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Diffstat (limited to 'cast/cast_core/api/runtime/cast_audio_channel_service.proto')
| -rw-r--r-- | cast/cast_core/api/runtime/cast_audio_channel_service.proto | 334 |
1 files changed, 334 insertions, 0 deletions
diff --git a/cast/cast_core/api/runtime/cast_audio_channel_service.proto b/cast/cast_core/api/runtime/cast_audio_channel_service.proto new file mode 100644 index 00000000..61c9c658 --- /dev/null +++ b/cast/cast_core/api/runtime/cast_audio_channel_service.proto @@ -0,0 +1,334 @@ +// Copyright 2021 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. +// +// **** DO NOT EDIT - this file was automatically generated. **** +syntax = "proto3"; + +package cast.media; + +import "google/protobuf/duration.proto"; + +option optimize_for = LITE_RUNTIME; + +// Cast audio service hosted by Cast Core. +// +// It defines a state machine with the following states: +// - Uninitialized +// - Playing +// - Stopped +// - Paused +// +// Note that the received ordering between different RPC calls is not +// guaranteed to match the sent order. +service CastAudioChannelService { + // Initializes the service and places the pipeline into the 'Stopped' state. + // This must be the first call received by the server, and no other calls + // may be sent prior to receiving this call's response. + rpc Initialize(InitializeRequest) returns (InitializeResponse); + + // Returns the minimum buffering delay (min_delay) required by Cast. This is + // a constant value and only needs to be queried once for each service. + // During a StartRequest or ResumeRequest, the system timestamp must be + // greater than this delay and the current time in order for the buffer to be + // successfully rendered on remote devices. + rpc GetMinimumBufferDelay(GetMinimumBufferDelayRequest) + returns (GetMinimumBufferDelayResponse); + + // Update the pipeline state. + // + // StartRequest: + // Places pipeline into 'Playing' state. Playback will start at the + // specified buffer and system timestamp. + // + // May only be called in the 'Stopped' state, and following this call the + // state machine will be in the 'Playing' state. + // + // StopRequest + // Stops media playback and drops all pushed buffers which have not yet been + // played. + // + // May only be called in the 'Playing' or 'Paused' states, and following + // this call the state machine will be in the 'Stopped' state. + // + // PauseRequest + // Pauses media playback. + // + // May only be called in the 'Playing' state, and following this call the + // state machine will be in the 'Paused' state. + // + // ResumeRequest + // Resumes media playback at the specified buffer and system timestamp. + // + // May only be called in the 'Paused' state, and following this call the + // state machine will be in the 'Playing'' state. + // + // TimestampUpdateRequest + // Sends a timestamp update for a specified buffer for audio + // synchronization. This should be called when operating in + // CAST_AUDIO_DECODER_MODE_MULTIROOM_ONLY when the runtime has detected a + // discrepancy in the system clock or pipeline delay from the original + // playback schedule. See example below: + // + // Assume all buffers have duration of 100us. + // + // StartRequest(id=1, system_timestamp=0); + // -> Cast expects id=1 to play at 0, id=2 at 100us, id=3 at 200 us... + // + // TimestampUpdateRequest(id=4, system_timestamp=405us); + // -> Cast expects id=4 to play at 405, id=5 at 505us, id=6 at 605 us... + // + // May be called from any state. + // + // A state transition may only occur after a successful PushBuffer() + // call has been made with a valid configuration. + rpc StateChange(StateChangeRequest) returns (StateChangeResponse); + + // Sets the volume multiplier for this audio stream. + // The multiplier is in the range [0.0, 1.0]. If not called, a default + // multiplier of 1.0 is assumed. + // + // May be called in any state, and following this call the state machine + // will be in the same state. + rpc SetVolume(SetVolumeRequest) returns (SetVolumeResponse); + + // Sets the playback rate for this audio stream. + // + // May be called in any state, and following this call the state machine + // will be in the same state. + rpc SetPlaybackRate(SetPlaybackRateRequest) returns (SetPlaybackRateResponse); + + // Sends decoded bits and responses to the audio service. The client must + // wait for a response from the server before sending another + // PushBufferRequest. + // + // May only be called in the 'Playing' or 'Paused' states, and following + // this call the state machine will remain the same state. + // + rpc PushBuffer(PushBufferRequest) returns (PushBufferResponse); + + // Returns the current media time that has been rendered. + rpc GetMediaTime(GetMediaTimeRequest) returns (GetMediaTimeResponse); +} + +message InitializeRequest { + // Cast session ID. + string cast_session_id = 1; + + // Configures how the server should operate. + CastAudioDecoderMode mode = 2; +} + +message InitializeResponse {} + +message GetMinimumBufferDelayRequest {} + +message GetMinimumBufferDelayResponse { + // The minimum buffering delay in microseconds. + int64 delay_micros = 1; +} + +message StateChangeRequest { + oneof request { + StartRequest start = 1; + StopRequest stop = 2; + PauseRequest pause = 3; + ResumeRequest resume = 4; + TimestampUpdateRequest timestamp_update = 5; + } +} + +message StateChangeResponse { + // Pipeline state after state change. + PipelineState state = 1; +} + +message SetVolumeRequest { + // The multiplier is in the range [0.0, 1.0]. + float multiplier = 1; +} + +message SetVolumeResponse {} + +message SetPlaybackRateRequest { + // Playback rate greater than 0. + double rate = 1; +} + +message SetPlaybackRateResponse {} + +message PushBufferRequest { + AudioDecoderBuffer buffer = 1; + + // Audio configuration for this buffer and all subsequent buffers. This + // field must be populated for the first request or if there is an audio + // configuration change. + AudioConfiguration audio_config = 2; +} + +message PushBufferResponse { + // The total number of decoded bytes. + int64 decoded_bytes = 1; +} + +message GetMediaTimeRequest {} + +message GetMediaTimeResponse { + // The current media time that has been rendered. + MediaTime media_time = 1; +} + +enum PipelineState { + PIPELINE_STATE_UNINITIALIZED = 0; + PIPELINE_STATE_STOPPED = 1; + PIPELINE_STATE_PLAYING = 2; + PIPELINE_STATE_PAUSED = 3; +} + +enum CastAudioDecoderMode { + // Both multiroom and audio rendering is enabled. + CAST_AUDIO_DECODER_MODE_ALL = 0; + + // Only multiroom is enabled and audio rendering is disabled. This should + // be used if the runtime is taking over responsibility for rendering audio. + CAST_AUDIO_DECODER_MODE_MULTIROOM_ONLY = 1; + + // Only audio rendering is enabled and multiroom is disabled. + CAST_AUDIO_DECODER_MODE_AUDIO_ONLY = 2; +} + +message AudioConfiguration { + enum AudioCodec { + AUDIO_CODEC_UNKNOWN = 0; + AUDIO_CODEC_AAC = 1; + AUDIO_CODEC_MP3 = 2; + AUDIO_CODEC_PCM = 3; + AUDIO_CODEC_PCM_S16BE = 4; + AUDIO_CODEC_VORBIS = 5; + AUDIO_CODEC_OPUS = 6; + AUDIO_CODEC_EAC3 = 7; + AUDIO_CODEC_AC3 = 8; + AUDIO_CODEC_DTS = 9; + AUDIO_CODEC_FLAC = 10; + AUDIO_CODEC_MPEG_H_AUDIO = 11; + } + + enum ChannelLayout { + CHANNEL_LAYOUT_UNSUPPORTED = 0; + + // Front C + CHANNEL_LAYOUT_MONO = 1; + + // Front L, Front R + CHANNEL_LAYOUT_STEREO = 2; + + // Front L, Front R, Front C, LFE, Side L, Side R + CHANNEL_LAYOUT_SURROUND_5_1 = 3; + + // Actual channel layout is specified in the bitstream and the actual + // channel count is unknown at Chromium media pipeline level (useful for + // audio pass-through mode). + CHANNEL_LAYOUT_BITSTREAM = 4; + + // Channels are not explicitly mapped to speakers. + CHANNEL_LAYOUT_DISCRETE = 5; + } + + enum SampleFormat { + SAMPLE_FORMAT_UNKNOWN = 0; + SAMPLE_FORMAT_U8 = 1; // Unsigned 8-bit w/ bias of 128. + SAMPLE_FORMAT_S16 = 2; // Signed 16-bit. + SAMPLE_FORMAT_S32 = 3; // Signed 32-bit. + SAMPLE_FORMAT_F32 = 4; // Float 32-bit. + SAMPLE_FORMAT_PLANAR_S16 = 5; // Signed 16-bit planar. + SAMPLE_FORMAT_PLANAR_F32 = 6; // Float 32-bit planar. + SAMPLE_FORMAT_PLANAR_S32 = 7; // Signed 32-bit planar. + SAMPLE_FORMAT_S24 = 8; // Signed 24-bit. + } + + // Audio codec. + AudioCodec codec = 1; + + // Audio channel layout. + ChannelLayout channel_layout = 2; + + // The format of each audio sample. + SampleFormat sample_format = 3; + + // Number of bytes in each channel. + int64 bytes_per_channel = 4; + + // Number of channels in this audio stream. + int32 channel_number = 5; + + // Number of audio samples per second. + int64 samples_per_second = 6; + + // Extra data buffer for certain codec initialization. + bytes extra_data = 7; +} + +// The data buffer associated with a single frame of audio data. +message AudioDecoderBuffer { + // The PTS of the frame in microseconds. This is a property of the audio frame + // and is used by the receiver to correctly order the audio frames and to + // determine when they should be decoded. + int64 pts_micros = 1; + + // A single frame of audio data as a byte array. + bytes data = 2; + + // Indicates if this is a special frame that indicates the end of the stream. + // If true, functions to access the frame content cannot be called. + bool end_of_stream = 3; + + // Unique identifier. This field should be greater than equal to 0 and + // incremented by one for each PushBuffeRequest. + int64 id = 4; +} + +message MediaTime { + // The currents PTS that has been rendered. + int64 current_pts_micros = 1; + + // The end of stream has been rendered. + bool end_of_stream = 2; + + // Capture time with respect to CLOCK_MONOTONIC_RAW at which the delay + // measurement was taken. + google.protobuf.Duration capture_time = 3; +} + +message TimestampInfo { + // System timestamp with respect to CLOCK_MONOTONIC_RAW at which the + // corresponding buffer is expected to be rendered. + google.protobuf.Duration system_timestamp = 1; + + // AudioDecoderBuffer.id associated with the |system_timestamp|. + int64 buffer_id = 2; +} + +message StartRequest { + // The start presentation timestamp in microseconds. + int64 pts_micros = 1; + + // Timestamp information associated with the request. + // This field is optional and only used when this service is configured + // for CAST_AUDIO_DECODER_MODE_MULTIROOM_ONLY. + TimestampInfo timestamp_info = 2; +} + +message StopRequest {} + +message PauseRequest {} + +message ResumeRequest { + // Timestamp information associated with the request. + // This field is optional and only used when this service is configured + // for CAST_AUDIO_DECODER_MODE_MULTIROOM_ONLY. + TimestampInfo resume_timestamp_info = 1; +} + +message TimestampUpdateRequest { + TimestampInfo timestamp_info = 1; +} |