| // Copyright 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #ifndef MEDIA_CAST_CAST_CONFIG_H_ |
| #define MEDIA_CAST_CAST_CONFIG_H_ |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include <string> |
| |
| #include "base/callback.h" |
| #include "base/memory/ref_counted.h" |
| #include "base/memory/unsafe_shared_memory_region.h" |
| #include "base/single_thread_task_runner.h" |
| #include "base/time/time.h" |
| |
| namespace media { |
| class VideoEncodeAccelerator; |
| |
| namespace cast { |
| |
| enum Codec { |
| CODEC_UNKNOWN, |
| CODEC_AUDIO_OPUS, |
| CODEC_AUDIO_PCM16, |
| CODEC_AUDIO_AAC, |
| CODEC_AUDIO_REMOTE, |
| CODEC_VIDEO_FAKE, |
| CODEC_VIDEO_VP8, |
| CODEC_VIDEO_H264, |
| CODEC_VIDEO_REMOTE, |
| CODEC_LAST = CODEC_VIDEO_REMOTE |
| }; |
| |
| // Describes the content being transported over RTP streams. |
| enum class RtpPayloadType { |
| UNKNOWN = -1, |
| |
| // Cast Streaming will encode raw audio frames using one of its available |
| // codec implementations, and transport encoded data in the RTP stream. |
| FIRST = 96, |
| AUDIO_OPUS = 96, |
| AUDIO_AAC = 97, |
| AUDIO_PCM16 = 98, |
| |
| // Audio frame data is not modified, and should be transported reliably and |
| // in-sequence. No assumptions about the data can be made. |
| REMOTE_AUDIO = 99, |
| |
| AUDIO_LAST = REMOTE_AUDIO, |
| |
| // Cast Streaming will encode raw video frames using one of its available |
| // codec implementations, and transport encoded data in the RTP stream. |
| VIDEO_VP8 = 100, |
| VIDEO_H264 = 101, |
| |
| // Video frame data is not modified, and should be transported reliably and |
| // in-sequence. No assumptions about the data can be made. |
| REMOTE_VIDEO = 102, |
| |
| LAST = REMOTE_VIDEO |
| }; |
| |
| // TODO(miu): Eliminate these after moving "default config" into the top-level |
| // media/cast directory. http://crbug.com/530839 |
| enum SuggestedDefaults { |
| // Audio encoder bitrate. Zero means "auto," which asks the encoder to select |
| // a bitrate that dynamically adjusts to the content. Otherwise, a constant |
| // bitrate is used. |
| kDefaultAudioEncoderBitrate = 0, |
| |
| // Suggested default audio sampling rate. |
| kDefaultAudioSamplingRate = 48000, |
| |
| // RTP timebase for media remoting RTP streams. |
| kRemotingRtpTimebase = 90000, |
| |
| // Suggested default maximum video frame rate. |
| kDefaultMaxFrameRate = 30, |
| |
| // End-to-end latency in milliseconds. |
| // |
| // DO NOT USE THIS (400 ms is proven as ideal for general-purpose use). |
| // |
| // TODO(miu): Change to 400, and confirm nothing has broken in later change. |
| // http://crbug.com/530839 |
| kDefaultRtpMaxDelayMs = 100, |
| |
| // Suggested minimum and maximum video bitrates for general-purpose use (up to |
| // 1080p, 30 FPS). |
| kDefaultMinVideoBitrate = 300000, |
| kDefaultMaxVideoBitrate = 5000000, |
| |
| // Minimum and Maximum VP8 quantizer in default configuration. |
| kDefaultMaxQp = 63, |
| kDefaultMinQp = 4, |
| |
| kDefaultMaxCpuSaverQp = 25, |
| |
| // Number of video buffers in default configuration (applies only to certain |
| // external codecs). |
| kDefaultNumberOfVideoBuffers = 1, |
| }; |
| |
| // These parameters are only for video encoders. |
| struct VideoCodecParams { |
| VideoCodecParams(); |
| VideoCodecParams(const VideoCodecParams& other); |
| ~VideoCodecParams(); |
| |
| int max_qp; |
| int min_qp; |
| |
| // The maximum |min_quantizer| set to the encoder when CPU is constrained. |
| // This is a trade-off between higher resolution with lower encoding quality |
| // and lower resolution with higher encoding quality. The set value indicates |
| // the maximum quantizer that the encoder might produce better quality video |
| // at this resolution than lowering resolution with similar CPU usage and |
| // smaller quantizer. The set value has to be between |min_qp| and |max_qp|. |
| // Suggested value range: [4, 30]. It is only used by software VP8 codec. |
| int max_cpu_saver_qp; |
| |
| // This field is used differently by various encoders. |
| // |
| // It defaults to 1. |
| // |
| // For VP8, this field is ignored. |
| // |
| // For H.264 on Mac or iOS, it controls the max number of frames the encoder |
| // may hold before emitting a frame. A larger window may allow higher encoding |
| // efficiency at the cost of latency and memory. Set to 0 to let the encoder |
| // choose a suitable value for the platform and other encoding settings. |
| int max_number_of_video_buffers_used; |
| |
| int number_of_encode_threads; |
| }; |
| |
| struct FrameSenderConfig { |
| FrameSenderConfig(); |
| FrameSenderConfig(const FrameSenderConfig& other); |
| ~FrameSenderConfig(); |
| |
| // The sender's SSRC identifier. |
| uint32_t sender_ssrc; |
| |
| // The receiver's SSRC identifier. |
| uint32_t receiver_ssrc; |
| |
| // 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 should be |
| // set to 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.). |
| base::TimeDelta min_playout_delay; |
| base::TimeDelta max_playout_delay; |
| |
| // Starting playout delay when streaming animated content. |
| base::TimeDelta animated_playout_delay; |
| |
| // RTP payload type enum: Specifies the type/encoding of frame data. |
| RtpPayloadType rtp_payload_type; |
| |
| // If true, use an external HW encoder rather than the built-in |
| // software-based one. |
| bool use_external_encoder; |
| |
| // 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 rtp_timebase; |
| |
| // 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; |
| |
| // For now, only fixed bitrate is used for audio encoding. So for audio, |
| // |max_bitrate| is used, and the other two will be overriden if they are not |
| // equal to |max_bitrate|. |
| int max_bitrate; |
| int min_bitrate; |
| int start_bitrate; |
| |
| double max_frame_rate; |
| |
| // Codec used for the compression of signal data. |
| Codec 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; |
| |
| // These are codec specific parameters for video streams only. |
| VideoCodecParams video_codec_params; |
| }; |
| |
| // TODO(miu): Naming and minor type changes are badly needed in a later CL. |
| struct FrameReceiverConfig { |
| FrameReceiverConfig(); |
| FrameReceiverConfig(const FrameReceiverConfig& other); |
| ~FrameReceiverConfig(); |
| |
| // The receiver's SSRC identifier. |
| uint32_t receiver_ssrc; |
| |
| // The sender's SSRC identifier. |
| uint32_t sender_ssrc; |
| |
| // 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. |
| |
| // RTP payload type enum: Specifies the type/encoding of frame data. |
| RtpPayloadType rtp_payload_type; |
| |
| // 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 rtp_timebase; |
| |
| // 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; |
| |
| // 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. |
| double 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. |
| Codec 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; |
| }; |
| |
| typedef base::OnceCallback<void(scoped_refptr<base::SingleThreadTaskRunner>, |
| std::unique_ptr<media::VideoEncodeAccelerator>)> |
| ReceiveVideoEncodeAcceleratorCallback; |
| typedef base::RepeatingCallback<void(ReceiveVideoEncodeAcceleratorCallback)> |
| CreateVideoEncodeAcceleratorCallback; |
| typedef base::OnceCallback<void(base::UnsafeSharedMemoryRegion)> |
| ReceiveVideoEncodeMemoryCallback; |
| } // namespace cast |
| } // namespace media |
| |
| #endif // MEDIA_CAST_CAST_CONFIG_H_ |