/*
 * Copyright (c) 2010-2022 Belledonne Communications SARL.
 *
 * This file is part of mediastreamer2
 * (see https://gitlab.linphone.org/BC/public/mediastreamer2).
 *
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU Affero General Public License as
 * published by the Free Software Foundation, either version 3 of the
 * License, or (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU Affero General Public License for more details.
 *
 * You should have received a copy of the GNU Affero General Public License
 * along with this program. If not, see <http://www.gnu.org/licenses/>.
 */

#ifndef MEDIASTREAM_H
#define MEDIASTREAM_H

#include "ortp/port.h"
#include <ortp/event.h>
#include <ortp/nack.h>
#include <ortp/ortp.h>

#include <mediastreamer2/baudot.h>
#include <mediastreamer2/bitratecontrol.h>
#include <mediastreamer2/dtls_srtp.h>
#include <mediastreamer2/ice.h>
#include <mediastreamer2/ms_srtp.h>
#include <mediastreamer2/msequalizer.h>
#include <mediastreamer2/msfactory.h>
#include <mediastreamer2/msfilter.h>
#include <mediastreamer2/msscreensharing.h>
#include <mediastreamer2/mssndcard.h>
#include <mediastreamer2/msticker.h>
#include <mediastreamer2/msvideo.h>
#include <mediastreamer2/msvideoqualitycontroller.h>
#include <mediastreamer2/mswebcam.h>
#include <mediastreamer2/qualityindicator.h>
#include <mediastreamer2/zrtp.h>

#ifdef __cplusplus
extern "C" {
#endif

#define MS_RING_STREAM_MAX_SND_CARDS 3

/**
 * @addtogroup ring_api
 * @{
 **/

struct _RingStream {
	MSSndCard *card[MS_RING_STREAM_MAX_SND_CARDS];
	MSTicker *ticker;
	MSFilter *source;
	MSFilter *gendtmf;
	MSFilter *write_resampler;
	MSFilter *sndwrite[MS_RING_STREAM_MAX_SND_CARDS];
	MSFilter *decoder;
	MSFilter *sndwrite_tee;
	int srcpin;
};

typedef struct _RingStream RingStream;

MS2_PUBLIC RingStream *ring_start(MSFactory *factory, const char *file, int interval, const bctbx_list_t *snd_cards);
MS2_PUBLIC RingStream *ring_start_with_cb(MSFactory *factory,
                                          const char *file,
                                          int interval,
                                          const bctbx_list_t *snd_cards,
                                          MSFilterNotifyFunc func,
                                          void *user_data);
MS2_PUBLIC void ring_stop(RingStream *stream);

/**
 * Asks the ring filter to route to the selected sound card (currently only used for AAudio and OpenSLES)
 * @param[in] stream The RingStream object
 * @param[in] snd_card The wanted audio output soundcard
 * @deprecated 29/08/2025 Use ring_stream_set_output_ms_snd_cards() instead
 */
MS2_PUBLIC MS2_DEPRECATED void ring_stream_set_output_ms_snd_card(RingStream *stream, MSSndCard *snd_card);

/**
 * Retrieve the current sound card from the ring filter (currently only used for AAudio and OpenSLES)
 * @param[in] stream The RingStream object
 * @deprecated 29/08/2025 Use ring_stream_get_output_ms_snd_cards() instead.
 */
MS2_PUBLIC MS2_DEPRECATED MSSndCard *ring_stream_get_output_ms_snd_card(const RingStream *stream);

/**
 * Asks the ring filter to route to the selected sound cards (currently only used for AAudio and OpenSLES)
 * @param stream The RingStream object
 * @param snd_cards The list of wanted audio output soundcards
 */
MS2_PUBLIC void ring_stream_set_output_ms_snd_cards(RingStream *stream, const bctbx_list_t *snd_cards);

/**
 * Retrieve the current list of sound cards from the ring filter (currently only used for AAudio and OpenSLES).
 * @param stream The RingStream object
 * @return The current list of sound cards from the ring filter
 */
MS2_PUBLIC bctbx_list_t *ring_stream_get_output_ms_snd_cards(const RingStream *stream);

/**
 * @}
 **/
/**
 * The MediaStream is an object describing a stream (one of AudioStream or VideoStream).
 **/
typedef struct _MediaStream MediaStream;

/*
 * internal cb to process rtcp stream
 * */
typedef void (*media_stream_process_rtcp_callback_t)(MediaStream *stream, const mblk_t *m);

struct _MSMediaStreamSessions {
	RtpSession *rtp_session;
	RtpSession *fec_session;
	MSSrtpCtx *srtp_context;
	MSZrtpContext *zrtp_context;
	MSDtlsSrtpContext *dtls_context;
	MSTicker *ticker;
	bctbx_list_t *auxiliary_sessions; /**< a list of RtpSessions created by the mediastream on new SSRC
	                                        multiplexed in a bundle */
};

#ifndef MS_MEDIA_STREAM_SESSIONS_DEFINED
typedef struct _MSMediaStreamSessions MSMediaStreamSessions;
#define MS_MEDIA_STREAM_SESSIONS_DEFINED 1
#endif

MS2_PUBLIC void ms_media_stream_sessions_uninit(MSMediaStreamSessions *sessions);

typedef enum _MSStreamState { MSStreamInitialized, MSStreamPreparing, MSStreamStarted, MSStreamStopped } MSStreamState;

typedef enum MediaStreamDir {
	MediaStreamSendRecv,
	MediaStreamSendOnly,
	MediaStreamRecvOnly,
} MediaStreamDir;

/**
 * Base struct for both AudioStream and VideoStream structure.
 **/
struct _MediaStream {
	MSFormatType type;
	MSStreamState state;
	MSMediaStreamSessions sessions;
	OrtpEvQueue *evq;
	MSFilter *rtprecv;
	MSFilter *rtpsend;
	MSFilter *encoder;
	MSFilter *decoder;
	MSFilter *voidsink;
	MSBitrateController *rc;
	MSQualityIndicator *qi;
	IceCheckList *ice_check_list;
	time_t start_time;
	time_t last_iterate_time;
	uint64_t last_packet_count;
	time_t last_packet_time;
	MSQosAnalyzerAlgorithm rc_algorithm;
	PayloadType *current_pt; /*doesn't need to be freed*/
	char *log_tag;
	bool_t rc_enable;
	bool_t is_beginning;
	bool_t owns_sessions;
	bool_t stun_allowed; /* Whether sending basic stun packet as keepalive is permitted, default is yes */
	/**
	 * defines encoder target network bit rate, uses #media_stream_set_target_network_bitrate() setter.
	 * */
	int target_bitrate;
	int max_target_bitrate;
	media_stream_process_rtcp_callback_t process_rtcp;
	OrtpEvDispatcher *evd;
	MSFactory *factory;
	MSBandwidthController *bandwidth_controller;
	MSVideoQualityController *video_quality_controller;
	MediaStreamDir direction;
	FecParams *fec_parameters;
	FecStream *fec_stream;
	bool_t local_mix_conference; /**< true when this client stream is part of a conference perfoming mix locally -
	                                conference server uses transfer session to forward the RTP packet not just the
	                                payload */
	bool_t transfer_mode; /** When enable and the session is part of a bundle, a new SSRC detected on outgoing packet
	                                will create a new RTP Session in transfer mode */
	mblk_t *last_goog_remb_received;
};

MS2_PUBLIC void media_stream_init(MediaStream *stream, MSFactory *factory, const MSMediaStreamSessions *sessions);

MS2_PUBLIC MSFactory *media_stream_get_factory(MediaStream *stream);

MS2_PUBLIC void media_stream_set_log_tag(MediaStream *stream, const char *log_tag);

/**
 * @addtogroup audio_stream_api
 * @{
 **/

MS2_PUBLIC bool_t media_stream_started(MediaStream *stream);

MS2_PUBLIC int media_stream_join_multicast_group(MediaStream *stream, const char *ip);

MS2_PUBLIC bool_t media_stream_dtls_supported(void);

/* enable DTLS on the media stream */
MS2_PUBLIC void media_stream_enable_dtls(MediaStream *stream, const MSDtlsSrtpParams *params);

MS2_PUBLIC void media_stream_set_rtcp_information(MediaStream *stream, const char *cname, const char *tool);

MS2_PUBLIC void media_stream_get_local_rtp_stats(MediaStream *stream, rtp_stats_t *stats);

MS2_PUBLIC void media_stream_get_local_fec_stats(MediaStream *stream, fec_stats *stats);

MS2_PUBLIC int media_stream_set_dscp(MediaStream *stream, int dscp);

MS2_PUBLIC void media_stream_enable_adaptive_bitrate_control(MediaStream *stream, bool_t enabled);

MS2_PUBLIC void media_stream_set_adaptive_bitrate_algorithm(MediaStream *stream, MSQosAnalyzerAlgorithm algorithm);

MS2_PUBLIC void media_stream_enable_adaptive_jittcomp(MediaStream *stream, bool_t enabled);

MS2_PUBLIC void media_stream_set_ice_check_list(MediaStream *stream, IceCheckList *cl);

MS2_PUBLIC void media_stream_set_stun_allowed(MediaStream *stream, bool_t value);

/*
 * deprecated, use media_stream_set_srtp_recv_key and media_stream_set_srtp_send_key.
 **/
MS2_PUBLIC bool_t media_stream_enable_srtp(MediaStream *stream,
                                           MSCryptoSuite suite,
                                           const char *snd_key,
                                           const char *rcv_key);

/**
 * @param[in] stream MediaStream object
 * @return true if stream is encrypted
 * */
MS2_PUBLIC bool_t media_stream_secured(const MediaStream *stream);
#define media_stream_is_secured media_stream_secured

/**
 * Get the source(SDES, ZRTP, DTLS-SRTP) of srtp key used to secure this stream
 * @param[in] 		stream MediaStream object
 * @param[in]		dir	stream direction (send, recv or both)
 * @param[in]		is_inner get the source of the inner encrytion if true
 * @return the srtp key source if they are consistent:
 * 			both direction (send and receive) have the same source when dir is both
 * 			MSSrtpKeySourceUnavailable otherwise
 */
MS2_PUBLIC MSSrtpKeySource media_stream_get_srtp_key_source(const MediaStream *stream,
                                                            MediaStreamDir dir,
                                                            bool_t is_inner);

/**
 * Get the crypto suite used to secure this stream
 * @param[in] 		stream MediaStream object
 * @param[in]		dir	stream direction (send, recv or both)
 * @param[in]		is_inner get the suite of the inner encrytion if true
 * @return the srtp key crypto suite if they are consistent:
 * 			both direction (send and receive) have the same suite when dir is both
 * 			MS_CRYPTO_SUITE_INVALID otherwise
 */
MS2_PUBLIC MSCryptoSuite media_stream_get_srtp_crypto_suite(const MediaStream *stream,
                                                            MediaStreamDir dir,
                                                            bool_t is_inner);

/**
 * Get the source(SDES, ZRTP, DTLS-SRTP) of srtp key used to secure this stream
 * @param[in]		stream_sessions	Pointer to the stream session structure
 * @param[in]		dir	stream direction (send, recv or both)
 * @param[in]		is_inner get the source of the inner encrytion if true
 * @return the srtp key source if they are consistent:
 * 			both direction (send and receive) have the same source when dir is both
 * 			MSSrtpKeySourceUnavailable otherwise
 */
MS2_PUBLIC MSSrtpKeySource ms_media_stream_sessions_get_srtp_key_source(const MSMediaStreamSessions *sessions,
                                                                        MediaStreamDir dir,
                                                                        bool_t is_inner);

/**
 * Get the crypto suite used to secure this stream
 * @param[in]		stream_sessions	Pointer to the stream session structure
 * @param[in]		dir	stream direction (send, recv or both)
 * @param[in]		is_inner get the suite of the inner encrytion if true
 * @return the srtp key crypto suite if they are consistent:
 * 			both direction (send and receive) have the same source when dir is both
 * 			MS_CRYPTO_SUITE_INVALID otherwise
 */
MS2_PUBLIC MSCryptoSuite ms_media_stream_sessions_get_srtp_crypto_suite(const MSMediaStreamSessions *sessions,
                                                                        MediaStreamDir dir,
                                                                        bool_t is_inner);

/**
 * Tells whether AVPF is enabled or not.
 * @param[in] stream #MediaStream object.
 * @return True if AVPF is enabled, false otherwise.
 */
MS2_PUBLIC bool_t media_stream_avpf_enabled(const MediaStream *stream);

/**
 * Gets the AVPF Regular RTCP report interval.
 * @param[in] stream #MediaStream object.
 * @return The AVPF Regular RTCP report interval in seconds.
 */
MS2_PUBLIC uint16_t media_stream_get_avpf_rr_interval(const MediaStream *stream);

/**
 * Gets the RTP session of the media stream.
 * @param[in] stream #MediaStream object.
 * @return The RTP session of the media stream.
 */
MS2_PUBLIC RtpSession *media_stream_get_rtp_session(const MediaStream *stream);

MS2_PUBLIC const MSQualityIndicator *media_stream_get_quality_indicator(MediaStream *stream);
/* *
 * returns a realtime indicator of the stream quality between 0 and 5
 * */
MS2_PUBLIC float media_stream_get_quality_rating(MediaStream *stream);

MS2_PUBLIC float media_stream_get_average_quality_rating(MediaStream *stream);

MS2_PUBLIC float media_stream_get_lq_quality_rating(MediaStream *stream);

MS2_PUBLIC float media_stream_get_average_lq_quality_rating(MediaStream *stream);

/**
 * <br>For multirate codecs like OPUS, encoder output target bitrate must be set.
 * <br>Encoder will compute output codec bitrate from this value.
 * <br> default value is the value corresponding the rtp PayloadType
 * @param stream stream to apply parameter on
 * @param target_bitrate in bit per seconds
 * @return 0 if succeed
 * */
MS2_PUBLIC int media_stream_set_target_network_bitrate(MediaStream *stream, int target_bitrate);

/**
 * Set a maximum target bitrate for the stream. Indeed, the MSBandwidthController may adapt the target bitrate
 * according to network conditions, which includes the possibility to increase it if remote side sends a TMMBR
 * to invite to increase bitrate. The max_network_bitrate defines the upper limit for increasing the bitrate usage
 *automatically.
 * @param stream stream to apply parameter on
 * @param target_bitrate in bit per seconds
 **/
MS2_PUBLIC int media_stream_set_max_network_bitrate(MediaStream *stream, int max_bitrate);

/**
 * get the stream target bitrate.
 * @param stream stream to apply parameter on
 * @return target_bitrate in bit per seconds
 * */
MS2_PUBLIC int media_stream_get_target_network_bitrate(const MediaStream *stream);

/**
 * get current stream  upload bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_up_bw(const MediaStream *stream);

/**
 * get current stream download bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_down_bw(const MediaStream *stream);

/**
 * get current FEC stream upload bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_fec_up_bw(const MediaStream *stream);

/**
 * get current FEC stream download bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_fec_down_bw(const MediaStream *stream);

/**
 * get current stream rtcp upload bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_rtcp_up_bw(const MediaStream *stream);

/**
 * get current stream rtcp download bitrate. Value is updated every seconds
 * @param stream
 * @return bitrate in bit per seconds
 * */
MS2_PUBLIC float media_stream_get_rtcp_down_bw(const MediaStream *stream);

/**
 * Returns the sessions that were used in the media stream (RTP, SRTP, ZRTP...) so that they can be re-used.
 * As a result of calling this function, the media stream no longer owns the sessions and thus will not free them.
 **/
MS2_PUBLIC void media_stream_reclaim_sessions(MediaStream *stream, MSMediaStreamSessions *sessions);

MS2_PUBLIC void media_stream_iterate(MediaStream *stream);

MS2_PUBLIC void media_stream_set_direction(MediaStream *stream, MediaStreamDir dir);

MS2_PUBLIC MediaStreamDir media_stream_get_direction(const MediaStream *stream);

/**
 * Returns TRUE if stream was still actively receiving packets (RTP or RTCP) in the last period specified in
 *timeout_seconds.
 **/
MS2_PUBLIC bool_t media_stream_alive(MediaStream *stream, int timeout_seconds);

/**
 * @return current streams state
 * */
MS2_PUBLIC MSStreamState media_stream_get_state(const MediaStream *stream);

MS2_PUBLIC OrtpEvDispatcher *media_stream_get_event_dispatcher(const MediaStream *stream);

/**
 * Retrieve the receive ssrc of the stream
 *
 * @param stream the media stream
 * @return the receive ssrc of the stream
 */
MS2_PUBLIC uint32_t media_stream_get_recv_ssrc(const MediaStream *stream);

/**
 * Returns the parameters for flexible FEC.
 *
 * @param fec_payload_type the payload type for flexible FEC
 * @return the parameters for flexible FEC
 */
MS2_PUBLIC FecParams *media_stream_extract_fec_params(const PayloadType *fec_payload_type);

/**
 * Create a RTP session with a FEC stream for flexible FEC. The source session is the one given in ms. The rtp bundle
 * mode must be enabled.
 *
 * @param stream the media stream
 */
MS2_PUBLIC void media_stream_create_or_update_fec_session(MediaStream *ms);

/**
 * Destroy the FEC stream of the RTP session of the media stream. The FEC session remains.
 *
 * @param stream the media stream
 */
MS2_PUBLIC void media_stream_destroy_fec_stream(MediaStream *ms);

/**
 * Ask the media stream if it has a flexible FEC stream. Note: the FEC stream can exist and be inactive if the
 * FEC parameters L and D are set to 0 (no repair packets are sent nor received).
 *
 * @param[in] stream The media stream object.
 * @return true if the media stream has a FEC stream, false otherwise.
 */
MS2_PUBLIC bool_t media_stream_fec_enabled(MediaStream *stream);

/**
 * Retrieve the send ssrc of the stream
 *
 * @param stream the media stream
 * @return the send ssrc of the stream
 */
MS2_PUBLIC uint32_t media_stream_get_send_ssrc(const MediaStream *stream);

/**
 * Enable/Disable the local conference mix capabilities. Enable it when the conference server is doing session transfer
 * forwarding whole RTP packet not just the payload On a videostream: force instant resync on incoming SSRC change: so
 * the active speaker session is able to switch from one session to another instantly On an audiostream: allow creation
 * of bundled RtpSession upon reception of a new SSRC
 *
 * Do this setting before start
 */
MS2_PUBLIC void media_stream_enable_conference_local_mix(MediaStream *stream, bool_t enabled);

/**
 * When the stream is in transfer mode, new RTP Sessions are generated
 * when a new SSRC is detected in an outgoing packet
 * to be done before start
 */
MS2_PUBLIC void media_stream_enable_transfer_mode(MediaStream *stream, bool_t enable);

typedef enum EchoLimiterType { ELInactive, ELControlMic, ELControlFull } EchoLimiterType;

typedef enum EqualizerLocation { MSEqualizerHP = 0, MSEqualizerMic } EqualizerLocation;

typedef enum MSResourceType {
	MSResourceInvalid,
	MSResourceDefault,
	MSResourceFile,
	MSResourceRtp,
	MSResourceCamera,
	MSResourceSoundcard,
	MSResourceVoid,
	MSResourceItc,
	MSResourceScreenSharing
} MSResourceType;

MS2_PUBLIC const char *ms_resource_type_to_string(MSResourceType type);

/**
 * Structure describing the input or the output of a MediaStream.
 * type must be set to one the member of the MSResourceType enum, and the correspoding
 * resource argument must be set: the file name (const char*) for MSResourceFile,
 * the RtpSession for MSResourceRtp, an MSWebCam for MSResourceCamera, an MSSndCard for MSResourceSoundcard,
 * an MSFilter for MSResourceItc, an MSScreenSharingDesc for MSResourceScreenSharing
 * @warning due to implementation, if RTP is to be used for input and output, the same RtpSession must be passed for
 * both sides.
 */
typedef struct _MSMediaResource {
	MSResourceType type;
	union {
		void *resource_arg;
		const char *file;
		RtpSession *session;
		MSWebCam *camera;
		MSSndCard *soundcard;
		MSFilter *itc;
		MSScreenSharingDesc screen_sharing;
	};
} MSMediaResource;

MS2_PUBLIC bool_t ms_media_resource_is_consistent(const MSMediaResource *r);
#define ms_media_resource_get_file(r) (((r)->type == MSResourceFile) ? (r)->file : NULL)
#define ms_media_resource_get_rtp_session(r) (((r)->type == MSResourceRtp) ? (r)->session : NULL)
#define ms_media_resource_get_camera(r) (((r)->type == MSResourceCamera) ? (r)->camera : NULL)
#define ms_media_resource_get_soundcard(r) (((r)->type == MSResourceSoundcard) ? (r)->soundcard : NULL)
/**
 * Structure describing the input/output of a MediaStream.
 * Input and output are described as MSMediaResource.
 */
typedef struct _MSMediaStreamIO {
	MSMediaResource input;
	MSMediaResource output;
} MSMediaStreamIO;

#define MS_MEDIA_STREAM_IO_INITIALIZER                                                                                 \
	{                                                                                                                  \
		{MSResourceInvalid}, {                                                                                         \
			MSResourceInvalid                                                                                          \
		}                                                                                                              \
	}

MS2_PUBLIC bool_t ms_media_stream_io_is_consistent(const MSMediaStreamIO *io);

typedef void (*AudioStreamIsSpeakingCallback)(void *user_pointer, uint32_t speaker_ssrc, bool_t is_speaking);
typedef void (*AudioStreamIsMutedCallback)(void *user_pointer, uint32_t speaker_ssrc, bool_t is_muted);
typedef void (*AudioStreamActiveSpeakerCallback)(void *user_pointer, uint32_t speaker_ssrc);
typedef void (*MSAudioRouteChangedCallback)(void *audioStream,
                                            bool_t needReloadSoundDevices,
                                            char *newInputPort,
                                            char *newOutputPort);

struct _AudioStream {
	MediaStream ms;
	MSSndCard *playcard;
	MSSndCard *captcard;
	MSFilter *soundread;
	MSFilter *soundwrite;
	MSFilter *dtmfgen;
	MSFilter *dtmfgen_rtp;
	MSFilter *baudot_generator;
	MSFilter *baudot_detector;
	MSFilter *plc;
	MSFilter *ec;                /*echo canceler*/
	MSFilter *volsend, *volrecv; /*MSVolumes*/
	MSFilter *vad;
	MSFilter *local_mixer;
	MSFilter *local_player;
	MSFilter *local_player_resampler;
	MSFilter *read_decoder;  /* Used when the input is done via RTP */
	MSFilter *write_encoder; /* Used when the output is done via RTP */
	MSFilter *read_resampler;
	MSFilter *write_resampler;
	MSFilter *mic_equalizer;
	MSFilter *spk_equalizer;
	MSFilter *dummy;
	MSFilter *recv_tee;
	MSFilter *recorder_mixer;
	MSFilter *recorder;
	MSFilter *outbound_mixer;
	MSFilter *noise_suppressor;
	struct {
		MSFilter *resampler;
		MSFilter *encoder;
		MSFilter *recorder;
		MSFilter *video_input;
	} av_recorder;
	struct _AVPlayer {
		MSFilter *player;
		MSFilter *resampler;
		MSFilter *decoder;
		MSFilter *video_output;
		int audiopin;
		int videopin;
		bool_t plumbed;
	} av_player;
	MSFilter *flowcontrol;
	RtpSession *rtp_io_session; /**< The RTP session used for RTP input/output. */
	MSFilter *vaddtx;
	char *recorder_file;
	EchoLimiterType el_type; /*use echo limiter: two MSVolume, measured input level controlling local output level*/
	EqualizerLocation eq_loc;
	uint32_t features;
	int sample_rate;
	int nchannels;
	struct _VideoStream *videostream; /*the stream with which this audiostream is paired*/
	MSAudioRoute audio_route;
	bool_t play_dtmfs;
	bool_t use_ec;
	bool_t force_software_ec;
	bool_t use_gc;
	bool_t use_agc;
	bool_t use_ns;

	bool_t mic_eq_active;
	bool_t spk_eq_active;
	bool_t use_ng; /*noise gate*/
	bool_t is_ec_delay_set;
	bool_t disable_record_on_mute;
	float last_mic_gain_level_db;
	struct _AudioStreamVolumes *participants_volumes;
	int mixer_to_client_extension_id;
	int client_to_mixer_extension_id;
	AudioStreamIsSpeakingCallback is_speaking_cb;
	void *is_speaking_user_pointer;
	AudioStreamIsMutedCallback is_muted_cb;
	void *is_muted_user_pointer;
	AudioStreamActiveSpeakerCallback active_speaker_cb;
	void *active_speaker_user_pointer;
	uint32_t active_speaker_ssrc;
	MSAudioRouteChangedCallback audio_route_changed_cb;
	void *audio_route_changed_cb_user_data;
	bctbx_list_t *bundled_recv_branches; /**< a list of AudioStreamMixedRecvBranch added upon reception of new stream in
	                                      a locally mixed audio conference */
};

/**
 * The AudioStream holds all resources to create and run typical VoIP audiostream.
 **/
typedef struct _AudioStream AudioStream;

/**
 * Structure to store an input branch added to an audiostream by conference local mixing upon reception of a new stream
 */
typedef struct _AudioStreamMixedRecvBranch {
	MSFilter *recv;      /**< the receiver filter */
	MSFilter *dec;       /**<  the decoder filter */
	MSFilter *mixer;     /**<  the audio mixer we connected to - needed to unplug this branch */
	MSTicker *ticker;    /**< the ticker used to schedule the recv filter - needed to detach */
	int mixerPin;        /**< pin used as input on the local audio mixer */
	RtpSession *session; /**< the rtp session used in the recv filter -  needed to reset it when a session is recycled
	                        for a new incoming stream and all mixer input are occupied */
} AudioStreamMixedRecvBranch;

/* start a thread that does sampling->encoding->rtp_sending|rtp_receiving->decoding->playing */
MS2_PUBLIC AudioStream *audio_stream_start(MSFactory *factory,
                                           RtpProfile *prof,
                                           int locport,
                                           const char *remip,
                                           int remport,
                                           int payload_type,
                                           int jitt_comp,
                                           bool_t echo_cancel);

MS2_PUBLIC AudioStream *audio_stream_start_with_sndcards(MSFactory *factory,
                                                         RtpProfile *prof,
                                                         int locport,
                                                         const char *remip4,
                                                         int remport,
                                                         int payload_type,
                                                         int jitt_comp,
                                                         MSSndCard *playcard,
                                                         MSSndCard *captcard,
                                                         bool_t echocancel);

MS2_PUBLIC int audio_stream_start_with_files(AudioStream *stream,
                                             RtpProfile *prof,
                                             const char *remip,
                                             int remport,
                                             int rem_rtcp_port,
                                             int pt,
                                             int jitt_comp,
                                             const char *infile,
                                             const char *outfile);

/**
 * Start an audio stream according to the specified AudioStreamIO.
 *
 * @param[in] stream AudioStream object previously created with audio_stream_new().
 * @param[in] profile RtpProfile object holding the PayloadType that can be used during the audio session.
 * @param[in] rem_rtp_ip The remote IP address where to send the encoded audio to.
 * @param[in] rem_rtp_port The remote port where to send the encoded audio to.
 * @param[in] rem_rtcp_ip The remote IP address for RTCP.
 * @param[in] rem_rtcp_port The remote port for RTCP.
 * @param[in] payload_type The payload type number used to send the audio stream. A valid PayloadType must be available
 * at this index in the profile.
 * @param[in] io A MSMediaStreamIO describing the local input/output of the audio stream.
 */
MS2_PUBLIC int audio_stream_start_from_io(AudioStream *stream,
                                          RtpProfile *profile,
                                          const char *rem_rtp_ip,
                                          int rem_rtp_port,
                                          const char *rem_rtcp_ip,
                                          int rem_rtcp_port,
                                          int payload_type,
                                          const MSMediaStreamIO *io);

/**
 * Starts an audio stream from/to local wav files or soundcards.
 *
 * This method starts the processing of the audio stream, that is playing from wav file or soundcard, voice processing,
 *encoding, sending through RTP, receiving from RTP, decoding, voice processing and wav file recording or soundcard
 *playback.
 *
 *
 * @param stream an AudioStream previously created with audio_stream_new().
 * @param profile a RtpProfile containing all PayloadType possible during the audio session.
 * @param rem_rtp_ip remote IP address where to send the encoded audio.
 * @param rem_rtp_port remote IP port where to send the encoded audio.
 * @param rem_rtcp_ip remote IP address for RTCP.
 * @param rem_rtcp_port remote port for RTCP.
 * @param payload payload type index to use for the sending stream. This index must point to a valid PayloadType in the
 *RtpProfile.
 * @param jitt_comp Nominal jitter buffer size in milliseconds.
 * @param infile path to wav file to play out (can be NULL)
 * @param outfile path to wav file to record into (can be NULL)
 * @param playcard The soundcard to be used for playback (can be NULL)
 * @param captcard The soundcard to be used for catpure. (can be NULL)
 * @param use_ec whether echo cancellation is to be performed.
 * @return 0 if sucessful, -1 otherwise.
 **/
MS2_PUBLIC int audio_stream_start_full(AudioStream *stream,
                                       RtpProfile *profile,
                                       const char *rem_rtp_ip,
                                       int rem_rtp_port,
                                       const char *rem_rtcp_ip,
                                       int rem_rtcp_port,
                                       int payload,
                                       int jitt_comp,
                                       const char *infile,
                                       const char *outfile,
                                       MSSndCard *playcard,
                                       MSSndCard *captcard,
                                       bool_t use_ec);

MS2_PUBLIC void audio_stream_play(AudioStream *st, const char *name);
MS2_PUBLIC void audio_stream_record(AudioStream *st, const char *name);

static MS2_INLINE void audio_stream_set_rtcp_information(AudioStream *st, const char *cname, const char *tool) {
	media_stream_set_rtcp_information(&st->ms, cname, tool);
}

MS2_PUBLIC void audio_stream_play_received_dtmfs(AudioStream *st, bool_t yesno);

/**
 * Creates an AudioStream object listening on a RTP port.
 * @param loc_rtp_port the local UDP port to listen for RTP packets.
 * @param loc_rtcp_port the local UDP port to listen for RTCP packets
 * @param ipv6 TRUE if ipv6 must be used.
 * @param factory
 * @return a new AudioStream.
 **/
MS2_PUBLIC AudioStream *audio_stream_new(MSFactory *factory, int loc_rtp_port, int loc_rtcp_port, bool_t ipv6);

/**
 * Creates an AudioStream object listening on a RTP port for a dedicated address.
 * @param loc_ip the local ip to listen for RTP packets. Can be ::, O.O.O.O or any ip4/6 addresses
 * @param loc_rtp_port the local UDP port to listen for RTP packets.
 * @param loc_rtcp_port the local UDP port to listen for RTCP packets
 * @param factory
 * @return a new AudioStream.
 **/
MS2_PUBLIC AudioStream *audio_stream_new2(MSFactory *factory, const char *ip, int loc_rtp_port, int loc_rtcp_port);

/**Creates an AudioStream object from initialized MSMediaStreamSessions.
 * @param sessions the MSMediaStreamSessions
 * @param factory the MSFActory from the core object
 * @return a new AudioStream
 **/
MS2_PUBLIC AudioStream *audio_stream_new_with_sessions(MSFactory *factory, const MSMediaStreamSessions *sessions);

#define AUDIO_STREAM_FEATURE_PLC (1 << 0)
#define AUDIO_STREAM_FEATURE_EC (1 << 1)
#define AUDIO_STREAM_FEATURE_EQUALIZER (1 << 2)
#define AUDIO_STREAM_FEATURE_VOL_SND (1 << 3)
#define AUDIO_STREAM_FEATURE_VOL_RCV (1 << 4)
#define AUDIO_STREAM_FEATURE_DTMF (1 << 5)
#define AUDIO_STREAM_FEATURE_DTMF_ECHO (1 << 6)
#define AUDIO_STREAM_FEATURE_MIXED_RECORDING (1 << 7)
#define AUDIO_STREAM_FEATURE_LOCAL_PLAYING (1 << 8)
#define AUDIO_STREAM_FEATURE_REMOTE_PLAYING (1 << 9)
#define AUDIO_STREAM_FEATURE_FLOW_CONTROL (1 << 10)
#define AUDIO_STREAM_FEATURE_VAD (1 << 11)
#define AUDIO_STREAM_FEATURE_BAUDOT (1 << 12)
#define AUDIO_STREAM_FEATURE_NOISE_SUPPRESSION (1 << 13)

#define AUDIO_STREAM_FEATURE_ALL                                                                                       \
	(AUDIO_STREAM_FEATURE_PLC | AUDIO_STREAM_FEATURE_EC | AUDIO_STREAM_FEATURE_EQUALIZER |                             \
	 AUDIO_STREAM_FEATURE_VOL_SND | AUDIO_STREAM_FEATURE_VOL_RCV | AUDIO_STREAM_FEATURE_DTMF |                         \
	 AUDIO_STREAM_FEATURE_DTMF_ECHO | AUDIO_STREAM_FEATURE_MIXED_RECORDING | AUDIO_STREAM_FEATURE_LOCAL_PLAYING |      \
	 AUDIO_STREAM_FEATURE_REMOTE_PLAYING | AUDIO_STREAM_FEATURE_FLOW_CONTROL | AUDIO_STREAM_FEATURE_BAUDOT |           \
	 AUDIO_STREAM_FEATURE_NOISE_SUPPRESSION)

MS2_PUBLIC uint32_t audio_stream_get_features(AudioStream *st);
MS2_PUBLIC void audio_stream_set_features(AudioStream *st, uint32_t features);

MS2_PUBLIC void audio_stream_prepare_sound(AudioStream *st, MSSndCard *playcard, MSSndCard *captcard);
MS2_PUBLIC void audio_stream_unprepare_sound(AudioStream *st);
MS2_PUBLIC bool_t audio_stream_started(AudioStream *stream);
/**
 * Starts an audio stream from local soundcards.
 *
 * This method starts the processing of the audio stream, that is capture from soundcard, voice processing, encoding,
 * sending through RTP, receiving from RTP, decoding, voice processing and soundcard playback.
 *
 * @param stream an AudioStream previously created with audio_stream_new().
 * @param prof a RtpProfile containing all PayloadType possible during the audio session.
 * @param remip remote IP address where to send the encoded audio.
 * @param remport remote IP port where to send the encoded audio
 * @param rem_rtcp_port remote port for RTCP.
 * @param payload_type payload type index to use for the sending stream. This index must point to a valid PayloadType in
 *the RtpProfile.
 * @param jitt_comp Nominal jitter buffer size in milliseconds.
 * @param playcard The soundcard to be used for playback
 * @param captcard The soundcard to be used for catpure.
 * @param echo_cancel whether echo cancellation is to be performed.
 **/
MS2_PUBLIC int audio_stream_start_now(AudioStream *stream,
                                      RtpProfile *prof,
                                      const char *remip,
                                      int remport,
                                      int rem_rtcp_port,
                                      int payload_type,
                                      int jitt_comp,
                                      MSSndCard *playcard,
                                      MSSndCard *captcard,
                                      bool_t echo_cancel);
MS2_PUBLIC void audio_stream_set_relay_session_id(AudioStream *stream, const char *relay_session_id);
/*returns true if we are still receiving some data from remote end in the last timeout seconds*/
MS2_PUBLIC bool_t audio_stream_alive(AudioStream *stream, int timeout);

/**
 * Executes background low priority tasks related to audio processing (RTP statistics analysis).
 * It should be called periodically, for example with an interval of 100 ms or so.
 */
MS2_PUBLIC void audio_stream_iterate(AudioStream *stream);

/**
 * enable echo-limiter dispositve: one MSVolume in input branch controls a MSVolume in the output branch
 * */
MS2_PUBLIC void audio_stream_enable_echo_limiter(AudioStream *stream, EchoLimiterType type);

/**
 * enable gain control, to be done before start()
 * */
MS2_PUBLIC void audio_stream_enable_gain_control(AudioStream *stream, bool_t val);

/**
 * enable automatic gain control, to be done before start()
 * */
MS2_PUBLIC void audio_stream_enable_automatic_gain_control(AudioStream *stream, bool_t val);

/**
 * to be done before start
 *  */
MS2_PUBLIC void audio_stream_set_echo_canceller_params(AudioStream *st, int tail_len_ms, int delay_ms, int framesize);

/**
 * to be done before start
 *  */
MS2_PUBLIC void audio_stream_enable_echo_canceller(AudioStream *st, bool_t enabled);

/**
 * Forces the use of a software echo canceller if available instead of hardwared/built-in one.
 */
MS2_PUBLIC void audio_stream_force_software_echo_canceller(AudioStream *st, bool_t force);

/**
 * enable adaptive rate control
 * */
static MS2_INLINE void audio_stream_enable_adaptive_bitrate_control(AudioStream *stream, bool_t enabled) {
	media_stream_enable_adaptive_bitrate_control(&stream->ms, enabled);
}

/**
 *  Enable adaptive jitter compensation
 *  */
static MS2_INLINE void audio_stream_enable_adaptive_jittcomp(AudioStream *stream, bool_t enabled) {
	media_stream_enable_adaptive_jittcomp(&stream->ms, enabled);
}
/**
 * Calling this method with disable=true will cause the microhone to be completely deactivated when muted
 * Currently only implemented for IOS, this will cause the playback sound to be interrupted for a short moment while the
 * audio is reconfigured. On IOS 14, it will also disable Apple's microphone recording indicator when microphone is
 * muted.
 *
 * @param stream The stream.
 * @param disable True if you wish to entirely stop the audio recording when muting the microphone.
 */
MS2_PUBLIC void audio_stream_disable_record_on_mute(AudioStream *stream, bool_t disable);

/**
 * Enable or disable noise suppression on audio streams. This feature is supported only if the noise suppressor filter
 * has been built.
 *
 * @param stream The stream.
 * @param enable Wether the noise suppression must be enabled.
 */
MS2_PUBLIC void audio_stream_enable_noise_suppression(AudioStream *stream, bool_t enable);

/**
 * Mute or unmute the microphone
 * For IOS, if "audio_stream_disable_record_on_mute" was enabled, this will completely stop the microphone recording.
 * Else, sound recording remains active but silence is sent instead of recorded audiow@
 *
 * @param stream The stream.
 * @param enable Wether the microphone must be enabled.
 */
MS2_PUBLIC void audio_stream_enable_mic(AudioStream *stream, bool_t enable);

/**
 * Apply a software gain on the microphone.
 * Be note that this method neither changes the volume gain
 * of the sound card nor the system mixer one. If you intends
 * to control the volume gain at sound card level, you should
 * use audio_stream_set_sound_card_input_gain() instead.
 *
 * @param stream The stream.
 * @param gain_db Gain to apply in dB.
 */
MS2_PUBLIC void audio_stream_set_mic_gain_db(AudioStream *stream, float gain_db);

/**
 * @deprecated
 * Like audio_stream_set_mic_gain_db() excepted that the gain is specified
 * in percentage.
 *
 * @param stream The stream.
 * @param gain Gain to apply in percentage of the max supported gain.
 */
MS2_PUBLIC void audio_stream_set_mic_gain(AudioStream *stream, float gain);

/**
 *  enable/disable rtp stream
 */
MS2_PUBLIC void audio_stream_mute_rtp(AudioStream *stream, bool_t val);

/**
 * Apply a gain on received RTP packets.
 * @param stream An AudioStream.
 * @param gain_db Gain to apply in dB.
 */
MS2_PUBLIC void audio_stream_set_spk_gain_db(AudioStream *stream, float gain_db);

/**
 * Like audio_stream_set_spk_gain_db() excepted that the gain is specified
 * in percentage.
 *
 * @param stream The stream.
 * @param gain Gain to apply in percentage of the max supported gain.
 */
MS2_PUBLIC void audio_stream_set_spk_gain(AudioStream *stream, float gain);

/**
 * Set microphone volume gain.
 * If the sound backend supports it, the set volume gain will be synchronized
 * with the host system mixer. If you intended to apply a static software gain,
 * you should use audio_stream_set_mic_gain_db() or audio_stream_set_mic_gain().
 *
 * @param stream The audio stream.
 * @param gain Percentage of the max supported volume gain. Valid values are in [0.0 : 1.0].
 */
MS2_PUBLIC void audio_stream_set_sound_card_input_gain(AudioStream *stream, float gain);

/**
 * Get microphone volume gain.
 * @param stream The audio stream.
 * @return double Volume gain in percentage of the max suppored gain.
 * Valid returned values are in [0.0 : 1.0]. A negative value is returned in case of failure.
 */
MS2_PUBLIC float audio_stream_get_sound_card_input_gain(const AudioStream *stream);

/**
 * Set speaker volume gain.
 * If the sound backend supports it, the set volume gain will be synchronized
 * with the host system mixer.
 * @param stream The audio stream.
 * @param gain Percentage of the max supported volume gain. Valid values are in [0.0 : 1.0].
 */
MS2_PUBLIC void audio_stream_set_sound_card_output_gain(AudioStream *stream, float volume);

/**
 * Get speaker volume gain.
 * @param stream The audio stream.
 * @return Volume gain in percentage of the max suppored gain.
 * Valid returned values are in [0.0 : 1.0]. A negative value is returned in case of failure.
 */
MS2_PUBLIC float audio_stream_get_sound_card_output_gain(const AudioStream *stream);

/**
 * enable noise gate, must be done before start()
 * */
MS2_PUBLIC void audio_stream_enable_noise_gate(AudioStream *stream, bool_t val);

/**
 * Enable a parametric equalizer
 * @param[in] stream An AudioStream
 * @param[in] location Location of the equalizer to enable (speaker or microphone)
 * @param[in] enabled Whether the equalizer must be enabled
 */
MS2_PUBLIC void audio_stream_enable_equalizer(AudioStream *stream, EqualizerLocation location, bool_t enabled);

/**
 * Apply a gain on a given frequency band.
 * @param[in] stream An AudioStream
 * @param[in] location Location of the concerned equalizer (speaker or microphone)
 * @param[in] gain Description of the band and the gain to apply.
 */
MS2_PUBLIC void
audio_stream_equalizer_set_gain(AudioStream *stream, EqualizerLocation location, const MSEqualizerGain *gain);

/**
 *  stop the audio streaming thread and free everything
 *  */
MS2_PUBLIC void audio_stream_stop(AudioStream *stream);

/**
 *  send a dtmf
 *  */
MS2_PUBLIC int audio_stream_send_dtmf(AudioStream *stream, char dtmf);

/**
 *  Are telephone events supported? Used to determine if DTMFs can be sent out-of-band
 */
MS2_PUBLIC bool_t audio_stream_supports_telephone_events(AudioStream *stream);

MS2_PUBLIC MSFilter *audio_stream_get_local_player(AudioStream *stream);

MS2_PUBLIC int audio_stream_set_mixed_record_file(AudioStream *st, const char *filename);

MS2_PUBLIC int audio_stream_mixed_record_start(AudioStream *st);

MS2_PUBLIC int audio_stream_mixed_record_stop(AudioStream *st);

/**
 * Open a player to play an audio/video file to remote end.
 * The player is returned as a MSFilter so that application can make usual player controls on it using the
 *MSPlayerInterface.
 **/
MS2_PUBLIC MSFilter *audio_stream_open_remote_play(AudioStream *stream, const char *filename);

MS2_PUBLIC void audio_stream_close_remote_play(AudioStream *stream);

MS2_PUBLIC void audio_stream_set_default_card(int cardindex);

/* retrieve RTP statistics*/
static MS2_INLINE void audio_stream_get_local_rtp_stats(AudioStream *stream, rtp_stats_t *stats) {
	media_stream_get_local_rtp_stats(&stream->ms, stats);
}

/* returns a realtime indicator of the stream quality between 0 and 5 */
MS2_PUBLIC float audio_stream_get_quality_rating(AudioStream *stream);

/* returns the quality rating as an average since the start of the streaming session.*/
MS2_PUBLIC float audio_stream_get_average_quality_rating(AudioStream *stream);

/* returns a realtime indicator of the listening quality of the stream between 0 and 5 */
MS2_PUBLIC float audio_stream_get_lq_quality_rating(AudioStream *stream);

/* returns the listening quality rating as an average since the start of the streaming session.*/
MS2_PUBLIC float audio_stream_get_average_lq_quality_rating(AudioStream *stream);

/* enable ZRTP on the audio stream */
MS2_PUBLIC void audio_stream_enable_zrtp(AudioStream *stream, MSZrtpParams *params);
MS2_PUBLIC void audio_stream_start_zrtp(AudioStream *stream);

/**
 * return TRUE if zrtp is enabled, it does not mean that stream is encrypted, but only that zrtp is configured to know
 * encryption status, uses #
 * */
bool_t audio_stream_zrtp_enabled(const AudioStream *stream);

/* enable SRTP on the audio stream */
static MS2_INLINE bool_t audio_stream_enable_srtp(AudioStream *stream,
                                                  MSCryptoSuite suite,
                                                  const char *snd_key,
                                                  const char *rcv_key) {
	return media_stream_enable_srtp(&stream->ms, suite, snd_key, rcv_key);
}

static MS2_INLINE int audio_stream_set_dscp(AudioStream *stream, int dscp) {
	return media_stream_set_dscp(&stream->ms, dscp);
}

/**
 * Gets the RTP session of an audio stream.
 * @param[in] stream #MediaStream object.
 * @return The RTP session of the audio stream.
 */
static MS2_INLINE RtpSession *audio_stream_get_rtp_session(const AudioStream *stream) {
	return media_stream_get_rtp_session(&stream->ms);
}

/**
 * Sets the header extension id for mixer to client audio level indication.
 * This has to be called before starting the audio stream.
 *
 * @param stream the audio stream
 * @param extension_id the extension id
 */
MS2_PUBLIC void audio_stream_set_mixer_to_client_extension_id(AudioStream *stream, int extension_id);

/**
 * Sets the header extension id for client to mixer audio level indication.
 * This has to be called before starting the audio stream.
 *
 * @param stream the audio stream
 * @param extension_id the extension id
 */
MS2_PUBLIC void audio_stream_set_client_to_mixer_extension_id(AudioStream *stream, int extension_id);

MS2_PUBLIC void
audio_stream_set_is_speaking_callback(AudioStream *s, AudioStreamIsSpeakingCallback cb, void *user_pointer);
MS2_PUBLIC void audio_stream_set_is_muted_callback(AudioStream *s, AudioStreamIsMutedCallback cb, void *user_pointer);
MS2_PUBLIC void
audio_stream_set_active_speaker_callback(AudioStream *s, AudioStreamActiveSpeakerCallback cb, void *user_pointer);

/**
 * Retrieve the volume of the given participant.
 *
 * @param stream the audio stream
 * @param participant_ssrc the ssrc of the participant
 * @return the volume of the participant in dbm0, if participant isn't found it will return the lowest volume.
 */
MS2_PUBLIC int audio_stream_get_participant_volume(const AudioStream *stream, uint32_t participant_ssrc);

/**
 * Retrieve the receive ssrc of the stream
 *
 * @param stream the audio stream
 * @return the receive ssrc of the stream
 */
MS2_PUBLIC uint32_t audio_stream_get_recv_ssrc(const AudioStream *stream);

/**
 * Retrieve the send ssrc of the stream
 *
 * @param stream the audio stream
 * @return the send ssrc of the stream
 */
MS2_PUBLIC uint32_t audio_stream_get_send_ssrc(const AudioStream *stream);

/* Map api to be able to keep participants volumes */
typedef struct _AudioStreamVolumes AudioStreamVolumes;

#define AUDIOSTREAMVOLUMES_NOT_FOUND INT16_MIN

MS2_PUBLIC AudioStreamVolumes *audio_stream_volumes_new(void);
MS2_PUBLIC void audio_stream_volumes_delete(AudioStreamVolumes *volumes);

MS2_PUBLIC void audio_stream_volumes_insert(AudioStreamVolumes *volumes, uint32_t ssrc, int volume);
MS2_PUBLIC void audio_stream_volumes_erase(AudioStreamVolumes *volumes, uint32_t ssrc);
MS2_PUBLIC void audio_stream_volumes_clear(AudioStreamVolumes *volumes);

MS2_PUBLIC int audio_stream_volumes_size(AudioStreamVolumes *volumes);
MS2_PUBLIC int audio_stream_volumes_find(AudioStreamVolumes *volumes, uint32_t ssrc);
MS2_PUBLIC int audio_stream_volumes_append(AudioStreamVolumes *volumes, AudioStreamVolumes *append);

MS2_PUBLIC void audio_stream_volumes_reset_values(AudioStreamVolumes *volumes);

MS2_PUBLIC void audio_stream_volumes_populate_audio_levels(AudioStreamVolumes *volumes,
                                                           rtp_audio_level_t *audio_levels);

MS2_PUBLIC bool_t audio_stream_volumes_is_speaking(AudioStreamVolumes *volumes);
MS2_PUBLIC uint32_t audio_stream_volumes_get_best(AudioStreamVolumes *volumes);

/**
 * Asks the audio playback filter to route to the selected device (currently only used for blackberry)
 * @param[in] stream The AudioStream object
 * @param[in] route The wanted audio output device (earpiece, speaker)
 */
MS2_PUBLIC void audio_stream_set_audio_route(AudioStream *stream, MSAudioRoute route);

/**
 *  Set a callback to be called when an audio route change is notified (IOS only)
 * @param[in] stream The AudioStream object
 * @param[in] callback The function to call when receiving the route change event
 * @param[in] audio_route_changed_cb_user_data Context for the callback
 */
MS2_PUBLIC void audio_stream_set_audio_route_changed_callback(AudioStream *stream,
                                                              MSAudioRouteChangedCallback callback,
                                                              void *audio_route_changed_cb_user_data);
/**
 * Asks the audio capture filter to route to the selected sound card (currently only used for AAudio and OpenSLES)
 * @param[in] stream The AudioStream object
 * @param[in] sndcard_capture The wanted audio input soundcard
 * @return '0' if MS_AUDIO_CAPTURE_SET_INTERNAL_ID has been successfully called, '-1' if not
 */
MS2_PUBLIC int audio_stream_set_input_ms_snd_card(AudioStream *stream, MSSndCard *sndcard_capture);

/**
 * Asks the audio playback filter to route to the selected sound card (currently only used for AAudio and OpenSLES)
 * @param[in] stream The AudioStream object
 * @param[in] sndcard_playback The wanted audio output soundcard
 * @return '0' if MS_AUDIO_PLAYBACK_SET_INTERNAL_ID has been successfully called, '-1' if not
 */
MS2_PUBLIC int audio_stream_set_output_ms_snd_card(AudioStream *stream, MSSndCard *sndcard_playback);

/**
 * Retrieve the current sound card from the audio capture filter (currently only used for AAudio and OpenSLES)
 * @param[in] stream The AudioStream object
 */
MS2_PUBLIC MSSndCard *audio_stream_get_input_ms_snd_card(AudioStream *stream);

/**
 * Retrieve the current sound card from the audio playback filter (currently only used for AAudio and OpenSLES)
 * @param[in] stream The AudioStream object
 */
MS2_PUBLIC MSSndCard *audio_stream_get_output_ms_snd_card(AudioStream *stream);

/**
 * Send character as Baudot tones on the audio stream.
 * @param[in] stream The AudioStream object
 * @param[in] c The character to be sent
 */
MS2_PUBLIC void audio_stream_send_baudot_character(AudioStream *stream, const char c);

/**
 * Send text as Baudot tones on the audio stream.
 * @param[in] stream The AudioStream object
 * @param[in] text The text to be sent
 */
MS2_PUBLIC void audio_stream_send_baudot_string(AudioStream *stream, const char *text);

/**
 * Enable/disable Baudot tones decoding.
 * @param[in] stream The AudioStream object
 * @param[in] enabled Whether to enable or disable Baudot tones decoding.
 */
MS2_PUBLIC void audio_stream_enable_baudot_decoding(AudioStream *stream, bool_t enabled);

/**
 * Set the Baudot mode to use in the sending_path.
 * @param[in] stream The AudioStream object
 * @param[in] mode The Baudot mode to use in the sending path.
 */
MS2_PUBLIC void audio_stream_set_baudot_sending_mode(AudioStream *stream, MSBaudotMode mode);

/**
 * Set the Baudot significant pause timeout after which a LETTERS tone is retransmitted before resuming transmission (in
 * seconds). Default is 5s.
 * @param[in] stream The AudioStream object
 * @param[in] seconds The significant pause timeout in seconds.
 */
MS2_PUBLIC void audio_stream_set_baudot_pause_timeout(AudioStream *stream, uint8_t seconds);

/**
 * Enable/disable Baudot tones detection.
 * @param[in] stream The AudioStream object
 * @param[in] enabled Whether to enable or disable Baudot tones detection.
 */
MS2_PUBLIC void audio_stream_enable_baudot_detection(AudioStream *stream, bool_t enabled);

/**
 * @}
 **/

/**
 * @addtogroup video_stream_api
 * @{
 **/

typedef void (*VideoStreamRenderCallback)(void *user_pointer,
                                          const MSPicture *local_view,
                                          const MSPicture *remote_view);
typedef void (*VideoStreamEventCallback)(void *user_pointer,
                                         const MSFilter *f,
                                         const unsigned int event_id,
                                         const void *args);
typedef void (*VideoStreamDisplayCallback)(
    void *user_pointer,
    const unsigned int event_id,
    const void *args); /* if coming from OpenGL and event_id==MS_VIDEO_DISPLAY_ERROR_OCCURRED, args is an int from
                          eglGetError() : https://registry.khronos.org/EGL/sdk/docs/man/html/eglGetError.xhtml */
typedef void (*VideoStreamCameraNotWorkingCallback)(void *user_pointer, const MSWebCam *old_webcam);
typedef void (*VideoStreamEncoderControlCb)(struct _VideoStream *, unsigned int method_id, void *arg, void *user_data);
typedef void (*VideoStreamCsrcChangedCb)(void *user_pointer, uint32_t new_csrc);

struct _MediastreamVideoStat {
	int counter_rcvd_pli;  /*Picture Loss Indication counter */
	int counter_rcvd_sli;  /* Slice Loss Indication counter */
	int counter_rcvd_rpsi; /*Reference Picture Selection Indication */
	int counter_rcvd_fir;  /* Full INTRA-frame Request */
};

typedef struct _MediastreamVideoStat MediaStreamVideoStat;

typedef enum _MSVideoContent { MSVideoContentDefault, MSVideoContentSpeaker, MSVideoContentThumbnail } MSVideoContent;

// This number should not be higher than 9 since the VideoAggregator filter only has 10 inputs.
// The main RtpSession always takes the first input.
#define VIDEO_STREAM_MAX_BRANCHES 3

/**
 * Structure to store an input branch added to a videostream upon reception of a new stream
 */
typedef struct _VideoStreamRecvBranch {
	MSFilter *recv;      /**< the receiver filter */
	RtpSession *session; /**< the rtp session used in the recv filter -  needed to reset it when a session is recycled
	                        for a new incoming stream and all branches are occupied */
} VideoStreamRecvBranch;

struct _VideoStream {
	MediaStream ms;
	MSFilter *jpegwriter;
	MSFilter *local_jpegwriter;
	MSFilter *output;
	MSFilter *output2;
	MSFilter *pixconv;
	MSFilter *qrcode;
	MSFilter *recorder_output; /*can be an ItcSink to send video to the audiostream's multimedia recorder, or directly a
	                              MkvRecorder */
	MSFilter *sizeconv;
	MSFilter *source;
	MSFilter *tee;
	MSFilter *tee2;
	MSFilter *tee3;
	MSFilter *void_source;
	MSFilter *itcsink;
	MSFilter *forward_sink;
	MSFilter *aggregator;
	VideoStreamRecvBranch branches[VIDEO_STREAM_MAX_BRANCHES];
	MSVideoSize sent_vsize;
	MSVideoSize preview_vsize;
	MSVideoSize max_sent_vsize;
	float forced_fps;     /*the target fps explicitely set by application, overrides internally selected fps*/
	float configured_fps; /*the fps that was configured to the encoder. It might be different from the one really
	                         obtained from camera.*/
	float real_fps;       /*the fps obtained from camera.*/
	int corner;           /*for selfview*/
	VideoStreamRenderCallback rendercb;
	void *render_pointer;
	VideoStreamEventCallback eventcb;
	void *event_pointer;
	VideoStreamDisplayCallback displaycb;
	void *display_pointer;
	char *display_name;
	void *window_id;
	void *preview_window_id;
	void *video_descriptor;
	MediaStreamDir dir; /* Not used anymore, see direction in MediaStream */
	MSRect decode_rect; // Used for the qrcode decoder
	MSWebCam *cam;
	RtpSession *rtp_io_session; /**< The RTP session used for RTP input/output. */
	char *preset;
	MSVideoConfiguration *vconf_list;
	struct _AudioStream *audiostream; /*the audio stream with which this videostream is paired*/
	OrtpNackContext *nack_context;
	int device_orientation; /* warning: meaning of this variable depends on the platform (Android, iOS, ...) */
	uint64_t last_reported_decoding_error_time;
	uint64_t last_fps_check;
	uint64_t last_camera_check;
	int dead_camera_check_count;
	VideoStreamCameraNotWorkingCallback cameracb;
	void *camera_pointer;
	MediaStreamVideoStat ms_video_stat;
	VideoStreamEncoderControlCb encoder_control_cb;
	void *encoder_control_cb_user_data;
	MSVideoDisplayMode display_mode;
	MSVideoDisplayMode preview_display_mode;
	int frame_marking_extension_id;
	char *label;
	MSVideoContent content;
	bool_t use_preview_window;
	bool_t enable_qrcode_decoder;
	bool_t freeze_on_error;
	bool_t display_filter_auto_rotate_enabled;
	bool_t source_performs_encoding;
	bool_t output_performs_decoding;
	bool_t player_active;
	bool_t staticimage_webcam_fps_optimization; /* if TRUE, the StaticImage webcam will ignore the fps target in order
	                                               to save CPU time. Default is TRUE */
	bool_t is_forwarding;
	bool_t wait_for_frame_decoded;
	bool_t fallback_to_dummy_codec; /**< if TRUE, enable fallback to dummy codec when the requested one is not supported
	                                   - used to build a videostream in the conference server */
	VideoStreamCsrcChangedCb csrc_changed_cb;
	void *csrc_changed_cb_user_data;
	uint32_t new_csrc;
	bool_t is_thumbnail; /* if TRUE, the stream is generated from ItcResource and is SizeConverted */
	FecStream *fec_stream;
	bool_t local_screen_sharing_enabled;
	bool_t active_speaker_mode;
	bool_t csrc_change_received;
};

typedef struct _VideoStream VideoStream;

MS2_PUBLIC VideoStream *video_stream_new(MSFactory *factory, int loc_rtp_port, int loc_rtcp_port, bool_t use_ipv6);
/**
 * Creates a VideoStream object listening on a RTP port for a dedicated address.
 * @param loc_ip the local ip to listen for RTP packets. Can be ::, O.O.O.O or any ip4/6 addresses
 * @param [in] loc_rtp_port the local UDP port to listen for RTP packets.
 * @param [in] loc_rtcp_port the local UDP port to listen for RTCP packets
 * @return a new VideoStream.
 **/
MS2_PUBLIC VideoStream *video_stream_new2(MSFactory *factory, const char *ip, int loc_rtp_port, int loc_rtcp_port);

MS2_PUBLIC VideoStream *video_stream_new_with_sessions(MSFactory *factory, const MSMediaStreamSessions *sessions);

/**
 * Use media_stream_set_direction() instead
 **/
MS2_DEPRECATED MS2_PUBLIC void video_stream_set_direction(VideoStream *vs, MediaStreamDir dir);
static MS2_INLINE void video_stream_enable_adaptive_bitrate_control(VideoStream *stream, bool_t enabled) {
	media_stream_enable_adaptive_bitrate_control(&stream->ms, enabled);
}
static MS2_INLINE void video_stream_enable_adaptive_jittcomp(VideoStream *stream, bool_t enabled) {
	media_stream_enable_adaptive_jittcomp(&stream->ms, enabled);
}
MS2_PUBLIC void video_stream_set_render_callback(VideoStream *s, VideoStreamRenderCallback cb, void *user_pointer);
MS2_PUBLIC void video_stream_set_event_callback(VideoStream *s, VideoStreamEventCallback cb, void *user_pointer);
MS2_PUBLIC void video_stream_set_display_callback(VideoStream *s, VideoStreamDisplayCallback cb, void *user_pointer);

MS2_PUBLIC void video_stream_set_camera_not_working_callback(VideoStream *s,
                                                             VideoStreamCameraNotWorkingCallback cb,
                                                             void *user_pointer);
MS2_PUBLIC void video_stream_set_display_filter_name(VideoStream *s, const char *fname);
MS2_PUBLIC void video_stream_set_label(VideoStream *s, const char *label);
MS2_PUBLIC void video_stream_set_content(VideoStream *s, MSVideoContent content);
MS2_PUBLIC MSVideoContent video_stream_get_content(const VideoStream *vs);

MS2_PUBLIC int video_stream_start_with_source(VideoStream *stream,
                                              RtpProfile *profile,
                                              const char *rem_rtp_ip,
                                              int rem_rtp_port,
                                              const char *rem_rtcp_ip,
                                              int rem_rtcp_port,
                                              int payload,
                                              int jitt_comp,
                                              MSWebCam *cam,
                                              MSFilter *source);
MS2_PUBLIC int video_stream_start(VideoStream *stream,
                                  RtpProfile *profile,
                                  const char *rem_rtp_ip,
                                  int rem_rtp_port,
                                  const char *rem_rtcp_ip,
                                  int rem_rtcp_port,
                                  int payload,
                                  int jitt_comp,
                                  MSWebCam *device);
MS2_PUBLIC int video_stream_start_with_files(VideoStream *stream,
                                             RtpProfile *profile,
                                             const char *rem_rtp_ip,
                                             int rem_rtp_port,
                                             const char *rem_rtcp_ip,
                                             int rem_rtcp_port,
                                             int payload_type,
                                             const char *play_file,
                                             const char *record_file);

/**
 * Start a video stream according to the specified VideoStreamIO.
 *
 * @param[in] stream VideoStream object previously created with video_stream_new().
 * @param[in] profile RtpProfile object holding the PayloadType that can be used during the video session.
 * @param[in] rem_rtp_ip The remote IP address where to send the encoded video to.
 * @param[in] rem_rtp_port The remote port where to send the encoded video to.
 * @param[in] rem_rtcp_ip The remote IP address for RTCP.
 * @param[in] rem_rtcp_port The remote port for RTCP.
 * @param[in] payload_type The payload type number used to send the video stream. A valid PayloadType must be available
 * at this index in the profile.
 * @param[in] io A VideoStreamIO describing the input/output of the video stream.
 */
MS2_PUBLIC int video_stream_start_from_io(VideoStream *stream,
                                          RtpProfile *profile,
                                          const char *rem_rtp_ip,
                                          int rem_rtp_port,
                                          const char *rem_rtcp_ip,
                                          int rem_rtcp_port,
                                          int payload_type,
                                          const MSMediaStreamIO *io);

/**
 * Link a video stream with ItcSink filter. Used for starting another video stream.
 *
 * @param[in] stream VideoStream object previously created with video_stream_new().
 */
MS2_PUBLIC void link_video_stream_with_itc_sink(VideoStream *stream);

MS2_PUBLIC void video_stream_prepare_video(VideoStream *stream);
MS2_PUBLIC void video_stream_unprepare_video(VideoStream *stream);

MS2_PUBLIC void video_stream_set_relay_session_id(VideoStream *stream, const char *relay_session_id);
static MS2_INLINE void video_stream_set_rtcp_information(VideoStream *st, const char *cname, const char *tool) {
	media_stream_set_rtcp_information(&st->ms, cname, tool);
}
/*
 * returns current MSWebCam for a given stream
 * */
MS2_PUBLIC const MSWebCam *video_stream_get_camera(const VideoStream *stream);

/**
 * Returns the current video stream source filter. Be careful, this source will be
 * destroyed if the stream is stopped.
 * @return current stream source
 */
MS2_PUBLIC MSFilter *video_stream_get_source_filter(const VideoStream *stream);

MS2_PUBLIC void video_preview_stream_change_camera(VideoStream *stream, MSWebCam *cam);
MS2_PUBLIC void video_stream_change_camera(VideoStream *stream, MSWebCam *cam);

MS2_PUBLIC void video_stream_change_camera_skip_bitrate(VideoStream *stream, MSWebCam *cam);

/**
 * @brief This functions changes the source filter for the passed video stream.
 * @details This is quite the same function as \ref video_stream_change_camera, but this one
 * allows you to pass the source filter that is created for the camera and reuse it. This gives you the
 * ability to switch rapidly between two streams, whereas re-creating them each time would be
 * costly (especially with webcams).
 *
 * @note Since the \ref video_stream_stop() will automatically destroy the source, it is
 *		advised that you use \ref video_stream_stop_keep_source() instead, so that you
 *		can manually destroy the source filters after the stream is stopped.
 *
 * Example usage:
 *
 *		video_stream_start(stream, profile, [...], noWebcamDevice);
 *		// We manage the sources for the stream ourselves:
 *		MSFilter* noWebCamFilter = video_stream_get_source_filter(stream);
 *		MSFilter* frontCamFilter = ms_web_cam_create_reader(frontCamDevice);
 *
 * 		sleep(1);
 * 		video_stream_change_source_filter(stream, frontCamDevice, frontCamFilter, TRUE); // will keep the previous
 *filter sleep(1); video_stream_change_source_filter(stream, noWebcamDevice, noWebCamFilter, TRUE); // keep the previous
 *filter
 *
 *		sleep(1)
 *		video_stream_stop_keep_source(stream);
 *		ms_filter_destroy(noWebCamFilter);
 *		ms_filter_destroy(frontCamFilter);
 *
 *
 * @param stream the video stream to modify
 * @param cam the camera that you want to set as the new source
 * @param cam_filter the filter for this camera. It can be obtained with ms_web_cam_create_reader(cam)
 * @return the previous source if keep_previous_source is TRUE, otherwise NULL
 */
MS2_PUBLIC MSFilter *
video_stream_change_source_filter(VideoStream *stream, MSWebCam *cam, MSFilter *filter, bool_t keep_previous_source);

/**
 * @brief This function forwards the video from the source stream to the specifed one.
 * @details This will connect the received video stream from source and will pass it to current stream
 * to send. Using \ref video_stream_change_camera will stop the forwarding and use the provided camera.
 * If the source stream is stopped before this stream, a call to \ref video_stream_change_camera is also
 * required in order to restart the graph.
 * @param stream the video stream to modify
 * @param source the video stream that will be used as a source
 */
MS2_PUBLIC void video_stream_forward_source_stream(VideoStream *stream, VideoStream *source);

/**
 * @brief This is the same function as \ref video_stream_change_source_filter() called with keep_source=1, but
 *  the new filter will be created from the MSWebcam that is passed as argument.
 *
 *  @param stream the video stream
 *  @param cam the MSWebcam from which the new source filter should be created.
 *  @return the previous source filter
 */
MS2_PUBLIC MSFilter *video_stream_change_camera_keep_previous_source(VideoStream *stream, MSWebCam *cam);

/* Calling video_stream_set_sent_video_size() or changing the bitrate value in the used PayloadType during a stream is
running does nothing. The following functions allow to take into account new parameters by redrawing the sending graph*/
MS2_PUBLIC void video_preview_stream_update_video_params(VideoStream *stream);
MS2_PUBLIC void video_stream_update_video_params(VideoStream *stream);
/*function to call periodically to handle various events */
MS2_PUBLIC void video_stream_iterate(VideoStream *stream);

/*
 * Assign a specific callback to process PLI, SLI, FIR received by RTCP.
 * When set to NULL, or not assigned, the default behavior is to target these commands to the video encoder.
 */
MS2_PUBLIC void
video_stream_set_encoder_control_callback(VideoStream *stream, VideoStreamEncoderControlCb cb, void *user_data);

/**
 * Asks the video stream to send a Full-Intra Request.
 * @param[in] stream The videostream object.
 */
MS2_PUBLIC void video_stream_send_fir(VideoStream *stream);

/**
 * Asks the video stream to generate a Video Fast Update (generally after receiving a Full-Intra Request.
 * @param[in] stream The videostream object.
 */
MS2_PUBLIC void video_stream_send_vfu(VideoStream *stream);

MS2_PUBLIC void video_stream_stop(VideoStream *stream);

/**
 * Stop the video stream, but does not destroy the source of the video. This function
 * can be use in conjunction with \ref video_stream_change_source_filter() to allow
 * manual management of the source filters for a video stream.
 * @param stream the stream to stop
 * @return returns the source of the video stream, which you should manually destroy when appropriate.
 */
MS2_PUBLIC MSFilter *video_stream_stop_keep_source(VideoStream *stream);

MS2_PUBLIC bool_t video_stream_started(VideoStream *stream);

/**
 * Try to set the size of the video that is sent. Since this relies also on the
 * bitrate specified, make sure to set the payload bitrate accordingly with
 * rtp_profile_get_payload and normal_bitrate value otherwise the best
 * possible resolution will be taken instead of the requested one.
 * @param[in] stream The videostream for which to get the sent video size.
 * @param[in] vsize The sent video size wished.
 */
MS2_PUBLIC void video_stream_set_sent_video_size(VideoStream *stream, MSVideoSize vsize);

/**
 * Gets the size of the video that is sent.
 * @param[in] stream The videostream for which to get the sent video size.
 * @return The sent video size or MS_VIDEO_SIZE_UNKNOWN if not available.
 */
MS2_PUBLIC MSVideoSize video_stream_get_sent_video_size(const VideoStream *stream);

/**
 * Gets the size of the video that is received.
 * @param[in] stream The videostream for which to get the received video size.
 * @return The received video size or MS_VIDEO_SIZE_UNKNOWN if not available.
 */
MS2_PUBLIC MSVideoSize video_stream_get_received_video_size(const VideoStream *stream);

/**
 * Gets the framerate of the video that is sent.
 * @param[in] stream The videostream.
 * @return The actual framerate, 0 if not available..
 */
MS2_PUBLIC float video_stream_get_sent_framerate(const VideoStream *stream);

/**
 * Gets the framerate of the video that is received.
 * @param[in] stream The videostream.
 * @return The received framerate or 0 if not available.
 */
MS2_PUBLIC float video_stream_get_received_framerate(const VideoStream *stream);

MS2_PUBLIC void video_stream_enable_self_view(VideoStream *stream, bool_t val);
MS2_PUBLIC void *video_stream_create_native_window_id(VideoStream *stream, void *context);
MS2_PUBLIC void *video_stream_get_native_window_id(VideoStream *stream);
MS2_PUBLIC void video_stream_set_native_window_id(VideoStream *stream, void *id);
MS2_PUBLIC void *video_stream_create_native_preview_window_id(VideoStream *stream, void *context);
MS2_PUBLIC void video_stream_set_native_preview_window_id(VideoStream *stream, void *id);
MS2_PUBLIC void *video_stream_get_native_preview_window_id(VideoStream *stream);
MS2_PUBLIC void *video_stream_get_video_source_descriptor(VideoStream *stream);
MS2_PUBLIC void video_stream_use_preview_video_window(VideoStream *stream, bool_t yesno);
MS2_PUBLIC void video_stream_set_device_rotation(VideoStream *stream, int orientation);
MS2_PUBLIC void video_stream_show_video(VideoStream *stream, bool_t show);
MS2_PUBLIC void video_stream_set_freeze_on_error(VideoStream *stream, bool_t yesno);
MS2_PUBLIC void video_stream_set_display_mode(VideoStream *stream, MSVideoDisplayMode mode);
MS2_PUBLIC MSVideoDisplayMode video_stream_get_display_mode(VideoStream *stream);
MS2_PUBLIC void video_stream_set_preview_display_mode(VideoStream *stream, MSVideoDisplayMode mode);
MS2_PUBLIC MSVideoDisplayMode video_stream_get_preview_display_mode(VideoStream *stream);
MS2_PUBLIC void video_stream_set_fallback_to_dummy_codec(VideoStream *stream, bool_t yesno);

/**
 * @brief Gets the camera sensor rotation.
 *
 * This is needed on some mobile platforms to get the number of degrees the camera sensor
 * is rotated relative to the screen.
 *
 * @param stream The video stream related to the operation
 * @return The camera sensor rotation in degrees (0 to 360) or -1 if it could not be retrieved
 */
MS2_PUBLIC int video_stream_get_camera_sensor_rotation(VideoStream *stream);

/*provided for compatibility, use media_stream_set_direction() instead */
MS2_PUBLIC int video_stream_recv_only_start(
    VideoStream *videostream, RtpProfile *profile, const char *addr, int port, int used_pt, int jitt_comp);
MS2_PUBLIC int video_stream_send_only_start(VideoStream *videostream,
                                            RtpProfile *profile,
                                            const char *addr,
                                            int port,
                                            int rtcp_port,
                                            int used_pt,
                                            int jitt_comp,
                                            MSWebCam *device);
MS2_PUBLIC void video_stream_recv_only_stop(VideoStream *vs);
MS2_PUBLIC void video_stream_send_only_stop(VideoStream *vs);

/* enable ZRTP on the video stream using information from the audio stream */
MS2_PUBLIC void video_stream_enable_zrtp(VideoStream *vstream, AudioStream *astream);
MS2_PUBLIC void video_stream_start_zrtp(VideoStream *stream);

/* enable SRTP on the video stream */
static MS2_INLINE bool_t video_stream_enable_strp(VideoStream *stream,
                                                  MSCryptoSuite suite,
                                                  const char *snd_key,
                                                  const char *rcv_key) {
	return media_stream_enable_srtp(&stream->ms, suite, snd_key, rcv_key);
}

/* if enabled, the display filter will internaly rotate the video, according to the device orientation */
MS2_PUBLIC void video_stream_enable_display_filter_auto_rotate(VideoStream *stream, bool_t enable);

/* retrieve RTP statistics*/
static MS2_INLINE void video_stream_get_local_rtp_stats(VideoStream *stream, rtp_stats_t *stats) {
	media_stream_get_local_rtp_stats(&stream->ms, stats);
}

static MS2_INLINE int video_stream_set_dscp(VideoStream *stream, int dscp) {
	return media_stream_set_dscp(&stream->ms, dscp);
}

/**
 * Gets the RTP session of a video stream.
 * @param[in] stream #MediaStream object.
 * @return The RTP session of the video stream.
 */
static MS2_INLINE RtpSession *video_stream_get_rtp_session(const VideoStream *stream) {
	return media_stream_get_rtp_session(&stream->ms);
}

/**
 * Ask the video stream whether a decoding error should be reported (eg. to send a VFU request).
 * @param[in] stream The VideoStream object.
 * @param[in] ms The minimum interval in milliseconds between to decoding error report.
 * @return TRUE if the decoding error should be reported, FALSE otherwise.
 */
MS2_PUBLIC bool_t video_stream_is_decoding_error_to_be_reported(VideoStream *stream, uint32_t ms);

/**
 * Tell the video stream that a decoding error has been reported.
 * @param[in] stream The VideoStream object.
 */
MS2_PUBLIC void video_stream_decoding_error_reported(VideoStream *stream);

/**
 * Tell the video stream that a decoding error has been recovered so that new decoding can be reported sooner.
 * @param[in] stream The VideoStream object.
 */
MS2_PUBLIC void video_stream_decoding_error_recovered(VideoStream *stream);

/**
 * Force a resolution for the preview.
 * @param[in] stream The VideoStream object.
 * @param[in] vsize video resolution.
 **/
MS2_PUBLIC void video_stream_set_preview_size(VideoStream *stream, MSVideoSize vsize);

/**
 * Force a resolution for the preview.
 * @param[in] stream The VideoStream object.
 * @param[in] fps the frame rate in frame/seconds. A value of zero means "use encoder default value".
 **/
MS2_PUBLIC void video_stream_set_fps(VideoStream *stream, float fps);

/**
 * Link the audio stream with an existing video stream.
 * This is necessary to enable recording of audio & video into a multimedia file.
 */
MS2_PUBLIC void audio_stream_link_video(AudioStream *stream, VideoStream *video);

/**
 * Unlink the audio stream from the video stream.
 * This must be done if the video stream is about to be stopped.
 **/
MS2_PUBLIC void audio_stream_unlink_video(AudioStream *stream, VideoStream *video);

/**
 * Set a video preset to be used for the video stream.
 * @param[in] stream VideoStream object
 * @param[in] preset The name of the video preset to be used.
 */
MS2_PUBLIC void video_stream_use_video_preset(VideoStream *stream, const char *preset);

/**
 * Returns the name of the video preset used for the video stream.
 * @param[in] stream VideoStream object
 */
MS2_PUBLIC const char *video_stream_get_video_preset(VideoStream *stream);

/**
 * Sets the header extension id for frame marking.
 * This has to be called before starting the video stream.
 *
 * @param stream the video stream
 * @param extension_id the extension id
 */
MS2_PUBLIC void video_stream_set_frame_marking_extension_id(VideoStream *stream, int extension_id);

/**
 * Open a player to play a video file (mkv) to remote end.
 * The player is returned as a MSFilter so that application can make usual player controls on it using the
 *MSPlayerInterface.
 **/
MS2_PUBLIC MSFilter *video_stream_open_remote_play(VideoStream *stream, const char *filename);

MS2_PUBLIC void video_stream_close_remote_play(VideoStream *stream);

/**
 * Open a recorder to record the video coming from remote end into a mkv file.
 * This must be done before the stream is started.
 **/
MS2_PUBLIC MSFilter *video_stream_open_remote_record(VideoStream *stream, const char *filename);

MS2_PUBLIC void video_stream_close_remote_record(VideoStream *stream);

MS2_PUBLIC void video_stream_enable_retransmission_on_nack(VideoStream *stream, bool_t enable);
MS2_PUBLIC void video_stream_set_retransmission_on_nack_max_packet(VideoStream *stream, unsigned int max);

/**
 * Sets the max size the sent video stream can reach.
 *
 * @param stream the video stream
 * @param max the max size the sent videostream can reach. 0x0 will remove this limitation.
 */
MS2_PUBLIC void video_stream_set_sent_video_size_max(VideoStream *stream, MSVideoSize max);

MS2_PUBLIC void
video_stream_set_csrc_changed_callback(VideoStream *stream, VideoStreamCsrcChangedCb cb, void *user_pointer);

/**
 * Screen sharing API to know if local (me) shares its screen.
 **/

MS2_PUBLIC bool_t video_stream_local_screen_sharing_enabled(VideoStream *stream);
MS2_PUBLIC void video_stream_enable_local_screen_sharing(VideoStream *stream, bool_t enable);

/**
 * Small API to display a local preview window.
 **/

typedef VideoStream VideoPreview;

MS2_PUBLIC VideoPreview *video_preview_new(MSFactory *factory);
#define video_preview_set_event_callback(p, c, u) video_stream_set_event_callback(p, c, u)
#define video_preview_set_display_callback(p, c, u) video_stream_set_display_callback(p, c, u)
#define video_preview_set_size(p, s) video_stream_set_sent_video_size(p, s)
#define video_preview_set_display_filter_name(p, dt) video_stream_set_display_filter_name(p, dt)
#define video_preview_create_native_window_id(p, ctx) video_stream_create_native_preview_window_id(p, ctx)
#define video_preview_set_native_window_id(p, id) video_stream_set_native_preview_window_id(p, id)
#define video_preview_get_native_window_id(p) video_stream_get_native_preview_window_id(p)
#define video_preview_set_fps(p, fps) video_stream_set_fps((VideoStream *)p, fps)
#define video_preview_set_device_rotation(p, r) video_stream_set_device_rotation(p, r)
MS2_PUBLIC void video_preview_start(VideoPreview *stream, MSWebCam *device);
MS2_PUBLIC void video_preview_enable_qrcode(VideoPreview *stream, bool_t enable);
MS2_PUBLIC void video_preview_set_decode_rect(VideoPreview *stream, MSRect rect);
MS2_PUBLIC bool_t video_preview_qrcode_enabled(VideoPreview *stream);
MS2_PUBLIC MSVideoSize video_preview_get_current_size(VideoPreview *stream);
MS2_PUBLIC void video_preview_stop(VideoPreview *stream);
MS2_PUBLIC void video_preview_change_camera(VideoPreview *stream, MSWebCam *cam);
MS2_PUBLIC void video_preview_update_video_params(VideoPreview *stream);

/**
 * Stops the video preview graph but keep the source filter for reuse.
 * This is useful when transitioning from a preview-only to a duplex video.
 * The filter needs to be passed to the #video_stream_start_with_source function,
 * otherwise you should destroy it.
 * @param[in] stream VideoPreview object
 * @return The source filter to be passed to the #video_stream_start_with_source function.
 */
MS2_PUBLIC MSFilter *video_preview_stop_reuse_source(VideoPreview *stream);

/*
 * Returns the web cam descriptor for the mire kind of camera.
 **/
MS2_PUBLIC MSWebCamDesc *ms_mire_webcam_desc_get(void);

/**
 * Create an RTP session for duplex communication.
 * @param[in] local_ip The local IP to bind the RTP and RTCP sockets to.
 * @param[in] local_rtp_port The local port to bind the RTP socket to.
 * @param[in] local_rtcp_port The local port to bind the RTCP socket to.
 */
MS2_PUBLIC RtpSession *ms_create_duplex_rtp_session(const char *local_ip, int loc_rtp_port, int loc_rtcp_port, int mtu);

/**
 * Enable/Disable the active speaker mode for the stream.
 * @param[in] stream The stream.
 * @param[in] enable TRUE to enable active speaker mode, FALSE otherwise.
 */
MS2_PUBLIC void video_stream_enable_active_speaker_mode(VideoStream *stream, bool_t enable);

/**
 * @}
 **/

/**
 * @addtogroup text_stream_api
 * @{
 **/

struct _TextStream {
	MediaStream ms;
	MSFilter *rttsource;
	MSFilter *rttsink;
	int pt_t140;
	int pt_red;
};

typedef struct _TextStream TextStream;

/**
 * Creates a TextStream object listening on a RTP port.
 * @param loc_rtp_port the local UDP port to listen for RTP packets.
 * @param loc_rtcp_port the local UDP port to listen for RTCP packets
 * @param ipv6 TRUE if ipv6 must be used.
 * @param factory
 * @return a new TextStream.
 **/
MS2_PUBLIC TextStream *text_stream_new(MSFactory *factory, int loc_rtp_port, int loc_rtcp_port, bool_t ipv6);

/**
 * Creates a TextStream object from initialized MSMediaStreamSessions.
 * @param sessions the MSMediaStreamSessions
 * @param factory
 * @return a new TextStream
 **/
MS2_PUBLIC TextStream *text_stream_new_with_sessions(MSFactory *factory, const MSMediaStreamSessions *sessions);

/**
 * Creates a TextStream object listening on a RTP port for a dedicated address.
 * @param loc_ip the local ip to listen for RTP packets. Can be ::, O.O.O.O or any ip4/6 addresses
 * @param [in] loc_rtp_port the local UDP port to listen for RTP packets.
 * @param [in] loc_rtcp_port the local UDP port to listen for RTCP packets
 * @param factory
 * @return a new TextStream.
 **/
MS2_PUBLIC TextStream *text_stream_new2(MSFactory *factory, const char *ip, int loc_rtp_port, int loc_rtcp_port);

/**
 * Starts a text stream.
 *
 * @param[in] stream TextStream object previously created with text_stream_new().
 * @param[in] profile RtpProfile object holding the PayloadType that can be used during the text session.
 * @param[in] rem_rtp_addr The remote IP address where to send the text to.
 * @param[in] rem_rtp_port The remote port where to send the text to.
 * @param[in] rem_rtcp_addr The remote IP address for RTCP.
 * @param[in] rem_rtcp_port The remote port for RTCP.
 * @param[in] payload_type The payload type number used to send the text stream. A valid PayloadType must be available
 * at this index in the profile.
 * @param[in] factory
 */
MS2_PUBLIC TextStream *text_stream_start(TextStream *stream,
                                         RtpProfile *profile,
                                         const char *rem_rtp_addr,
                                         int rem_rtp_port,
                                         const char *rem_rtcp_addr,
                                         int rem_rtcp_port,
                                         int payload_type);

/**
 *  Stops the text streaming thread and free everything
 **/
MS2_PUBLIC void text_stream_stop(TextStream *stream);

/**
 * Executes background low priority tasks related to text processing (RTP statistics analysis).
 * It should be called periodically, for example with an interval of 100 ms or so.
 *
 * @param[in] stream TextStream object previously created with text_stream_new().
 */
MS2_PUBLIC void text_stream_iterate(TextStream *stream);

/**
 * Writes a character to stream in UTF-32 format.
 *
 * @param[in] stream TextStream object previously created with text_stream_new().
 * @param[in] i the Char in UTF-32 format.
 **/
MS2_PUBLIC void text_stream_putchar32(TextStream *stream, uint32_t i);

MS2_PUBLIC void text_stream_prepare_text(TextStream *stream);
MS2_PUBLIC void text_stream_unprepare_text(TextStream *stream);

/**
 * @}
 **/

#ifdef __cplusplus
}
#endif

#endif
