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/*
* libjingle
* Copyright 2004--2010, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. 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.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``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 AUTHOR 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.
*/
#ifndef TALK_SESSION_PHONE_MEDIACHANNEL_H_
#define TALK_SESSION_PHONE_MEDIACHANNEL_H_
#include <string>
#include <vector>
#include "talk/base/basictypes.h"
#include "talk/base/sigslot.h"
#include "talk/base/socket.h"
#include "talk/base/window.h"
#include "talk/session/phone/codec.h"
// TODO: re-evaluate this include
#include "talk/session/phone/audiomonitor.h"
namespace talk_base {
class Buffer;
}
namespace cricket {
class VideoRenderer;
const int kMinRtpHeaderExtensionId = 1;
const int kMaxRtpHeaderExtensionId = 255;
// A class for playing out soundclips.
class SoundclipMedia {
public:
enum SoundclipFlags {
SF_LOOP = 1,
};
virtual ~SoundclipMedia() {}
// Plays a sound out to the speakers with the given audio stream. The stream
// must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
// on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
// Returns whether it was successful.
virtual bool PlaySound(const char *clip, int len, int flags) = 0;
};
struct RtpHeaderExtension {
RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
std::string uri;
int id;
// TODO: SendRecv direction;
};
enum VoiceMediaChannelOptions {
OPT_CONFERENCE = 0x10000, // tune the audio stream for conference mode
OPT_AGC_MINUS_10DB = 0x80000000, // tune the audio stream for vcs
// with different target levels.
};
enum VideoMediaChannelOptions {
// Increase the output framerate by 2x by interpolating frames
OPT_INTERPOLATE = 0x10000,
// Enable video adaptation due to cpu load.
OPT_CPU_ADAPTATION = 0x20000
};
class MediaChannel : public sigslot::has_slots<> {
public:
class NetworkInterface {
public:
enum SocketType { ST_RTP, ST_RTCP };
virtual bool SendPacket(talk_base::Buffer* packet) = 0;
virtual bool SendRtcp(talk_base::Buffer* packet) = 0;
virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
int option) = 0;
virtual ~NetworkInterface() {}
};
MediaChannel() : network_interface_(NULL) {}
virtual ~MediaChannel() {}
// Gets/sets the abstract inteface class for sending RTP/RTCP data.
NetworkInterface *network_interface() { return network_interface_; }
virtual void SetInterface(NetworkInterface *iface) {
network_interface_ = iface;
}
// Called when a RTP packet is received.
virtual void OnPacketReceived(talk_base::Buffer* packet) = 0;
// Called when a RTCP packet is received.
virtual void OnRtcpReceived(talk_base::Buffer* packet) = 0;
// Sets the SSRC to be used for outgoing data.
virtual void SetSendSsrc(uint32 id) = 0;
// Set the CNAME of RTCP
virtual bool SetRtcpCName(const std::string& cname) = 0;
// Mutes the channel.
virtual bool Mute(bool on) = 0;
// Sets the RTP extension headers and IDs to use when sending RTP.
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
// Sets the rate control to use when sending data.
virtual bool SetSendBandwidth(bool autobw, int bps) = 0;
// Sets the media options to use.
virtual bool SetOptions(int options) = 0;
// TODO: add virtual int GetOptions() = 0;
protected:
NetworkInterface *network_interface_;
};
enum SendFlags {
SEND_NOTHING,
SEND_RINGBACKTONE,
SEND_MICROPHONE
};
struct VoiceSenderInfo {
VoiceSenderInfo()
: ssrc(0),
bytes_sent(0),
packets_sent(0),
packets_lost(0),
fraction_lost(0.0),
ext_seqnum(0),
rtt_ms(0),
jitter_ms(0),
audio_level(0),
echo_delay_median_ms(0),
echo_delay_std_ms(0),
echo_return_loss(0),
echo_return_loss_enhancement(0) {
}
uint32 ssrc;
std::string codec_name;
int bytes_sent;
int packets_sent;
int packets_lost;
float fraction_lost;
int ext_seqnum;
int rtt_ms;
int jitter_ms;
int audio_level;
int echo_delay_median_ms;
int echo_delay_std_ms;
int echo_return_loss;
int echo_return_loss_enhancement;
};
struct VoiceReceiverInfo {
VoiceReceiverInfo()
: ssrc(0),
bytes_rcvd(0),
packets_rcvd(0),
packets_lost(0),
fraction_lost(0.0),
ext_seqnum(0),
jitter_ms(0),
jitter_buffer_ms(0),
jitter_buffer_preferred_ms(0),
delay_estimate_ms(0),
audio_level(0) {
}
uint32 ssrc;
int bytes_rcvd;
int packets_rcvd;
int packets_lost;
float fraction_lost;
int ext_seqnum;
int jitter_ms;
int jitter_buffer_ms;
int jitter_buffer_preferred_ms;
int delay_estimate_ms;
int audio_level;
};
struct VideoSenderInfo {
VideoSenderInfo()
: ssrc(0),
bytes_sent(0),
packets_sent(0),
packets_cached(0),
packets_lost(0),
fraction_lost(0.0),
firs_rcvd(0),
nacks_rcvd(0),
rtt_ms(0),
frame_width(0),
frame_height(0),
framerate_input(0),
framerate_sent(0),
nominal_bitrate(0),
preferred_bitrate(0) {
}
uint32 ssrc;
std::string codec_name;
int bytes_sent;
int packets_sent;
int packets_cached;
int packets_lost;
float fraction_lost;
int firs_rcvd;
int nacks_rcvd;
int rtt_ms;
int frame_width;
int frame_height;
int framerate_input;
int framerate_sent;
int nominal_bitrate;
int preferred_bitrate;
};
struct VideoReceiverInfo {
VideoReceiverInfo()
: ssrc(0),
bytes_rcvd(0),
packets_rcvd(0),
packets_lost(0),
packets_concealed(0),
fraction_lost(0.0),
firs_sent(0),
nacks_sent(0),
frame_width(0),
frame_height(0),
framerate_rcvd(0),
framerate_decoded(0),
framerate_output(0) {
}
uint32 ssrc;
int bytes_rcvd;
// vector<int> layer_bytes_rcvd;
int packets_rcvd;
int packets_lost;
int packets_concealed;
float fraction_lost;
int firs_sent;
int nacks_sent;
int frame_width;
int frame_height;
int framerate_rcvd;
int framerate_decoded;
int framerate_output;
};
struct BandwidthEstimationInfo {
BandwidthEstimationInfo()
: available_send_bandwidth(0),
available_recv_bandwidth(0),
target_enc_bitrate(0),
actual_enc_bitrate(0),
retransmit_bitrate(0),
transmit_bitrate(0),
bucket_delay(0) {
}
int available_send_bandwidth;
int available_recv_bandwidth;
int target_enc_bitrate;
int actual_enc_bitrate;
int retransmit_bitrate;
int transmit_bitrate;
int bucket_delay;
};
struct VoiceMediaInfo {
void Clear() {
senders.clear();
receivers.clear();
}
std::vector<VoiceSenderInfo> senders;
std::vector<VoiceReceiverInfo> receivers;
};
struct VideoMediaInfo {
void Clear() {
senders.clear();
receivers.clear();
bw_estimations.clear();
}
std::vector<VideoSenderInfo> senders;
std::vector<VideoReceiverInfo> receivers;
std::vector<BandwidthEstimationInfo> bw_estimations;
};
class VoiceMediaChannel : public MediaChannel {
public:
enum Error {
ERROR_NONE = 0, // No error.
ERROR_OTHER, // Other errors.
ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open mic.
ERROR_REC_DEVICE_MUTED, // Mic was muted by OS.
ERROR_REC_DEVICE_SILENT, // No background noise picked up.
ERROR_REC_DEVICE_SATURATION, // Mic input is clipping.
ERROR_REC_DEVICE_REMOVED, // Mic was removed while active.
ERROR_REC_RUNTIME_ERROR, // Processing is encountering errors.
ERROR_REC_SRTP_ERROR, // Generic SRTP failure.
ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_REC_TYPING_NOISE_DETECTED, // Typing noise is detected.
ERROR_PLAY_DEVICE_OPEN_FAILED = 200, // Could not open playout.
ERROR_PLAY_DEVICE_MUTED, // Playout muted by OS.
ERROR_PLAY_DEVICE_REMOVED, // Playout removed while active.
ERROR_PLAY_RUNTIME_ERROR, // Errors in voice processing.
ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure.
ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
};
VoiceMediaChannel() {}
virtual ~VoiceMediaChannel() {}
// Sets the codecs/payload types to be used for incoming media.
virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) = 0;
// Sets the codecs/payload types to be used for outgoing media.
virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs) = 0;
// Starts or stops playout of received audio.
virtual bool SetPlayout(bool playout) = 0;
// Starts or stops sending (and potentially capture) of local audio.
virtual bool SetSend(SendFlags flag) = 0;
// Adds a new receive-only stream with the specified SSRC.
virtual bool AddStream(uint32 ssrc) = 0;
// Removes a stream added with AddStream.
virtual bool RemoveStream(uint32 ssrc) = 0;
// Gets current energy levels for all incoming streams.
virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
// Get the current energy level of the stream sent to the speaker.
virtual int GetOutputLevel() = 0;
// Set left and right scale for speaker output volume of the specified ssrc.
virtual bool SetOutputScaling(uint32 ssrc, double left, double right) = 0;
// Get left and right scale for speaker output volume of the specified ssrc.
virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) = 0;
// Specifies a ringback tone to be played during call setup.
virtual bool SetRingbackTone(const char *buf, int len) = 0;
// Plays or stops the aforementioned ringback tone
virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) = 0;
// Sends a out-of-band DTMF signal using the specified event.
virtual bool PressDTMF(int event, bool playout) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
// Gets last reported error for this media channel.
virtual void GetLastMediaError(uint32* ssrc,
VoiceMediaChannel::Error* error) {
ASSERT(error != NULL);
*error = ERROR_NONE;
}
// Signal errors from MediaChannel. Arguments are:
// ssrc(uint32), and error(VoiceMediaChannel::Error).
sigslot::signal2<uint32, VoiceMediaChannel::Error> SignalMediaError;
};
class VideoMediaChannel : public MediaChannel {
public:
enum Error {
ERROR_NONE = 0, // No error.
ERROR_OTHER, // Other errors.
ERROR_REC_DEVICE_OPEN_FAILED = 100, // Could not open camera.
ERROR_REC_DEVICE_NO_DEVICE, // No camera.
ERROR_REC_DEVICE_IN_USE, // Device is in already use.
ERROR_REC_DEVICE_REMOVED, // Device is removed.
ERROR_REC_SRTP_ERROR, // Generic sender SRTP failure.
ERROR_REC_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_PLAY_SRTP_ERROR = 200, // Generic receiver SRTP failure.
ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets.
ERROR_PLAY_SRTP_REPLAY, // Packet replay detected.
};
VideoMediaChannel() { renderer_ = NULL; }
virtual ~VideoMediaChannel() {}
// Sets the codecs/payload types to be used for incoming media.
virtual bool SetRecvCodecs(const std::vector<VideoCodec> &codecs) = 0;
// Sets the codecs/payload types to be used for outgoing media.
virtual bool SetSendCodecs(const std::vector<VideoCodec> &codecs) = 0;
// Starts or stops playout of received video.
virtual bool SetRender(bool render) = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
// Adds a new receive-only stream with the specified SSRC.
virtual bool AddStream(uint32 ssrc, uint32 voice_ssrc) = 0;
// Removes a stream added with AddStream.
virtual bool RemoveStream(uint32 ssrc) = 0;
// Sets the renderer object to be used for the specified stream.
// If SSRC is 0, the renderer is used for the 'default' stream.
virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) = 0;
virtual bool AddScreencast(uint32 ssrc, talk_base::WindowId id) = 0;
virtual bool RemoveScreencast(uint32 ssrc) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VideoMediaInfo* info) = 0;
// Send an intra frame to the receivers.
virtual bool SendIntraFrame() = 0;
// Reuqest each of the remote senders to send an intra frame.
virtual bool RequestIntraFrame() = 0;
// Signals events from the currently active window.
sigslot::signal2<uint32, talk_base::WindowEvent> SignalScreencastWindowEvent;
sigslot::signal2<uint32, Error> SignalMediaError;
protected:
VideoRenderer *renderer_;
};
} // namespace cricket
#endif // TALK_SESSION_PHONE_MEDIACHANNEL_H_