDecoder.FAST_DECODE;
// Use plenty of threads to try to utilize the CPU as best we can
threadCount = cpuCount - 1;
break;
default:
case MED_PERF:
avcFlags = AvcDecoder.BILINEAR_FILTERING |
AvcDecoder.FAST_DECODE;
// Only use 2 threads to minimize frame processing latency
threadCount = 2;
break;
}
// If the user wants quality, we'll remove the low IQ flags
if ((drFlags & DecoderRenderer.FLAG_PREFER_QUALITY) != 0) {
// Make sure the loop filter is enabled
avcFlags &= ~AvcDecoder.DISABLE_LOOP_FILTER;
// Disable the non-compliant speed optimizations
avcFlags &= ~AvcDecoder.FAST_DECODE;
System.out.println("Using high quality decoding");
}
int err = AvcDecoder.init(width, height, avcFlags, threadCount);
if (err != 0) {
throw new IllegalStateException("AVC decoder initialization failure: "+err);
}
decoderBuffer = ByteBuffer.allocate(92*1024);
System.out.println("Using software decoding (performance level: "+perfLevel+")");
}
*/
//@Override
public void start() {
rendererThread = new Thread() {
@Override
public void run() {
long nextFrameTime = System.currentTimeMillis();
while (!isInterrupted())
{
long diff = nextFrameTime - System.currentTimeMillis();
if (diff > WAIT_CEILING_MS) {
try {
Thread.sleep(diff);
} catch (InterruptedException e) {
return;
}
}
nextFrameTime = computePresentationTimeMs(targetFps);
// AvcDecoder.redraw(renderTarget);
}
}
};
rendererThread.setName("Video - Renderer (CPU)");
rendererThread.start();
}
private long computePresentationTimeMs(int frameRate) {
return System.currentTimeMillis() + (1000 / frameRate);
}
//@Override
public void stop() {
rendererThread.interrupt();
try {
rendererThread.join();
} catch (InterruptedException e) { }
}
//@Override
public void release() {
AvcDecoder.destroy();
}
//@Override
public boolean submitDecodeUnit(AvDecodeUnit decodeUnit) {
byte[] data;
// Use the reserved decoder buffer if this decode unit will fit
if (decodeUnit.getDataLength() <= decoderBuffer.limit()) {
decoderBuffer.clear();
for (AvByteBufferDescriptor bbd : decodeUnit.getBufferList()) {
decoderBuffer.put(bbd.data, bbd.offset, bbd.length);
}
data = decoderBuffer.array();
}
else {
data = new byte[decodeUnit.getDataLength()];
int offset = 0;
for (AvByteBufferDescriptor bbd : decodeUnit.getBufferList()) {
System.arraycopy(bbd.data, bbd.offset, data, offset, bbd.length);
offset += bbd.length;
}
}
return (AvcDecoder.decode(data, 0, decodeUnit.getDataLength()) == 0);
}
}

17
limelight-pc/src/com/limelight/nvstream/av/video/DecoderRenderer.java Normal file
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package com.limelight.nvstream.av.video;
import com.limelight.nvstream.av.AvDecodeUnit;
public interface DecoderRenderer {
public static int FLAG_PREFER_QUALITY = 0x1;
public void setup(int width, int height, int drFlags);
public void start();
public void stop();
public void release();
public boolean submitDecodeUnit(AvDecodeUnit decodeUnit);
}

65
limelight-pc/src/com/limelight/nvstream/input/NvController.java Normal file
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package com.limelight.nvstream.input;
import java.io.IOException;
import java.io.OutputStream;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.net.Socket;
public class NvController {
public final static int PORT = 35043;
public final static int CONTROLLER_TIMEOUT = 3000;
private InetAddress host;
private Socket s;
private OutputStream out;
public NvController(InetAddress host)
{
this.host = host;
}
public void initialize() throws IOException
{
s = new Socket();
s.connect(new InetSocketAddress(host, PORT), CONTROLLER_TIMEOUT);
s.setTcpNoDelay(true);
out = s.getOutputStream();
}
public void close()
{
try {
s.close();
} catch (IOException e) {}
}
public void sendControllerInput(short buttonFlags, byte leftTrigger, byte rightTrigger,
short leftStickX, short leftStickY, short rightStickX, short rightStickY) throws IOException
{
out.write(new NvControllerPacket(buttonFlags, leftTrigger,
rightTrigger, leftStickX, leftStickY,
rightStickX, rightStickY).toWire());
out.flush();
}
public void sendMouseButtonDown() throws IOException
{
out.write(new NvMouseButtonPacket(true).toWire());
out.flush();
}
public void sendMouseButtonUp() throws IOException
{
out.write(new NvMouseButtonPacket(false).toWire());
out.flush();
}
public void sendMouseMove(short deltaX, short deltaY) throws IOException
{
out.write(new NvMouseMovePacket(deltaX, deltaY).toWire());
out.flush();
}
}

89
limelight-pc/src/com/limelight/nvstream/input/NvControllerPacket.java Normal file
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package com.limelight.nvstream.input;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
public class NvControllerPacket extends NvInputPacket {
public static final byte[] HEADER =
{
0x0A,
0x00,
0x00,
0x00,
0x00,
0x14
};
public static final byte[] TAIL =
{
(byte)0x9C,
0x00,
0x00,
0x00,
0x55,
0x00
};
public static final int PACKET_TYPE = 0x18;
public static final short A_FLAG = 0x1000;
public static final short B_FLAG = 0x2000;
public static final short X_FLAG = 0x4000;
public static final short Y_FLAG = (short)0x8000;
public static final short UP_FLAG = 0x0001;
public static final short DOWN_FLAG = 0x0002;
public static final short LEFT_FLAG = 0x0004;
public static final short RIGHT_FLAG = 0x0008;
public static final short LB_FLAG = 0x0100;
public static final short RB_FLAG = 0x0200;
public static final short PLAY_FLAG = 0x0010;
public static final short BACK_FLAG = 0x0020;
public static final short LS_CLK_FLAG = 0x0040;
public static final short RS_CLK_FLAG = 0x0080;
public static final short SPECIAL_BUTTON_FLAG = 0x0400;
public static final short PAYLOAD_LENGTH = 24;
public static final short PACKET_LENGTH = PAYLOAD_LENGTH +
NvInputPacket.HEADER_LENGTH;
private short buttonFlags;
private byte leftTrigger;
private byte rightTrigger;
private short leftStickX;
private short leftStickY;
private short rightStickX;
private short rightStickY;
public NvControllerPacket(short buttonFlags, byte leftTrigger, byte rightTrigger,
short leftStickX, short leftStickY,
short rightStickX, short rightStickY)
{
super(PACKET_TYPE);
this.buttonFlags = buttonFlags;
this.leftTrigger = leftTrigger;
this.rightTrigger = rightTrigger;
this.leftStickX = leftStickX;
this.leftStickY = leftStickY;
this.rightStickX = rightStickX;
this.rightStickY = rightStickY;
}
public byte[] toWire()
{
ByteBuffer bb = ByteBuffer.allocate(PACKET_LENGTH).order(ByteOrder.LITTLE_ENDIAN);
bb.put(toWireHeader());
bb.put(HEADER);
bb.putShort(buttonFlags);
bb.put(leftTrigger);
bb.put(rightTrigger);
bb.putShort(leftStickX);
bb.putShort(leftStickY);
bb.putShort(rightStickX);
bb.putShort(rightStickY);
bb.put(TAIL);
return bb.array();
}
}

26
limelight-pc/src/com/limelight/nvstream/input/NvInputPacket.java Normal file
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package com.limelight.nvstream.input;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
public abstract class NvInputPacket {
public static final int HEADER_LENGTH = 0x4;
protected int packetType;
public NvInputPacket(int packetType)
{
this.packetType = packetType;
}
public abstract byte[] toWire();
public byte[] toWireHeader()
{
ByteBuffer bb = ByteBuffer.allocate(4).order(ByteOrder.BIG_ENDIAN);
bb.putInt(packetType);
return bb.array();
}
}

36
limelight-pc/src/com/limelight/nvstream/input/NvMouseButtonPacket.java Normal file
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package com.limelight.nvstream.input;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
public class NvMouseButtonPacket extends NvInputPacket {
private byte buttonEventType;
public static final int PACKET_TYPE = 0x5;
public static final int PAYLOAD_LENGTH = 5;
public static final int PACKET_LENGTH = PAYLOAD_LENGTH +
NvInputPacket.HEADER_LENGTH;
public static final byte PRESS_EVENT = 0x07;
public static final byte RELEASE_EVENT = 0x08;
public NvMouseButtonPacket(boolean leftButtonDown)
{
super(PACKET_TYPE);
buttonEventType = leftButtonDown ?
PRESS_EVENT : RELEASE_EVENT;
}
@Override
public byte[] toWire() {
ByteBuffer bb = ByteBuffer.allocate(PACKET_LENGTH).order(ByteOrder.BIG_ENDIAN);
bb.put(toWireHeader());
bb.put(buttonEventType);
bb.putInt(1); // FIXME: button index?
return bb.array();
}
}

42
limelight-pc/src/com/limelight/nvstream/input/NvMouseMovePacket.java Normal file
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package com.limelight.nvstream.input;
import java.nio.ByteBuffer;
public class NvMouseMovePacket extends NvInputPacket {
private static final byte[] HEADER =
{
0x06,
0x00,
0x00,
0x00
};
public static final int PACKET_TYPE = 0x8;
public static final int PAYLOAD_LENGTH = 8;
public static final int PACKET_LENGTH = PAYLOAD_LENGTH +
NvInputPacket.HEADER_LENGTH;
private short deltaX;
private short deltaY;
public NvMouseMovePacket(short deltaX, short deltaY)
{
super(PACKET_TYPE);
this.deltaX = deltaX;
this.deltaY = deltaY;
}
@Override
public byte[] toWire() {
ByteBuffer bb = ByteBuffer.allocate(PACKET_LENGTH);
bb.put(toWireHeader());
bb.put(HEADER);
bb.putShort(deltaX);
bb.putShort(deltaY);
return bb.array();
}
}