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/*
* 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_FAKE_MEDIA_ENGINE_H_
#define MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
#include <atomic>
#include <list>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/functional/any_invocable.h"
#include "api/audio/audio_processing.h"
#include "api/call/audio_sink.h"
#include "api/media_types.h"
#include "media/base/audio_source.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/media_engine.h"
#include "media/base/rtp_utils.h"
#include "media/base/stream_params.h"
#include "media/engine/webrtc_video_engine.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network_route.h"
#include "rtc_base/thread.h"
#include "test/explicit_key_value_config.h"
#include "test/scoped_key_value_config.h"
using webrtc::RtpExtension;
namespace cricket {
class FakeMediaEngine;
class FakeVideoEngine;
class FakeVoiceEngine;
// A common helper class that handles sending and receiving RTP/RTCP packets.
template <class Base>
class RtpReceiveChannelHelper : public Base, public MediaChannelUtil {
public:
explicit RtpReceiveChannelHelper(webrtc::TaskQueueBase* network_thread)
: MediaChannelUtil(network_thread),
playout_(false),
fail_set_recv_codecs_(false),
transport_overhead_per_packet_(0),
num_network_route_changes_(0) {}
virtual ~RtpReceiveChannelHelper() = default;
const std::vector<RtpExtension>& recv_extensions() {
return recv_extensions_;
}
bool playout() const { return playout_; }
const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
bool SendRtcp(const void* data, size_t len) {
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return Base::SendRtcp(&packet, rtc::PacketOptions());
}
bool CheckRtp(const void* data, size_t len) {
bool success = !rtp_packets_.empty();
if (success) {
std::string packet = rtp_packets_.front();
rtp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckRtcp(const void* data, size_t len) {
bool success = !rtcp_packets_.empty();
if (success) {
std::string packet = rtcp_packets_.front();
rtcp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckNoRtp() { return rtp_packets_.empty(); }
bool CheckNoRtcp() { return rtcp_packets_.empty(); }
void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
void ResetUnsignaledRecvStream() override {}
std::optional<uint32_t> GetUnsignaledSsrc() const override {
return std::nullopt;
}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {}
virtual bool SetLocalSsrc(const StreamParams& sp) { return true; }
void OnDemuxerCriteriaUpdatePending() override {}
void OnDemuxerCriteriaUpdateComplete() override {}
bool AddRecvStream(const StreamParams& sp) override {
if (absl::c_linear_search(receive_streams_, sp)) {
return false;
}
receive_streams_.push_back(sp);
rtp_receive_parameters_[sp.first_ssrc()] =
CreateRtpParametersWithEncodings(sp);
return true;
}
bool RemoveRecvStream(uint32_t ssrc) override {
auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
if (parameters_iterator != rtp_receive_parameters_.end()) {
rtp_receive_parameters_.erase(parameters_iterator);
}
return RemoveStreamBySsrc(&receive_streams_, ssrc);
}
webrtc::RtpParameters GetRtpReceiverParameters(uint32_t ssrc) const override {
auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
if (parameters_iterator != rtp_receive_parameters_.end()) {
return parameters_iterator->second;
}
return webrtc::RtpParameters();
}
webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override {
return webrtc::RtpParameters();
}
const std::vector<StreamParams>& recv_streams() const {
return receive_streams_;
}
bool HasRecvStream(uint32_t ssrc) const {
return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
}
const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
int transport_overhead_per_packet() const {
return transport_overhead_per_packet_;
}
rtc::NetworkRoute last_network_route() const { return last_network_route_; }
int num_network_route_changes() const { return num_network_route_changes_; }
void set_num_network_route_changes(int changes) {
num_network_route_changes_ = changes;
}
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us) {
rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
}
void SetFrameDecryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override {}
void SetDepacketizerToDecoderFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {}
void SetInterface(MediaChannelNetworkInterface* iface) override {
network_interface_ = iface;
MediaChannelUtil::SetInterface(iface);
}
protected:
void set_playout(bool playout) { playout_ = playout; }
bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
recv_extensions_ = extensions;
return true;
}
void set_recv_rtcp_parameters(const RtcpParameters& params) {
recv_rtcp_parameters_ = params;
}
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override {
rtp_packets_.push_back(
std::string(packet.Buffer().cdata<char>(), packet.size()));
}
bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
private:
bool playout_;
std::vector<RtpExtension> recv_extensions_;
std::list<std::string> rtp_packets_;
std::list<std::string> rtcp_packets_;
std::vector<StreamParams> receive_streams_;
RtcpParameters recv_rtcp_parameters_;
std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
bool fail_set_recv_codecs_;
std::string rtcp_cname_;
int transport_overhead_per_packet_;
rtc::NetworkRoute last_network_route_;
int num_network_route_changes_;
MediaChannelNetworkInterface* network_interface_ = nullptr;
};
// A common helper class that handles sending and receiving RTP/RTCP packets.
template <class Base>
class RtpSendChannelHelper : public Base, public MediaChannelUtil {
public:
explicit RtpSendChannelHelper(webrtc::TaskQueueBase* network_thread)
: MediaChannelUtil(network_thread),
sending_(false),
fail_set_send_codecs_(false),
send_ssrc_(0),
ready_to_send_(false),
transport_overhead_per_packet_(0),
num_network_route_changes_(0) {}
virtual ~RtpSendChannelHelper() = default;
const std::vector<RtpExtension>& send_extensions() {
return send_extensions_;
}
bool sending() const { return sending_; }
const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
bool SendPacket(const void* data,
size_t len,
const rtc::PacketOptions& options) {
if (!sending_) {
return false;
}
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return MediaChannelUtil::SendPacket(&packet, options);
}
bool SendRtcp(const void* data, size_t len) {
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return MediaChannelUtil::SendRtcp(&packet, rtc::PacketOptions());
}
bool CheckRtp(const void* data, size_t len) {
bool success = !rtp_packets_.empty();
if (success) {
std::string packet = rtp_packets_.front();
rtp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckRtcp(const void* data, size_t len) {
bool success = !rtcp_packets_.empty();
if (success) {
std::string packet = rtcp_packets_.front();
rtcp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckNoRtp() { return rtp_packets_.empty(); }
bool CheckNoRtcp() { return rtcp_packets_.empty(); }
void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
bool AddSendStream(const StreamParams& sp) override {
if (absl::c_linear_search(send_streams_, sp)) {
return false;
}
send_streams_.push_back(sp);
rtp_send_parameters_[sp.first_ssrc()] =
CreateRtpParametersWithEncodings(sp);
if (ssrc_list_changed_callback_) {
std::set<uint32_t> ssrcs_in_use;
for (const auto& send_stream : send_streams_) {
ssrcs_in_use.insert(send_stream.first_ssrc());
}
ssrc_list_changed_callback_(ssrcs_in_use);
}
return true;
}
bool RemoveSendStream(uint32_t ssrc) override {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
rtp_send_parameters_.erase(parameters_iterator);
}
return RemoveStreamBySsrc(&send_streams_, ssrc);
}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {
ssrc_list_changed_callback_ = std::move(callback);
}
void SetExtmapAllowMixed(bool extmap_allow_mixed) override {
return MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed);
}
bool ExtmapAllowMixed() const override {
return MediaChannelUtil::ExtmapAllowMixed();
}
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
return parameters_iterator->second;
}
return webrtc::RtpParameters();
}
webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback) override {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
auto result = CheckRtpParametersInvalidModificationAndValues(
parameters_iterator->second, parameters,
webrtc::test::ExplicitKeyValueConfig(""));
if (!result.ok()) {
return webrtc::InvokeSetParametersCallback(callback, result);
}
parameters_iterator->second = parameters;
return webrtc::InvokeSetParametersCallback(callback,
webrtc::RTCError::OK());
}
// Replicate the behavior of the real media channel: return false
// when setting parameters for unknown SSRCs.
return InvokeSetParametersCallback(
callback, webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR));
}
bool IsStreamMuted(uint32_t ssrc) const {
bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
// If |ssrc = 0| check if the first send stream is muted.
if (!ret && ssrc == 0 && !send_streams_.empty()) {
return muted_streams_.find(send_streams_[0].first_ssrc()) !=
muted_streams_.end();
}
return ret;
}
const std::vector<StreamParams>& send_streams() const {
return send_streams_;
}
bool HasSendStream(uint32_t ssrc) const {
return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
}
// TODO(perkj): This is to support legacy unit test that only check one
// sending stream.
uint32_t send_ssrc() const {
if (send_streams_.empty())
return 0;
return send_streams_[0].first_ssrc();
}
const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
bool ready_to_send() const { return ready_to_send_; }
int transport_overhead_per_packet() const {
return transport_overhead_per_packet_;
}
rtc::NetworkRoute last_network_route() const { return last_network_route_; }
int num_network_route_changes() const { return num_network_route_changes_; }
void set_num_network_route_changes(int changes) {
num_network_route_changes_ = changes;
}
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us) {
rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
}
// Stuff that deals with encryptors, transformers and the like
void SetFrameEncryptor(uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
frame_encryptor) override {}
void SetEncoderToPacketizerFrameTransformer(
uint32_t ssrc,
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {}
void SetInterface(MediaChannelNetworkInterface* iface) override {
network_interface_ = iface;
MediaChannelUtil::SetInterface(iface);
}
bool HasNetworkInterface() const override {
return network_interface_ != nullptr;
}
protected:
bool MuteStream(uint32_t ssrc, bool mute) {
if (!HasSendStream(ssrc) && ssrc != 0) {
return false;
}
if (mute) {
muted_streams_.insert(ssrc);
} else {
muted_streams_.erase(ssrc);
}
return true;
}
bool set_sending(bool send) {
sending_ = send;
return true;
}
bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
send_extensions_ = extensions;
return true;
}
void set_send_rtcp_parameters(const RtcpParameters& params) {
send_rtcp_parameters_ = params;
}
void OnPacketSent(const rtc::SentPacket& sent_packet) override {}
void OnReadyToSend(bool ready) override { ready_to_send_ = ready; }
void OnNetworkRouteChanged(absl::string_view transport_name,
const rtc::NetworkRoute& network_route) override {
last_network_route_ = network_route;
++num_network_route_changes_;
transport_overhead_per_packet_ = network_route.packet_overhead;
}
bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
private:
// TODO(bugs.webrtc.org/12783): This flag is used from more than one thread.
// As a workaround for tsan, it's currently std::atomic but that might not
// be the appropriate fix.
std::atomic<bool> sending_;
std::vector<RtpExtension> send_extensions_;
std::list<std::string> rtp_packets_;
std::list<std::string> rtcp_packets_;
std::vector<StreamParams> send_streams_;
RtcpParameters send_rtcp_parameters_;
std::set<uint32_t> muted_streams_;
std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
bool fail_set_send_codecs_;
uint32_t send_ssrc_;
std::string rtcp_cname_;
bool ready_to_send_;
int transport_overhead_per_packet_;
rtc::NetworkRoute last_network_route_;
int num_network_route_changes_;
MediaChannelNetworkInterface* network_interface_ = nullptr;
absl::AnyInvocable<void(const std::set<uint32_t>&)>
ssrc_list_changed_callback_ = nullptr;
};
class FakeVoiceMediaReceiveChannel
: public RtpReceiveChannelHelper<VoiceMediaReceiveChannelInterface> {
public:
struct DtmfInfo {
DtmfInfo(uint32_t ssrc, int event_code, int duration);
uint32_t ssrc;
int event_code;
int duration;
};
FakeVoiceMediaReceiveChannel(const AudioOptions& options,
webrtc::TaskQueueBase* network_thread);
virtual ~FakeVoiceMediaReceiveChannel();
// Test methods
const std::vector<Codec>& recv_codecs() const;
const std::vector<DtmfInfo>& dtmf_info_queue() const;
const AudioOptions& options() const;
int max_bps() const;
bool HasSource(uint32_t ssrc) const;
// Overrides
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return nullptr;
}
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return this;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
bool SetReceiverParameters(const AudioReceiverParameters& params) override;
void SetPlayout(bool playout) override;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool SetDefaultOutputVolume(double volume) override;
bool GetOutputVolume(uint32_t ssrc, double* volume);
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
bool GetStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) override;
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
void SetDefaultRawAudioSink(
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
void SetReceiveNackEnabled(bool enabled) override {}
void SetRtcpMode(webrtc::RtcpMode mode) override {}
void SetReceiveNonSenderRttEnabled(bool enabled) override {}
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
public:
explicit VoiceChannelAudioSink(AudioSource* source);
~VoiceChannelAudioSink() override;
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
std::optional<int64_t> absolute_capture_timestamp_ms) override;
void OnClose() override;
int NumPreferredChannels() const override { return -1; }
AudioSource* source() const;
private:
AudioSource* source_;
};
bool SetRecvCodecs(const std::vector<Codec>& codecs);
bool SetMaxSendBandwidth(int bps);
bool SetOptions(const AudioOptions& options);
std::vector<Codec> recv_codecs_;
std::map<uint32_t, double> output_scalings_;
std::map<uint32_t, int> output_delays_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
std::unique_ptr<webrtc::AudioSinkInterface> sink_;
int max_bps_;
};
class FakeVoiceMediaSendChannel
: public RtpSendChannelHelper<VoiceMediaSendChannelInterface> {
public:
struct DtmfInfo {
DtmfInfo(uint32_t ssrc, int event_code, int duration);
uint32_t ssrc;
int event_code;
int duration;
};
FakeVoiceMediaSendChannel(const AudioOptions& options,
webrtc::TaskQueueBase* network_thread);
~FakeVoiceMediaSendChannel() override;
const std::vector<Codec>& send_codecs() const;
const std::vector<DtmfInfo>& dtmf_info_queue() const;
const AudioOptions& options() const;
int max_bps() const;
bool HasSource(uint32_t ssrc) const;
bool GetOutputVolume(uint32_t ssrc, double* volume);
// Overrides
VideoMediaSendChannelInterface* AsVideoSendChannel() override {
return nullptr;
}
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
bool SetSenderParameters(const AudioSenderParameter& params) override;
void SetSend(bool send) override;
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) override;
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
bool SenderNackEnabled() const override { return false; }
bool SenderNonSenderRttEnabled() const override { return false; }
void SetReceiveNackEnabled(bool enabled) {}
void SetReceiveNonSenderRttEnabled(bool enabled) {}
bool SendCodecHasNack() const override { return false; }
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
std::optional<Codec> GetSendCodec() const override;
bool GetStats(VoiceMediaSendInfo* stats) override;
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
public:
explicit VoiceChannelAudioSink(AudioSource* source);
~VoiceChannelAudioSink() override;
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
std::optional<int64_t> absolute_capture_timestamp_ms) override;
void OnClose() override;
int NumPreferredChannels() const override { return -1; }
AudioSource* source() const;
private:
AudioSource* source_;
};
bool SetSendCodecs(const std::vector<Codec>& codecs);
bool SetMaxSendBandwidth(int bps);
bool SetOptions(const AudioOptions& options);
bool SetLocalSource(uint32_t ssrc, AudioSource* source);
std::vector<Codec> send_codecs_;
std::map<uint32_t, double> output_scalings_;
std::map<uint32_t, int> output_delays_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
int max_bps_;
};
// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
bool CompareDtmfInfo(const FakeVoiceMediaSendChannel::DtmfInfo& info,
uint32_t ssrc,
int event_code,
int duration);
class FakeVideoMediaReceiveChannel
: public RtpReceiveChannelHelper<VideoMediaReceiveChannelInterface> {
public:
FakeVideoMediaReceiveChannel(const VideoOptions& options,
webrtc::TaskQueueBase* network_thread);
virtual ~FakeVideoMediaReceiveChannel();
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return this;
}
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return nullptr;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
const std::vector<Codec>& recv_codecs() const;
const std::vector<Codec>& send_codecs() const;
bool rendering() const;
const VideoOptions& options() const;
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
sinks() const;
int max_bps() const;
bool SetReceiverParameters(const VideoReceiverParameters& params) override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
void SetDefaultSink(
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool HasSink(uint32_t ssrc) const;
void SetReceive(bool receive) override {}
bool HasSource(uint32_t ssrc) const;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
override;
void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
void RequestRecvKeyFrame(uint32_t ssrc) override;
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
std::optional<int> rtx_time) override {}
bool GetStats(VideoMediaReceiveInfo* info) override;
bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
RTC_CHECK_NOTREACHED();
return false;
}
private:
bool SetRecvCodecs(const std::vector<Codec>& codecs);
bool SetSendCodecs(const std::vector<Codec>& codecs);
bool SetOptions(const VideoOptions& options);
bool SetMaxSendBandwidth(int bps);
std::vector<Codec> recv_codecs_;
std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_;
std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
std::map<uint32_t, int> output_delays_;
VideoOptions options_;
int max_bps_;
};
class FakeVideoMediaSendChannel
: public RtpSendChannelHelper<VideoMediaSendChannelInterface> {
public:
FakeVideoMediaSendChannel(const VideoOptions& options,
webrtc::TaskQueueBase* network_thread);
virtual ~FakeVideoMediaSendChannel();
VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
return nullptr;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
const std::vector<Codec>& send_codecs() const;
const std::vector<Codec>& codecs() const;
const VideoOptions& options() const;
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
sinks() const;
int max_bps() const;
bool SetSenderParameters(const VideoSenderParameters& params) override;
std::optional<Codec> GetSendCodec() const override;
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
bool HasSource(uint32_t ssrc) const;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
void GenerateSendKeyFrame(uint32_t ssrc,
const std::vector<std::string>& rids) override;
webrtc::RtcpMode SendCodecRtcpMode() const override {
return webrtc::RtcpMode::kCompound;
}
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
void SetSsrcListChangedCallback(
absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {}
bool SendCodecHasLntf() const override { return false; }
bool SendCodecHasNack() const override { return false; }
std::optional<int> SendCodecRtxTime() const override { return std::nullopt; }
bool GetStats(VideoMediaSendInfo* info) override;
private:
bool SetSendCodecs(const std::vector<Codec>& codecs);
bool SetOptions(const VideoOptions& options);
bool SetMaxSendBandwidth(int bps);
std::vector<Codec> send_codecs_;
std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
VideoOptions options_;
int max_bps_;
};
class FakeVoiceEngine : public VoiceEngineInterface {
public:
FakeVoiceEngine();
void Init() override;
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override;
std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::AudioCodecPairId codec_pair_id) override;
// TODO(ossu): For proper testing, These should either individually settable
// or the voice engine should reference mockable factories.
const std::vector<Codec>& send_codecs() const override;
const std::vector<Codec>& recv_codecs() const override;
void SetCodecs(const std::vector<Codec>& codecs);
void SetRecvCodecs(const std::vector<Codec>& codecs);
void SetSendCodecs(const std::vector<Codec>& codecs);
int GetInputLevel();
bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
void StopAecDump() override;
std::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
override;
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
void SetRtpHeaderExtensions(
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
private:
std::vector<Codec> recv_codecs_;
std::vector<Codec> send_codecs_;
bool fail_create_channel_;
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
friend class FakeMediaEngine;
};
class FakeVideoEngine : public VideoEngineInterface {
public:
FakeVideoEngine();
bool SetOptions(const VideoOptions& options);
std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options,
webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
override;
std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) override;
FakeVideoMediaSendChannel* GetSendChannel(size_t index);
FakeVideoMediaReceiveChannel* GetReceiveChannel(size_t index);
std::vector<Codec> send_codecs() const override { return send_codecs(true); }
std::vector<Codec> recv_codecs() const override { return recv_codecs(true); }
std::vector<Codec> send_codecs(bool include_rtx) const override;
std::vector<Codec> recv_codecs(bool include_rtx) const override;
void SetSendCodecs(const std::vector<Codec>& codecs);
void SetRecvCodecs(const std::vector<Codec>& codecs);
bool SetCapture(bool capture);
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
void SetRtpHeaderExtensions(
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
private:
std::vector<Codec> send_codecs_;
std::vector<Codec> recv_codecs_;
bool capture_;
VideoOptions options_;
bool fail_create_channel_;
std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
friend class FakeMediaEngine;
};
class FakeMediaEngine : public CompositeMediaEngine {
public:
FakeMediaEngine();
~FakeMediaEngine() override;
void SetAudioCodecs(const std::vector<Codec>& codecs);
void SetAudioRecvCodecs(const std::vector<Codec>& codecs);
void SetAudioSendCodecs(const std::vector<Codec>& codecs);
void SetVideoCodecs(const std::vector<Codec>& codecs);
void set_fail_create_channel(bool fail);
FakeVoiceEngine* fake_voice_engine() { return voice_; }
FakeVideoEngine* fake_video_engine() { return video_; }
private:
FakeVoiceEngine* const voice_;
FakeVideoEngine* const video_;
};
} // namespace cricket
#endif // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_