| /* |
| * 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_ |