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Moved call.h and most of api/call/* into call/

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BUG=webrtc:6716

Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
This commit is contained in:
ossu
2016-12-07 04:52:58 -08:00
committed by Commit bot
parent 7495c8c3ac
commit f515ab8c3f
56 changed files with 270 additions and 228 deletions
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4
webrtc/api/BUILD.gn
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@ -20,11 +20,7 @@ group("api") {
rtc_source_set("call_api") {
sources = [
"call/audio_receive_stream.h",
"call/audio_send_stream.cc",
"call/audio_send_stream.h",
"call/audio_sink.h",
"call/audio_state.h",
"call/flexfec_receive_stream.h",
]

12
webrtc/api/DEPS
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@ -18,10 +18,22 @@ specific_include_rules = {
"+base/android", # Allowed only for Android tests.
"+webrtc/voice_engine",
],
# The call/call.h exceptions are here only until the peerconnection
# implementation has been moved out of api/. See:
# http://bugs.webrtc.org/5883
"mediacontroller\.cc": [
"+webrtc/call/call.h"
],
"peerconnection\.cc": [
"+webrtc/call/call.h"
],
"peerconnection_jni\.cc": [
"+webrtc/voice_engine",
],
"peerconnectionfactory\.cc": [
"+webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h",
],
"webrtcsession\.cc": [
"+webrtc/call/call.h"
],
}

4
webrtc/api/call/DEPS
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@ -1,4 +0,0 @@
include_rules = [
"+webrtc/modules/audio_coding/codecs",
]

142
webrtc/api/call/audio_receive_stream.h
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@ -1,142 +0,0 @@
/*
* Copyright (c) 2015 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 WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
#define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/call/transport.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioSinkInterface;
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
class AudioReceiveStream {
public:
struct Stats {
uint32_t remote_ssrc = 0;
int64_t bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
uint32_t packets_lost = 0;
float fraction_lost = 0.0f;
std::string codec_name;
rtc::Optional<int> codec_payload_type;
uint32_t ext_seqnum = 0;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
uint32_t jitter_buffer_preferred_ms = 0;
uint32_t delay_estimate_ms = 0;
int32_t audio_level = -1;
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;
float secondary_decoded_rate = 0.0f;
float accelerate_rate = 0.0f;
float preemptive_expand_rate = 0.0f;
int32_t decoding_calls_to_silence_generator = 0;
int32_t decoding_calls_to_neteq = 0;
int32_t decoding_normal = 0;
int32_t decoding_plc = 0;
int32_t decoding_cng = 0;
int32_t decoding_plc_cng = 0;
int32_t decoding_muted_output = 0;
int64_t capture_start_ntp_time_ms = 0;
};
struct Config {
std::string ToString() const;
// Receive-stream specific RTP settings.
struct Rtp {
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
uint32_t remote_ssrc = 0;
// Sender SSRC used for sending RTCP (such as receiver reports).
uint32_t local_ssrc = 0;
// Enable feedback for send side bandwidth estimation.
// See
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
// for details.
bool transport_cc = false;
// See NackConfig for description.
NackConfig nack;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
Transport* rtcp_send_transport = nullptr;
// Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
// level components.
// TODO(solenberg): Remove when VoiceEngine channels are created outside
// of Call.
int voe_channel_id = -1;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video
// stream to one audio stream. Tracked by issue webrtc:4762.
std::string sync_group;
// Decoders for every payload that we can receive. Call owns the
// AudioDecoder instances once the Config is submitted to
// Call::CreateReceiveStream().
// TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
std::map<uint8_t, AudioDecoder*> decoder_map;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
};
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.
virtual void Start() = 0;
// Stops stream activity.
// When a stream is stopped, it can't receive, process or deliver packets.
virtual void Stop() = 0;
virtual Stats GetStats() const = 0;
// Sets an audio sink that receives unmixed audio from the receive stream.
// Ownership of the sink is passed to the stream and can be used by the
// caller to do lifetime management (i.e. when the sink's dtor is called).
// Only one sink can be set and passing a null sink clears an existing one.
// NOTE: Audio must still somehow be pulled through AudioTransport for audio
// to stream through this sink. In practice, this happens if mixed audio
// is being pulled+rendered and/or if audio is being pulled for the purposes
// of feeding to the AEC.
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
// Sets playback gain of the stream, applied when mixing, and thus after it
// is potentially forwarded to any attached AudioSinkInterface implementation.
virtual void SetGain(float gain) = 0;
protected:
virtual ~AudioReceiveStream() {}
};
} // namespace webrtc
#endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_

109
webrtc/api/call/audio_send_stream.cc
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@ -1,109 +0,0 @@
/*
* Copyright (c) 2015 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.
*/
#include "webrtc/api/call/audio_send_stream.h"
#include <string>
namespace {
std::string ToString(const webrtc::CodecInst& codec_inst) {
std::stringstream ss;
ss << "{pltype: " << codec_inst.pltype;
ss << ", plname: \"" << codec_inst.plname << "\"";
ss << ", plfreq: " << codec_inst.plfreq;
ss << ", pacsize: " << codec_inst.pacsize;
ss << ", channels: " << codec_inst.channels;
ss << ", rate: " << codec_inst.rate;
ss << '}';
return ss.str();
}
} // namespace
namespace webrtc {
AudioSendStream::Stats::Stats() = default;
AudioSendStream::Stats::~Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}
AudioSendStream::Config::~Config() = default;
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
ss << ", voe_channel_id: " << voe_channel_id;
ss << ", min_bitrate_bps: " << min_bitrate_bps;
ss << ", max_bitrate_bps: " << max_bitrate_bps;
ss << ", send_codec_spec: " << send_codec_spec.ToString();
ss << '}';
return ss.str();
}
AudioSendStream::Config::Rtp::Rtp() = default;
AudioSendStream::Config::Rtp::~Rtp() = default;
std::string AudioSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrc: " << ssrc;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << ", nack: " << nack.ToString();
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
AudioSendStream::Config::SendCodecSpec::SendCodecSpec() {
webrtc::CodecInst empty_inst = {0};
codec_inst = empty_inst;
codec_inst.pltype = -1;
}
std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
std::stringstream ss;
ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
ss << ", cng_payload_type: " << cng_payload_type;
ss << ", cng_plfreq: " << cng_plfreq;
ss << ", min_ptime: " << min_ptime_ms;
ss << ", max_ptime: " << max_ptime_ms;
ss << ", codec_inst: " << ::ToString(codec_inst);
ss << '}';
return ss.str();
}
bool AudioSendStream::Config::SendCodecSpec::operator==(
const AudioSendStream::Config::SendCodecSpec& rhs) const {
if (nack_enabled == rhs.nack_enabled &&
transport_cc_enabled == rhs.transport_cc_enabled &&
enable_codec_fec == rhs.enable_codec_fec &&
enable_opus_dtx == rhs.enable_opus_dtx &&
opus_max_playback_rate == rhs.opus_max_playback_rate &&
cng_payload_type == rhs.cng_payload_type &&
cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
return true;
}
return false;
}
} // namespace webrtc

145
webrtc/api/call/audio_send_stream.h
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@ -1,145 +0,0 @@
/*
* Copyright (c) 2015 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 WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
#define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/call/transport.h"
#include "webrtc/base/optional.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
class AudioSendStream {
public:
struct Stats {
Stats();
~Stats();
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
int64_t bytes_sent = 0;
int32_t packets_sent = 0;
int32_t packets_lost = -1;
float fraction_lost = -1.0f;
std::string codec_name;
rtc::Optional<int> codec_payload_type;
int32_t ext_seqnum = -1;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
int32_t audio_level = -1;
float aec_quality_min = -1.0f;
int32_t echo_delay_median_ms = -1;
int32_t echo_delay_std_ms = -1;
int32_t echo_return_loss = -100;
int32_t echo_return_loss_enhancement = -100;
float residual_echo_likelihood = -1.0f;
bool typing_noise_detected = false;
};
struct Config {
Config() = delete;
explicit Config(Transport* send_transport);
~Config();
std::string ToString() const;
// Send-stream specific RTP settings.
struct Rtp {
Rtp();
~Rtp();
std::string ToString() const;
// Sender SSRC.
uint32_t ssrc = 0;
// RTP header extensions used for the sent stream.
std::vector<RtpExtension> extensions;
// See NackConfig for description.
NackConfig nack;
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
// Transport for outgoing packets. The transport is expected to exist for
// the entire life of the AudioSendStream and is owned by the API client.
Transport* send_transport = nullptr;
// Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
// components.
// TODO(solenberg): Remove when VoiceEngine channels are created outside
// of Call.
int voe_channel_id = -1;
// Bitrate limits used for variable audio bitrate streams. Set both to -1 to
// disable audio bitrate adaptation.
// Note: This is still an experimental feature and not ready for real usage.
int min_bitrate_bps = -1;
int max_bitrate_bps = -1;
// Defines whether to turn on audio network adaptor, and defines its config
// string.
rtc::Optional<std::string> audio_network_adaptor_config;
struct SendCodecSpec {
SendCodecSpec();
std::string ToString() const;
bool operator==(const SendCodecSpec& rhs) const;
bool operator!=(const SendCodecSpec& rhs) const {
return !(*this == rhs);
}
bool nack_enabled = false;
bool transport_cc_enabled = false;
bool enable_codec_fec = false;
bool enable_opus_dtx = false;
int opus_max_playback_rate = 0;
int cng_payload_type = -1;
int cng_plfreq = -1;
int max_ptime_ms = -1;
int min_ptime_ms = -1;
webrtc::CodecInst codec_inst;
} send_codec_spec;
};
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.
virtual void Start() = 0;
// Stops stream activity.
// When a stream is stopped, it can't receive, process or deliver packets.
virtual void Stop() = 0;
// TODO(solenberg): Make payload_type a config property instead.
virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
int event, int duration_ms) = 0;
virtual void SetMuted(bool muted) = 0;
virtual Stats GetStats() const = 0;
protected:
virtual ~AudioSendStream() {}
};
} // namespace webrtc
#endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_

49
webrtc/api/call/audio_state.h
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@ -1,49 +0,0 @@
/*
* Copyright (c) 2015 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 WEBRTC_API_CALL_AUDIO_STATE_H_
#define WEBRTC_API_CALL_AUDIO_STATE_H_
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
namespace webrtc {
class VoiceEngine;
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public rtc::RefCountInterface {
public:
struct Config {
// VoiceEngine used for audio streams and audio/video synchronization.
// AudioState will tickle the VoE refcount to keep it alive for as long as
// the AudioState itself.
VoiceEngine* voice_engine = nullptr;
// The audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;
};
// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
static rtc::scoped_refptr<AudioState> Create(
const AudioState::Config& config);
virtual ~AudioState() {}
};
} // namespace webrtc
#endif // WEBRTC_API_CALL_AUDIO_STATE_H_

2
webrtc/api/mediacontroller.cc
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@ -15,7 +15,7 @@
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/call.h"
#include "webrtc/call/call.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/media/base/mediachannel.h"

2
webrtc/api/peerconnection.cc
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@ -36,7 +36,7 @@
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/pc/channelmanager.h"

2
webrtc/api/webrtcsession.cc
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@ -30,7 +30,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
#include "webrtc/call/call.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/p2p/base/portallocator.h"