diff options
Diffstat (limited to 'chromium/media/cast/cast_config.h')
| -rw-r--r-- | chromium/media/cast/cast_config.h | 230 |
1 files changed, 81 insertions, 149 deletions
diff --git a/chromium/media/cast/cast_config.h b/chromium/media/cast/cast_config.h index 27cc67e5dae..ea25d6b6cf7 100644 --- a/chromium/media/cast/cast_config.h +++ b/chromium/media/cast/cast_config.h @@ -12,65 +12,53 @@ #include "base/basictypes.h" #include "base/callback.h" #include "base/memory/ref_counted.h" +#include "base/memory/shared_memory.h" +#include "base/single_thread_task_runner.h" #include "media/cast/cast_defines.h" +#include "media/cast/transport/cast_transport_config.h" namespace media { +class VideoEncodeAccelerator; + namespace cast { enum RtcpMode { - kRtcpCompound, // Compound RTCP mode is described by RFC 4585. + kRtcpCompound, // Compound RTCP mode is described by RFC 4585. kRtcpReducedSize, // Reduced-size RTCP mode is described by RFC 5506. }; -enum VideoCodec { - kVp8, - kH264, - kExternalVideo, -}; - -enum AudioCodec { - kOpus, - kPcm16, - kExternalAudio, -}; - +// TODO(miu): Merge AudioSenderConfig and VideoSenderConfig and make their +// naming/documentation consistent with FrameReceiverConfig. struct AudioSenderConfig { AudioSenderConfig(); - uint32 sender_ssrc; + // The sender ssrc is in rtp_config.ssrc. uint32 incoming_feedback_ssrc; int rtcp_interval; std::string rtcp_c_name; RtcpMode rtcp_mode; - int rtp_history_ms; // The time RTP packets are stored for retransmissions. - int rtp_max_delay_ms; - int rtp_payload_type; + transport::RtpConfig rtp_config; bool use_external_encoder; int frequency; int channels; int bitrate; // Set to <= 0 for "auto variable bitrate" (libopus knows best). - AudioCodec codec; - - std::string aes_key; // Binary string of size kAesKeySize. - std::string aes_iv_mask; // Binary string of size kAesKeySize. + transport::AudioCodec codec; }; struct VideoSenderConfig { VideoSenderConfig(); - uint32 sender_ssrc; + // The sender ssrc is in rtp_config.ssrc. uint32 incoming_feedback_ssrc; int rtcp_interval; std::string rtcp_c_name; RtcpMode rtcp_mode; - int rtp_history_ms; // The time RTP packets are stored for retransmissions. - int rtp_max_delay_ms; - int rtp_payload_type; + transport::RtpConfig rtp_config; bool use_external_encoder; int width; // Incoming frames will be scaled to this size. @@ -84,148 +72,92 @@ struct VideoSenderConfig { int min_qp; int max_frame_rate; int max_number_of_video_buffers_used; // Max value depend on codec. - VideoCodec codec; - int number_of_cores; - - std::string aes_key; // Binary string of size kAesKeySize. - std::string aes_iv_mask; // Binary string of size kAesKeySize. + transport::VideoCodec codec; + int number_of_encode_threads; }; -struct AudioReceiverConfig { - AudioReceiverConfig(); - - uint32 feedback_ssrc; - uint32 incoming_ssrc; - - int rtcp_interval; - std::string rtcp_c_name; - RtcpMode rtcp_mode; - - // The time the receiver is prepared to wait for retransmissions. - int rtp_max_delay_ms; - int rtp_payload_type; - - bool use_external_decoder; - int frequency; - int channels; - AudioCodec codec; - - std::string aes_key; // Binary string of size kAesKeySize. - std::string aes_iv_mask; // Binary string of size kAesKeySize. -}; +// TODO(miu): Naming and minor type changes are badly needed in a later CL. +struct FrameReceiverConfig { + FrameReceiverConfig(); + ~FrameReceiverConfig(); -struct VideoReceiverConfig { - VideoReceiverConfig(); + // The receiver's SSRC identifier. + uint32 feedback_ssrc; // TODO(miu): Rename to receiver_ssrc for clarity. - uint32 feedback_ssrc; - uint32 incoming_ssrc; + // The sender's SSRC identifier. + uint32 incoming_ssrc; // TODO(miu): Rename to sender_ssrc for clarity. + // Mean interval (in milliseconds) between RTCP reports. + // TODO(miu): Remove this since it's never not kDefaultRtcpIntervalMs. int rtcp_interval; - std::string rtcp_c_name; - RtcpMode rtcp_mode; - - // The time the receiver is prepared to wait for retransmissions. - int rtp_max_delay_ms; - int rtp_payload_type; - - bool use_external_decoder; - int max_frame_rate; - - // Some HW decoders can not run faster than the frame rate, preventing it - // from catching up after a glitch. - bool decoder_faster_than_max_frame_rate; - VideoCodec codec; - - std::string aes_key; // Binary string of size kAesKeySize. - std::string aes_iv_mask; // Binary string of size kAesKeySize. -}; - -struct EncodedVideoFrame { - EncodedVideoFrame(); - ~EncodedVideoFrame(); - - VideoCodec codec; - bool key_frame; - uint32 frame_id; - uint32 last_referenced_frame_id; - std::string data; -}; - -// DEPRECATED: Do not use in new code. Please migrate existing code to use -// media::AudioBus. -struct PcmAudioFrame { - PcmAudioFrame(); - ~PcmAudioFrame(); - - int channels; // Samples in interleaved stereo format. L0, R0, L1 ,R1 ,... - int frequency; - std::vector<int16> samples; -}; - -struct EncodedAudioFrame { - EncodedAudioFrame(); - ~EncodedAudioFrame(); - - AudioCodec codec; - uint32 frame_id; // Needed to release the frame. - int samples; // Needed send side to advance the RTP timestamp. - // Not used receive side. - // Support for max sampling rate of 48KHz, 2 channels, 100 ms duration. - static const int kMaxNumberOfSamples = 48 * 2 * 100; - std::string data; -}; -typedef std::vector<uint8> Packet; -typedef std::vector<Packet> PacketList; - -class PacketSender { - public: - // All packets to be sent to the network will be delivered via these - // functions. - virtual bool SendPackets(const PacketList& packets) = 0; + // CNAME representing this receiver. + // TODO(miu): Remove this since it should be derived elsewhere (probably in + // the transport layer). + std::string rtcp_c_name; - virtual bool SendPacket(const Packet& packet) = 0; + // Determines amount of detail in RTCP reports. + // TODO(miu): Remove this since it's never anything but kRtcpReducedSize. + RtcpMode rtcp_mode; - virtual ~PacketSender() {} -}; + // The total amount of time between a frame's capture/recording on the sender + // and its playback on the receiver (i.e., shown to a user). This is fixed as + // a value large enough to give the system sufficient time to encode, + // transmit/retransmit, receive, decode, and render; given its run-time + // environment (sender/receiver hardware performance, network conditions, + // etc.). + int rtp_max_delay_ms; // TODO(miu): Change to TimeDelta target_playout_delay. -class PacketReceiver : public base::RefCountedThreadSafe<PacketReceiver> { - public: - // All packets received from the network should be delivered via this - // function. - virtual void ReceivedPacket(const uint8* packet, size_t length, - const base::Closure callback) = 0; + // RTP payload type enum: Specifies the type/encoding of frame data. + int rtp_payload_type; - static void DeletePacket(const uint8* packet); + // RTP timebase: The number of RTP units advanced per one second. For audio, + // this is the sampling rate. For video, by convention, this is 90 kHz. + int frequency; // TODO(miu): Rename to rtp_timebase for clarity. - protected: - virtual ~PacketReceiver() {} + // Number of channels. For audio, this is normally 2. For video, this must + // be 1 as Cast does not have support for stereoscopic video. + int channels; - private: - friend class base::RefCountedThreadSafe<PacketReceiver>; + // The target frame rate. For audio, this is normally 100 (i.e., frames have + // a duration of 10ms each). For video, this is normally 30, but any frame + // rate is supported. + int max_frame_rate; // TODO(miu): Rename to target_frame_rate. + + // Codec used for the compression of signal data. + // TODO(miu): Merge the AudioCodec and VideoCodec enums into one so this union + // is not necessary. + union MergedCodecPlaceholder { + transport::AudioCodec audio; + transport::VideoCodec video; + MergedCodecPlaceholder() : audio(transport::kUnknownAudioCodec) {} + } codec; + + // The AES crypto key and initialization vector. Each of these strings + // contains the data in binary form, of size kAesKeySize. If they are empty + // strings, crypto is not being used. + std::string aes_key; + std::string aes_iv_mask; }; -class VideoEncoderController { - public: - // Inform the encoder about the new target bit rate. - virtual void SetBitRate(int new_bit_rate) = 0; +// import from media::cast::transport +typedef transport::Packet Packet; +typedef transport::PacketList PacketList; - // Inform the encoder to not encode the next frame. - // Note: this setting is sticky and should last until called with false. - virtual void SkipNextFrame(bool skip_next_frame) = 0; +typedef base::Callback<void(CastInitializationStatus)> + CastInitializationCallback; - // Inform the encoder to encode the next frame as a key frame. - virtual void GenerateKeyFrame() = 0; +typedef base::Callback<void(scoped_refptr<base::SingleThreadTaskRunner>, + scoped_ptr<media::VideoEncodeAccelerator>)> + ReceiveVideoEncodeAcceleratorCallback; +typedef base::Callback<void(const ReceiveVideoEncodeAcceleratorCallback&)> + CreateVideoEncodeAcceleratorCallback; - // Inform the encoder to only reference frames older or equal to frame_id; - virtual void LatestFrameIdToReference(uint32 frame_id) = 0; - - // Query the codec about how many frames it has skipped due to slow ACK. - virtual int NumberOfSkippedFrames() const = 0; - - protected: - virtual ~VideoEncoderController() {} -}; +typedef base::Callback<void(scoped_ptr<base::SharedMemory>)> + ReceiveVideoEncodeMemoryCallback; +typedef base::Callback<void(size_t size, + const ReceiveVideoEncodeMemoryCallback&)> + CreateVideoEncodeMemoryCallback; } // namespace cast } // namespace media |