631 lines
22 KiB
C++
631 lines
22 KiB
C++
/*
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* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
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#define MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
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#include <list>
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <tuple>
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#include <vector>
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#include "absl/algorithm/container.h"
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#include "api/call/audio_sink.h"
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#include "media/base/audio_source.h"
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#include "media/base/media_engine.h"
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#include "media/base/rtp_utils.h"
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#include "media/base/stream_params.h"
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#include "media/engine/webrtc_video_engine.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network_route.h"
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using webrtc::RtpExtension;
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namespace cricket {
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class FakeMediaEngine;
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class FakeVideoEngine;
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class FakeVoiceEngine;
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// A common helper class that handles sending and receiving RTP/RTCP packets.
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template <class Base>
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class RtpHelper : public Base {
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public:
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RtpHelper()
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: sending_(false),
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playout_(false),
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fail_set_send_codecs_(false),
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fail_set_recv_codecs_(false),
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send_ssrc_(0),
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ready_to_send_(false),
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transport_overhead_per_packet_(0),
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num_network_route_changes_(0) {}
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virtual ~RtpHelper() = default;
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const std::vector<RtpExtension>& recv_extensions() {
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return recv_extensions_;
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}
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const std::vector<RtpExtension>& send_extensions() {
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return send_extensions_;
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}
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bool sending() const { return sending_; }
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bool playout() const { return playout_; }
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const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
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const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
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bool SendRtp(const void* data,
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size_t len,
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const rtc::PacketOptions& options) {
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if (!sending_) {
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return false;
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}
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rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
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kMaxRtpPacketLen);
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return Base::SendPacket(&packet, options);
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}
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bool SendRtcp(const void* data, size_t len) {
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rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
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kMaxRtpPacketLen);
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return Base::SendRtcp(&packet, rtc::PacketOptions());
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}
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bool CheckRtp(const void* data, size_t len) {
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bool success = !rtp_packets_.empty();
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if (success) {
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std::string packet = rtp_packets_.front();
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rtp_packets_.pop_front();
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success = (packet == std::string(static_cast<const char*>(data), len));
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}
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return success;
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}
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bool CheckRtcp(const void* data, size_t len) {
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bool success = !rtcp_packets_.empty();
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if (success) {
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std::string packet = rtcp_packets_.front();
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rtcp_packets_.pop_front();
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success = (packet == std::string(static_cast<const char*>(data), len));
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}
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return success;
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}
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bool CheckNoRtp() { return rtp_packets_.empty(); }
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bool CheckNoRtcp() { return rtcp_packets_.empty(); }
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void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
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void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
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virtual bool AddSendStream(const StreamParams& sp) {
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if (absl::c_linear_search(send_streams_, sp)) {
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return false;
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}
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send_streams_.push_back(sp);
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rtp_send_parameters_[sp.first_ssrc()] =
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CreateRtpParametersWithEncodings(sp);
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return true;
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}
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virtual bool RemoveSendStream(uint32_t ssrc) {
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auto parameters_iterator = rtp_send_parameters_.find(ssrc);
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if (parameters_iterator != rtp_send_parameters_.end()) {
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rtp_send_parameters_.erase(parameters_iterator);
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}
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return RemoveStreamBySsrc(&send_streams_, ssrc);
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}
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virtual void ResetUnsignaledRecvStream() {}
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virtual bool AddRecvStream(const StreamParams& sp) {
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if (absl::c_linear_search(receive_streams_, sp)) {
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return false;
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}
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receive_streams_.push_back(sp);
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rtp_receive_parameters_[sp.first_ssrc()] =
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CreateRtpParametersWithEncodings(sp);
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return true;
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}
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virtual bool RemoveRecvStream(uint32_t ssrc) {
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auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
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if (parameters_iterator != rtp_receive_parameters_.end()) {
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rtp_receive_parameters_.erase(parameters_iterator);
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}
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return RemoveStreamBySsrc(&receive_streams_, ssrc);
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}
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virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const {
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auto parameters_iterator = rtp_send_parameters_.find(ssrc);
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if (parameters_iterator != rtp_send_parameters_.end()) {
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return parameters_iterator->second;
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}
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return webrtc::RtpParameters();
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}
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virtual webrtc::RTCError SetRtpSendParameters(
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uint32_t ssrc,
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const webrtc::RtpParameters& parameters) {
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auto parameters_iterator = rtp_send_parameters_.find(ssrc);
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if (parameters_iterator != rtp_send_parameters_.end()) {
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auto result = CheckRtpParametersInvalidModificationAndValues(
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parameters_iterator->second, parameters);
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if (!result.ok())
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return result;
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parameters_iterator->second = parameters;
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return webrtc::RTCError::OK();
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}
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// Replicate the behavior of the real media channel: return false
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// when setting parameters for unknown SSRCs.
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return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
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}
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virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
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auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
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if (parameters_iterator != rtp_receive_parameters_.end()) {
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return parameters_iterator->second;
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}
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return webrtc::RtpParameters();
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}
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virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const {
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return webrtc::RtpParameters();
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}
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bool IsStreamMuted(uint32_t ssrc) const {
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bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
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// If |ssrc = 0| check if the first send stream is muted.
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if (!ret && ssrc == 0 && !send_streams_.empty()) {
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return muted_streams_.find(send_streams_[0].first_ssrc()) !=
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muted_streams_.end();
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}
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return ret;
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}
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const std::vector<StreamParams>& send_streams() const {
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return send_streams_;
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}
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const std::vector<StreamParams>& recv_streams() const {
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return receive_streams_;
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}
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bool HasRecvStream(uint32_t ssrc) const {
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return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
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}
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bool HasSendStream(uint32_t ssrc) const {
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return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
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}
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// TODO(perkj): This is to support legacy unit test that only check one
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// sending stream.
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uint32_t send_ssrc() const {
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if (send_streams_.empty())
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return 0;
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return send_streams_[0].first_ssrc();
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}
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// TODO(perkj): This is to support legacy unit test that only check one
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// sending stream.
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const std::string rtcp_cname() {
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if (send_streams_.empty())
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return "";
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return send_streams_[0].cname;
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}
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const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
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const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
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bool ready_to_send() const { return ready_to_send_; }
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int transport_overhead_per_packet() const {
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return transport_overhead_per_packet_;
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}
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rtc::NetworkRoute last_network_route() const { return last_network_route_; }
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int num_network_route_changes() const { return num_network_route_changes_; }
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void set_num_network_route_changes(int changes) {
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num_network_route_changes_ = changes;
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}
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
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int64_t packet_time_us) {
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rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
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}
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protected:
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bool MuteStream(uint32_t ssrc, bool mute) {
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if (!HasSendStream(ssrc) && ssrc != 0) {
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return false;
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}
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if (mute) {
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muted_streams_.insert(ssrc);
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} else {
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muted_streams_.erase(ssrc);
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}
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return true;
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}
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bool set_sending(bool send) {
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sending_ = send;
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return true;
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}
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void set_playout(bool playout) { playout_ = playout; }
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bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
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recv_extensions_ = extensions;
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return true;
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}
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bool SetSendExtmapAllowMixed(bool extmap_allow_mixed) {
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if (Base::ExtmapAllowMixed() != extmap_allow_mixed) {
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Base::SetExtmapAllowMixed(extmap_allow_mixed);
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}
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return true;
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}
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bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
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send_extensions_ = extensions;
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return true;
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}
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void set_send_rtcp_parameters(const RtcpParameters& params) {
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send_rtcp_parameters_ = params;
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}
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void set_recv_rtcp_parameters(const RtcpParameters& params) {
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recv_rtcp_parameters_ = params;
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}
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virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) {
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rtp_packets_.push_back(std::string(packet.cdata<char>(), packet.size()));
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}
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virtual void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
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virtual void OnNetworkRouteChanged(const std::string& transport_name,
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const rtc::NetworkRoute& network_route) {
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last_network_route_ = network_route;
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++num_network_route_changes_;
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transport_overhead_per_packet_ = network_route.packet_overhead;
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}
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bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
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bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
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private:
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bool sending_;
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bool playout_;
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std::vector<RtpExtension> recv_extensions_;
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std::vector<RtpExtension> send_extensions_;
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std::list<std::string> rtp_packets_;
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std::list<std::string> rtcp_packets_;
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std::vector<StreamParams> send_streams_;
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std::vector<StreamParams> receive_streams_;
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RtcpParameters send_rtcp_parameters_;
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RtcpParameters recv_rtcp_parameters_;
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std::set<uint32_t> muted_streams_;
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std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
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std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
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bool fail_set_send_codecs_;
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bool fail_set_recv_codecs_;
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uint32_t send_ssrc_;
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std::string rtcp_cname_;
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bool ready_to_send_;
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int transport_overhead_per_packet_;
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rtc::NetworkRoute last_network_route_;
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int num_network_route_changes_;
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};
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class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
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public:
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struct DtmfInfo {
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DtmfInfo(uint32_t ssrc, int event_code, int duration);
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uint32_t ssrc;
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int event_code;
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int duration;
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};
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explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
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const AudioOptions& options);
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~FakeVoiceMediaChannel();
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const std::vector<AudioCodec>& recv_codecs() const;
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const std::vector<AudioCodec>& send_codecs() const;
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const std::vector<AudioCodec>& codecs() const;
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const std::vector<DtmfInfo>& dtmf_info_queue() const;
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const AudioOptions& options() const;
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int max_bps() const;
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bool SetSendParameters(const AudioSendParameters& params) override;
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bool SetRecvParameters(const AudioRecvParameters& params) override;
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void SetPlayout(bool playout) override;
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void SetSend(bool send) override;
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bool SetAudioSend(uint32_t ssrc,
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bool enable,
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const AudioOptions* options,
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AudioSource* source) override;
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bool HasSource(uint32_t ssrc) const;
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bool AddRecvStream(const StreamParams& sp) override;
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bool RemoveRecvStream(uint32_t ssrc) override;
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bool CanInsertDtmf() override;
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bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
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bool SetOutputVolume(uint32_t ssrc, double volume) override;
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bool SetDefaultOutputVolume(double volume) override;
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bool GetOutputVolume(uint32_t ssrc, double* volume);
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bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
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absl::optional<int> GetBaseMinimumPlayoutDelayMs(
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uint32_t ssrc) const override;
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bool GetStats(VoiceMediaInfo* info, bool get_and_clear_legacy_stats) override;
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void SetRawAudioSink(
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uint32_t ssrc,
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std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
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void SetDefaultRawAudioSink(
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std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
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std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
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private:
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class VoiceChannelAudioSink : public AudioSource::Sink {
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public:
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explicit VoiceChannelAudioSink(AudioSource* source);
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~VoiceChannelAudioSink() override;
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void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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absl::optional<int64_t> absolute_capture_timestamp_ms) override;
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void OnClose() override;
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AudioSource* source() const;
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private:
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AudioSource* source_;
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};
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bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
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bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
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bool SetMaxSendBandwidth(int bps);
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bool SetOptions(const AudioOptions& options);
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bool SetLocalSource(uint32_t ssrc, AudioSource* source);
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FakeVoiceEngine* engine_;
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std::vector<AudioCodec> recv_codecs_;
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std::vector<AudioCodec> send_codecs_;
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std::map<uint32_t, double> output_scalings_;
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std::map<uint32_t, int> output_delays_;
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std::vector<DtmfInfo> dtmf_info_queue_;
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AudioOptions options_;
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std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
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std::unique_ptr<webrtc::AudioSinkInterface> sink_;
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int max_bps_;
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};
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// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
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bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
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uint32_t ssrc,
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int event_code,
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int duration);
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class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
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public:
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FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options);
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~FakeVideoMediaChannel();
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const std::vector<VideoCodec>& recv_codecs() const;
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const std::vector<VideoCodec>& send_codecs() const;
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const std::vector<VideoCodec>& codecs() const;
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bool rendering() const;
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const VideoOptions& options() const;
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const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
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sinks() const;
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int max_bps() const;
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bool SetSendParameters(const VideoSendParameters& params) override;
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bool SetRecvParameters(const VideoRecvParameters& params) override;
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bool AddSendStream(const StreamParams& sp) override;
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bool RemoveSendStream(uint32_t ssrc) override;
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bool GetSendCodec(VideoCodec* send_codec) override;
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bool SetSink(uint32_t ssrc,
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rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
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void SetDefaultSink(
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rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
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bool HasSink(uint32_t ssrc) const;
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bool SetSend(bool send) override;
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bool SetVideoSend(
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uint32_t ssrc,
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const VideoOptions* options,
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rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
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bool HasSource(uint32_t ssrc) const;
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bool AddRecvStream(const StreamParams& sp) override;
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bool RemoveRecvStream(uint32_t ssrc) override;
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void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
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bool GetStats(VideoMediaInfo* info) override;
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std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
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bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
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absl::optional<int> GetBaseMinimumPlayoutDelayMs(
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uint32_t ssrc) const override;
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void SetRecordableEncodedFrameCallback(
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uint32_t ssrc,
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std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
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override;
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void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
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void GenerateKeyFrame(uint32_t ssrc) override;
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private:
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bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
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bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
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bool SetOptions(const VideoOptions& options);
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bool SetMaxSendBandwidth(int bps);
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FakeVideoEngine* engine_;
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std::vector<VideoCodec> recv_codecs_;
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std::vector<VideoCodec> send_codecs_;
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std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_;
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std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
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std::map<uint32_t, int> output_delays_;
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VideoOptions options_;
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int max_bps_;
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};
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// Dummy option class, needed for the DataTraits abstraction in
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// channel_unittest.c.
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class DataOptions {};
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class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
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public:
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explicit FakeDataMediaChannel(void* unused, const DataOptions& options);
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~FakeDataMediaChannel();
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const std::vector<DataCodec>& recv_codecs() const;
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const std::vector<DataCodec>& send_codecs() const;
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const std::vector<DataCodec>& codecs() const;
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int max_bps() const;
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bool SetSendParameters(const DataSendParameters& params) override;
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bool SetRecvParameters(const DataRecvParameters& params) override;
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bool SetSend(bool send) override;
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bool SetReceive(bool receive) override;
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bool AddRecvStream(const StreamParams& sp) override;
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bool RemoveRecvStream(uint32_t ssrc) override;
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bool SendData(const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& payload,
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SendDataResult* result) override;
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SendDataParams last_sent_data_params();
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std::string last_sent_data();
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bool is_send_blocked();
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void set_send_blocked(bool blocked);
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private:
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bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
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bool SetSendCodecs(const std::vector<DataCodec>& codecs);
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bool SetMaxSendBandwidth(int bps);
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std::vector<DataCodec> recv_codecs_;
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std::vector<DataCodec> send_codecs_;
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SendDataParams last_sent_data_params_;
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std::string last_sent_data_;
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bool send_blocked_;
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int max_bps_;
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};
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class FakeVoiceEngine : public VoiceEngineInterface {
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public:
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FakeVoiceEngine();
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void Init() override;
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rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
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VoiceMediaChannel* CreateMediaChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const AudioOptions& options,
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const webrtc::CryptoOptions& crypto_options) override;
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FakeVoiceMediaChannel* GetChannel(size_t index);
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void UnregisterChannel(VoiceMediaChannel* channel);
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// TODO(ossu): For proper testing, These should either individually settable
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// or the voice engine should reference mockable factories.
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const std::vector<AudioCodec>& send_codecs() const override;
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const std::vector<AudioCodec>& recv_codecs() const override;
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void SetCodecs(const std::vector<AudioCodec>& codecs);
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void SetRecvCodecs(const std::vector<AudioCodec>& codecs);
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void SetSendCodecs(const std::vector<AudioCodec>& codecs);
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int GetInputLevel();
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bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
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void StopAecDump() override;
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std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
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const override;
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void SetRtpHeaderExtensions(
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std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
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private:
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std::vector<FakeVoiceMediaChannel*> channels_;
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std::vector<AudioCodec> recv_codecs_;
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std::vector<AudioCodec> send_codecs_;
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bool fail_create_channel_;
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std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
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friend class FakeMediaEngine;
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};
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class FakeVideoEngine : public VideoEngineInterface {
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public:
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FakeVideoEngine();
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bool SetOptions(const VideoOptions& options);
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VideoMediaChannel* CreateMediaChannel(
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webrtc::Call* call,
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const MediaConfig& config,
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const VideoOptions& options,
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const webrtc::CryptoOptions& crypto_options,
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webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
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override;
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FakeVideoMediaChannel* GetChannel(size_t index);
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void UnregisterChannel(VideoMediaChannel* channel);
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std::vector<VideoCodec> send_codecs() const override;
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std::vector<VideoCodec> recv_codecs() const override;
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void SetSendCodecs(const std::vector<VideoCodec>& codecs);
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|
void SetRecvCodecs(const std::vector<VideoCodec>& codecs);
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bool SetCapture(bool capture);
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|
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
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|
const override;
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|
void SetRtpHeaderExtensions(
|
|
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
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|
|
|
private:
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|
std::vector<FakeVideoMediaChannel*> channels_;
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|
std::vector<VideoCodec> send_codecs_;
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|
std::vector<VideoCodec> recv_codecs_;
|
|
bool capture_;
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|
VideoOptions options_;
|
|
bool fail_create_channel_;
|
|
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
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|
|
|
friend class FakeMediaEngine;
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|
};
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|
|
|
class FakeMediaEngine : public CompositeMediaEngine {
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|
public:
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|
FakeMediaEngine();
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|
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~FakeMediaEngine() override;
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|
|
|
void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
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|
void SetAudioRecvCodecs(const std::vector<AudioCodec>& codecs);
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|
void SetAudioSendCodecs(const std::vector<AudioCodec>& codecs);
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|
void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
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|
|
|
FakeVoiceMediaChannel* GetVoiceChannel(size_t index);
|
|
FakeVideoMediaChannel* GetVideoChannel(size_t index);
|
|
|
|
void set_fail_create_channel(bool fail);
|
|
|
|
private:
|
|
FakeVoiceEngine* const voice_;
|
|
FakeVideoEngine* const video_;
|
|
};
|
|
|
|
// Have to come afterwards due to declaration order
|
|
|
|
class FakeDataEngine : public DataEngineInterface {
|
|
public:
|
|
DataMediaChannel* CreateChannel(const MediaConfig& config) override;
|
|
|
|
FakeDataMediaChannel* GetChannel(size_t index);
|
|
|
|
void UnregisterChannel(DataMediaChannel* channel);
|
|
|
|
void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
|
|
|
|
const std::vector<DataCodec>& data_codecs() override;
|
|
|
|
private:
|
|
std::vector<FakeDataMediaChannel*> channels_;
|
|
std::vector<DataCodec> data_codecs_;
|
|
};
|
|
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|
} // namespace cricket
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#endif // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
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