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// 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 .proto was automatically generated. ****
syntax = "proto3";
package cast.media;
import "google/protobuf/duration.proto";
import "google/protobuf/empty.proto";
option optimize_for = LITE_RUNTIME;
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 InitializeRequest {
// Cast session ID.
string cast_session_id = 1;
// Configures how the server should operate.
CastAudioDecoderMode mode = 2;
}
message GetMinimumBufferingDelayResponse {
// The minimum buffering delay in microseconds.
int64 delay_micros = 1;
}
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;
}
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 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 SetVolumeRequest {
// The multiplier is in the range [0.0, 1.0].
float multiplier = 1;
}
message SetPlaybackRateRequest {
// Playback rate greater than 0.
double rate = 1;
}
message GetMediaTimeResponse {
// The current media time that has been rendered.
MediaTime media_time = 1;
}
// 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 CastRuntimeAudioChannel {
// 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 (google.protobuf.Empty);
// 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(google.protobuf.Empty)
returns (GetMinimumBufferingDelayResponse);
// 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 (google.protobuf.Empty);
// 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 SetPlayback(SetPlaybackRateRequest) returns (google.protobuf.Empty);
// 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.
//
// TODO(b/178523159): validate that this isn't a performance bottleneck as a
// non-streaming API. If it is, we should make this a bidirectional stream.
rpc PushBuffer(PushBufferRequest) returns (PushBufferResponse);
// Returns the current media time that has been rendered.
rpc GetMediaTime(google.protobuf.Empty) returns (GetMediaTimeResponse);
}