webrtc_m130/media/base/media_engine.h
Mirko Bonadei f5ecb5f22e Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""
This reverts commit 9cad4dccc96f09d76dce3766a076052df7d75ab8.

Reason for revert: Breaks downstream tests.

Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive video codecs"""
> 
> This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c
> 
> This CL lands all code except the code that activates the change,
> see media/engine/webrtc_video_engine.cc
> Once downstream projects are fixed, there will be a one-line change to
> activate the change to distinguish between send and receive video codecs.
> 
> Original change's description:
> > Reland "Reland "Distinguish between send and receive video codecs""
> >
> > This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> > >
> > > Reason for revert: Downstream test updated.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive video codecs""
> > > >
> > > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > > >
> > > > Reason for revert: Breaks another downstream test.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > > >
> > > > > Reason for revert: Downstream tests have been updated.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive video codecs"
> > > > > >
> > > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > > >
> > > > > > Reason for revert: Breaks downstream test.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive video codecs
> > > > > > >
> > > > > > > Even though send and receive codecs are the same,
> > > > > > > they might have different support in HW.
> > > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > > track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30079}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30097}
> >
> > Bug: chromium:1029737
> > Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30120}
> 
> Bug: chromium:1029737
> Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30219}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: chromium:1029737
Change-Id: I377f82866e56862f57383f96a3b96719344eef9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30225}
2020-01-13 09:03:37 +00:00

173 lines
5.6 KiB
C++

/*
* Copyright (c) 2004 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.
*/
#ifndef MEDIA_BASE_MEDIA_ENGINE_H_
#define MEDIA_BASE_MEDIA_ENGINE_H_
#include <memory>
#include <string>
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/audio_state.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/video_common.h"
#include "rtc_base/system/file_wrapper.h"
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
class AudioProcessing;
class Call;
} // namespace webrtc
namespace cricket {
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& new_parameters);
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_parameters,
const webrtc::RtpParameters& new_parameters);
struct RtpCapabilities {
RtpCapabilities();
~RtpCapabilities();
std::vector<webrtc::RtpExtension> header_extensions;
};
class VoiceEngineInterface {
public:
VoiceEngineInterface() = default;
virtual ~VoiceEngineInterface() = default;
RTC_DISALLOW_COPY_AND_ASSIGN(VoiceEngineInterface);
// Initialization
// Starts the engine.
virtual void Init() = 0;
// TODO(solenberg): Remove once VoE API refactoring is done.
virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
// MediaChannel creation
// Creates a voice media channel. Returns NULL on failure.
virtual VoiceMediaChannel* CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) = 0;
virtual const std::vector<AudioCodec>& send_codecs() const = 0;
virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
virtual RtpCapabilities GetCapabilities() const = 0;
// Starts AEC dump using existing file, a maximum file size in bytes can be
// specified. Logging is stopped just before the size limit is exceeded.
// If max_size_bytes is set to a value <= 0, no limit will be used.
virtual bool StartAecDump(webrtc::FileWrapper file,
int64_t max_size_bytes) = 0;
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
};
class VideoEngineInterface {
public:
VideoEngineInterface() = default;
virtual ~VideoEngineInterface() = default;
RTC_DISALLOW_COPY_AND_ASSIGN(VideoEngineInterface);
// Creates a video media channel, paired with the specified voice channel.
// Returns NULL on failure.
virtual VideoMediaChannel* CreateMediaChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory*
video_bitrate_allocator_factory) = 0;
virtual std::vector<VideoCodec> codecs() const = 0;
virtual RtpCapabilities GetCapabilities() const = 0;
};
// MediaEngineInterface is an abstraction of a media engine which can be
// subclassed to support different media componentry backends.
// It supports voice and video operations in the same class to facilitate
// proper synchronization between both media types.
class MediaEngineInterface {
public:
virtual ~MediaEngineInterface() {}
// Initialization
// Starts the engine.
virtual bool Init() = 0;
virtual VoiceEngineInterface& voice() = 0;
virtual VideoEngineInterface& video() = 0;
virtual const VoiceEngineInterface& voice() const = 0;
virtual const VideoEngineInterface& video() const = 0;
};
// CompositeMediaEngine constructs a MediaEngine from separate
// voice and video engine classes.
class CompositeMediaEngine : public MediaEngineInterface {
public:
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
~CompositeMediaEngine() override;
bool Init() override;
VoiceEngineInterface& voice() override;
VideoEngineInterface& video() override;
const VoiceEngineInterface& voice() const override;
const VideoEngineInterface& video() const override;
private:
std::unique_ptr<VoiceEngineInterface> voice_engine_;
std::unique_ptr<VideoEngineInterface> video_engine_;
};
enum DataChannelType {
DCT_NONE = 0,
DCT_RTP = 1,
DCT_SCTP = 2,
// Data channel transport over media transport.
DCT_MEDIA_TRANSPORT = 3,
// Data channel transport over datagram transport (with no fallback). This is
// the same behavior as data channel transport over media transport, and is
// usable without DTLS.
DCT_DATA_CHANNEL_TRANSPORT = 4,
// Data channel transport over datagram transport (with SCTP negotiation
// semantics and a fallback to SCTP). Only usable with DTLS.
DCT_DATA_CHANNEL_TRANSPORT_SCTP = 5,
};
class DataEngineInterface {
public:
virtual ~DataEngineInterface() {}
virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
virtual const std::vector<DataCodec>& data_codecs() = 0;
};
webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_ENGINE_H_