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Move audio-related MediaTransport interfaces to their own file and target

Browse Source 
Bug: webrtc:9719
Change-Id: I8bef979e4073d51be7cb93d38ee0e2ae22baef0e
Reviewed-on: https://webrtc-review.googlesource.com/c/121942
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26594}
This commit is contained in:
Niels Möller
2019-02-08 00:28:39 +01:00
committed by Commit Bot
parent e12778cb3a
commit ec3b9ffdb0
7 changed files with 199 additions and 123 deletions
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1
api/BUILD.gn
View File

@ -134,6 +134,7 @@ rtc_static_library("libjingle_peerconnection_api") {
"audio_codecs:audio_codecs_api",
"transport:bitrate_settings",
"transport:network_control",
"transport/media:audio_interfaces",
"units:data_rate",
"video:encoded_image",
"video:video_frame",

29
api/media_transport_interface.cc
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@ -29,35 +29,6 @@ MediaTransportSettings& MediaTransportSettings::operator=(
const MediaTransportSettings&) = default;
MediaTransportSettings::~MediaTransportSettings() = default;
MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
int sampling_rate_hz,
int starting_sample_index,
int samples_per_channel,
int sequence_number,
FrameType frame_type,
int payload_type,
std::vector<uint8_t> encoded_data)
: sampling_rate_hz_(sampling_rate_hz),
starting_sample_index_(starting_sample_index),
samples_per_channel_(samples_per_channel),
sequence_number_(sequence_number),
frame_type_(frame_type),
payload_type_(payload_type),
encoded_data_(std::move(encoded_data)) {}
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame() = default;

95
api/media_transport_interface.h
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@ -26,6 +26,7 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_error.h"
#include "api/transport/media/audio_transport.h"
#include "api/units/data_rate.h"
#include "api/video/encoded_image.h"
#include "rtc_base/copy_on_write_buffer.h"
@ -77,89 +78,6 @@ struct MediaTransportSettings final {
RtcEventLog* event_log = nullptr;
};
// Represents encoded audio frame in any encoding (type of encoding is opaque).
// To avoid copying of encoded data use move semantics when passing by value.
class MediaTransportEncodedAudioFrame final {
public:
enum class FrameType {
// Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
kSpeech,
// DTX frame (equivalent to webrtc::kAudioFrameCN).
// DTX frame (equivalent to webrtc::kAudioFrameCN).
kDiscontinuousTransmission,
// TODO(nisse): Mis-spelled version, update users, then delete.
kDiscountinuousTransmission = kDiscontinuousTransmission,
};
MediaTransportEncodedAudioFrame(
// Audio sampling rate, for example 48000.
int sampling_rate_hz,
// Starting sample index of the frame, i.e. how many audio samples were
// before this frame since the beginning of the call or beginning of time
// in one channel (the starting point should not matter for NetEq). In
// WebRTC it is used as a timestamp of the frame.
// TODO(sukhanov): Starting_sample_index is currently adjusted on the
// receiver side in RTP path. Non-RTP implementations should preserve it.
// For NetEq initial offset should not matter so we should consider fixing
// RTP path.
int starting_sample_index,
// Number of audio samples in audio frame in 1 channel.
int samples_per_channel,
// Sequence number of the frame in the order sent, it is currently
// required by NetEq, but we can fix NetEq, because starting_sample_index
// should be enough.
int sequence_number,
// If audio frame is a speech or discontinued transmission.
FrameType frame_type,
// Opaque payload type. In RTP codepath payload type is stored in RTP
// header. In other implementations it should be simply passed through the
// wire -- it's needed for decoder.
int payload_type,
// Vector with opaque encoded data.
std::vector<uint8_t> encoded_data);
~MediaTransportEncodedAudioFrame();
MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
MediaTransportEncodedAudioFrame& operator=(
const MediaTransportEncodedAudioFrame& other);
MediaTransportEncodedAudioFrame& operator=(
MediaTransportEncodedAudioFrame&& other);
MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
// Getters.
int sampling_rate_hz() const { return sampling_rate_hz_; }
int starting_sample_index() const { return starting_sample_index_; }
int samples_per_channel() const { return samples_per_channel_; }
int sequence_number() const { return sequence_number_; }
int payload_type() const { return payload_type_; }
FrameType frame_type() const { return frame_type_; }
rtc::ArrayView<const uint8_t> encoded_data() const { return encoded_data_; }
private:
int sampling_rate_hz_;
int starting_sample_index_;
int samples_per_channel_;
// TODO(sukhanov): Refactor NetEq so we don't need sequence number.
// Having sample_index and samples_per_channel should be enough.
int sequence_number_;
FrameType frame_type_;
int payload_type_;
std::vector<uint8_t> encoded_data_;
};
// Callback to notify about network route changes.
class MediaTransportNetworkChangeCallback {
public:
@ -170,17 +88,6 @@ class MediaTransportNetworkChangeCallback {
const rtc::NetworkRoute& new_network_route) = 0;
};
// Interface for receiving encoded audio frames from MediaTransportInterface
// implementations.
class MediaTransportAudioSinkInterface {
public:
virtual ~MediaTransportAudioSinkInterface() = default;
// Called when new encoded audio frame is received.
virtual void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;
};
// Represents encoded video frame, along with the codec information.
class MediaTransportEncodedVideoFrame final {
public:

20
api/transport/media/BUILD.gn Normal file
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@ -0,0 +1,20 @@
# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../../webrtc.gni")
rtc_source_set("audio_interfaces") {
visibility = [ "*" ]
sources = [
"audio_transport.cc",
"audio_transport.h",
]
deps = [
"../..:array_view",
]
}

3
api/transport/media/OWNERS Normal file
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@ -0,0 +1,3 @@
sukhanov@webrtc.org
psla@webrtc.org
mellem@webrtc.org

54
api/transport/media/audio_transport.cc Normal file
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@ -0,0 +1,54 @@
/*
* Copyright 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This is EXPERIMENTAL interface for media transport.
//
// The goal is to refactor WebRTC code so that audio and video frames
// are sent / received through the media transport interface. This will
// enable different media transport implementations, including QUIC-based
// media transport.
#include <utility>
#include "api/transport/media/audio_transport.h"
namespace webrtc {
MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
int sampling_rate_hz,
int starting_sample_index,
int samples_per_channel,
int sequence_number,
FrameType frame_type,
int payload_type,
std::vector<uint8_t> encoded_data)
: sampling_rate_hz_(sampling_rate_hz),
starting_sample_index_(starting_sample_index),
samples_per_channel_(samples_per_channel),
sequence_number_(sequence_number),
frame_type_(frame_type),
payload_type_(payload_type),
encoded_data_(std::move(encoded_data)) {}
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
MediaTransportEncodedAudioFrame&&) = default;
} // namespace webrtc

120
api/transport/media/audio_transport.h Normal file
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@ -0,0 +1,120 @@
/* Copyright 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This is EXPERIMENTAL interface for media transport.
//
// The goal is to refactor WebRTC code so that audio and video frames
// are sent / received through the media transport interface. This will
// enable different media transport implementations, including QUIC-based
// media transport.
#ifndef API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_
#define API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_
#include <vector>
#include "api/array_view.h"
namespace webrtc {
// Represents encoded audio frame in any encoding (type of encoding is opaque).
// To avoid copying of encoded data use move semantics when passing by value.
class MediaTransportEncodedAudioFrame final {
public:
enum class FrameType {
// Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
kSpeech,
// DTX frame (equivalent to webrtc::kAudioFrameCN).
kDiscontinuousTransmission,
// TODO(nisse): Mis-spelled version, update users, then delete.
kDiscountinuousTransmission = kDiscontinuousTransmission,
};
MediaTransportEncodedAudioFrame(
// Audio sampling rate, for example 48000.
int sampling_rate_hz,
// Starting sample index of the frame, i.e. how many audio samples were
// before this frame since the beginning of the call or beginning of time
// in one channel (the starting point should not matter for NetEq). In
// WebRTC it is used as a timestamp of the frame.
// TODO(sukhanov): Starting_sample_index is currently adjusted on the
// receiver side in RTP path. Non-RTP implementations should preserve it.
// For NetEq initial offset should not matter so we should consider fixing
// RTP path.
int starting_sample_index,
// Number of audio samples in audio frame in 1 channel.
int samples_per_channel,
// Sequence number of the frame in the order sent, it is currently
// required by NetEq, but we can fix NetEq, because starting_sample_index
// should be enough.
int sequence_number,
// If audio frame is a speech or discontinued transmission.
FrameType frame_type,
// Opaque payload type. In RTP codepath payload type is stored in RTP
// header. In other implementations it should be simply passed through the
// wire -- it's needed for decoder.
int payload_type,
// Vector with opaque encoded data.
std::vector<uint8_t> encoded_data);
~MediaTransportEncodedAudioFrame();
MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
MediaTransportEncodedAudioFrame& operator=(
const MediaTransportEncodedAudioFrame& other);
MediaTransportEncodedAudioFrame& operator=(
MediaTransportEncodedAudioFrame&& other);
MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
// Getters.
int sampling_rate_hz() const { return sampling_rate_hz_; }
int starting_sample_index() const { return starting_sample_index_; }
int samples_per_channel() const { return samples_per_channel_; }
int sequence_number() const { return sequence_number_; }
int payload_type() const { return payload_type_; }
FrameType frame_type() const { return frame_type_; }
rtc::ArrayView<const uint8_t> encoded_data() const { return encoded_data_; }
private:
int sampling_rate_hz_;
int starting_sample_index_;
int samples_per_channel_;
// TODO(sukhanov): Refactor NetEq so we don't need sequence number.
// Having sample_index and samples_per_channel should be enough.
int sequence_number_;
FrameType frame_type_;
int payload_type_;
std::vector<uint8_t> encoded_data_;
};
// Interface for receiving encoded audio frames from MediaTransportInterface
// implementations.
class MediaTransportAudioSinkInterface {
public:
virtual ~MediaTransportAudioSinkInterface() = default;
// Called when new encoded audio frame is received.
virtual void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;
};
} // namespace webrtc
#endif // API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_