Move talk/media to webrtc/media

talk/media/webrtc/fakewebrtccall.cc

Index: talk/media/webrtc/fakewebrtccall.cc
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
deleted file mode 100644
index d50a53cb63a59b787ce67c83f4293dae3bc7db13..0000000000000000000000000000000000000000
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ /dev/null
@@ -1,443 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/media/webrtc/fakewebrtccall.h"
-
-#include <algorithm>
-#include <utility>
-
-#include "talk/media/base/rtputils.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/audio/audio_sink.h"
-
-namespace cricket {
-FakeAudioSendStream::FakeAudioSendStream(
-    const webrtc::AudioSendStream::Config& config) : config_(config) {
-  RTC_DCHECK(config.voe_channel_id != -1);
-}
-
-const webrtc::AudioSendStream::Config&
-    FakeAudioSendStream::GetConfig() const {
-  return config_;
-}
-
-void FakeAudioSendStream::SetStats(
-    const webrtc::AudioSendStream::Stats& stats) {
-  stats_ = stats;
-}
-
-FakeAudioSendStream::TelephoneEvent
-    FakeAudioSendStream::GetLatestTelephoneEvent() const {
-  return latest_telephone_event_;
-}
-
-bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
-                                             uint32_t duration_ms) {
-  latest_telephone_event_.payload_type = payload_type;
-  latest_telephone_event_.event_code = event;
-  latest_telephone_event_.duration_ms = duration_ms;
-  return true;
-}
-
-webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
-  return stats_;
-}
-
-FakeAudioReceiveStream::FakeAudioReceiveStream(
-    const webrtc::AudioReceiveStream::Config& config)
-    : config_(config), received_packets_(0) {
-  RTC_DCHECK(config.voe_channel_id != -1);
-}
-
-const webrtc::AudioReceiveStream::Config&
-    FakeAudioReceiveStream::GetConfig() const {
-  return config_;
-}
-
-void FakeAudioReceiveStream::SetStats(
-    const webrtc::AudioReceiveStream::Stats& stats) {
-  stats_ = stats;
-}
-
-void FakeAudioReceiveStream::IncrementReceivedPackets() {
-  received_packets_++;
-}
-
-webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
-  return stats_;
-}
-
-void FakeAudioReceiveStream::SetSink(
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
-  sink_ = std::move(sink);
-}
-
-FakeVideoSendStream::FakeVideoSendStream(
-    const webrtc::VideoSendStream::Config& config,
-    const webrtc::VideoEncoderConfig& encoder_config)
-    : sending_(false),
-      config_(config),
-      codec_settings_set_(false),
-      num_swapped_frames_(0) {
-  RTC_DCHECK(config.encoder_settings.encoder != NULL);
-  ReconfigureVideoEncoder(encoder_config);
-}
-
-webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const {
-  return config_;
-}
-
-webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const {
-  return encoder_config_;
-}
-
-std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() {
-  return encoder_config_.streams;
-}
-
-bool FakeVideoSendStream::IsSending() const {
-  return sending_;
-}
-
-bool FakeVideoSendStream::GetVp8Settings(
-    webrtc::VideoCodecVP8* settings) const {
-  if (!codec_settings_set_) {
-    return false;
-  }
-
-  *settings = vpx_settings_.vp8;
-  return true;
-}
-
-bool FakeVideoSendStream::GetVp9Settings(
-    webrtc::VideoCodecVP9* settings) const {
-  if (!codec_settings_set_) {
-    return false;
-  }
-
-  *settings = vpx_settings_.vp9;
-  return true;
-}
-
-int FakeVideoSendStream::GetNumberOfSwappedFrames() const {
-  return num_swapped_frames_;
-}
-
-int FakeVideoSendStream::GetLastWidth() const {
-  return last_frame_.width();
-}
-
-int FakeVideoSendStream::GetLastHeight() const {
-  return last_frame_.height();
-}
-
-int64_t FakeVideoSendStream::GetLastTimestamp() const {
-  RTC_DCHECK(last_frame_.ntp_time_ms() == 0);
-  return last_frame_.render_time_ms();
-}
-
-void FakeVideoSendStream::IncomingCapturedFrame(
-    const webrtc::VideoFrame& frame) {
-  ++num_swapped_frames_;
-  last_frame_.ShallowCopy(frame);
-}
-
-void FakeVideoSendStream::SetStats(
-    const webrtc::VideoSendStream::Stats& stats) {
-  stats_ = stats;
-}
-
-webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
-  return stats_;
-}
-
-bool FakeVideoSendStream::ReconfigureVideoEncoder(
-    const webrtc::VideoEncoderConfig& config) {
-  encoder_config_ = config;
-  if (config.encoder_specific_settings != NULL) {
-    if (config_.encoder_settings.payload_name == "VP8") {
-      vpx_settings_.vp8 = *reinterpret_cast<const webrtc::VideoCodecVP8*>(
-                              config.encoder_specific_settings);
-    } else if (config_.encoder_settings.payload_name == "VP9") {
-      vpx_settings_.vp9 = *reinterpret_cast<const webrtc::VideoCodecVP9*>(
-                              config.encoder_specific_settings);
-    } else {
-      ADD_FAILURE() << "Unsupported encoder payload: "
-                    << config_.encoder_settings.payload_name;
-    }
-  }
-  codec_settings_set_ = config.encoder_specific_settings != NULL;
-  return true;
-}
-
-webrtc::VideoCaptureInput* FakeVideoSendStream::Input() {
-  return this;
-}
-
-void FakeVideoSendStream::Start() {
-  sending_ = true;
-}
-
-void FakeVideoSendStream::Stop() {
-  sending_ = false;
-}
-
-FakeVideoReceiveStream::FakeVideoReceiveStream(
-    const webrtc::VideoReceiveStream::Config& config)
-    : config_(config), receiving_(false) {
-}
-
-webrtc::VideoReceiveStream::Config FakeVideoReceiveStream::GetConfig() {
-  return config_;
-}
-
-bool FakeVideoReceiveStream::IsReceiving() const {
-  return receiving_;
-}
-
-void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame,
-                                         int time_to_render_ms) {
-  config_.renderer->RenderFrame(frame, time_to_render_ms);
-}
-
-webrtc::VideoReceiveStream::Stats FakeVideoReceiveStream::GetStats() const {
-  return stats_;
-}
-
-void FakeVideoReceiveStream::Start() {
-  receiving_ = true;
-}
-
-void FakeVideoReceiveStream::Stop() {
-  receiving_ = false;
-}
-
-void FakeVideoReceiveStream::SetStats(
-    const webrtc::VideoReceiveStream::Stats& stats) {
-  stats_ = stats;
-}
-
-FakeCall::FakeCall(const webrtc::Call::Config& config)
-    : config_(config),
-      network_state_(webrtc::kNetworkUp),
-      num_created_send_streams_(0),
-      num_created_receive_streams_(0) {}
-
-FakeCall::~FakeCall() {
-  EXPECT_EQ(0u, video_send_streams_.size());
-  EXPECT_EQ(0u, audio_send_streams_.size());
-  EXPECT_EQ(0u, video_receive_streams_.size());
-  EXPECT_EQ(0u, audio_receive_streams_.size());
-}
-
-webrtc::Call::Config FakeCall::GetConfig() const {
-  return config_;
-}
-
-const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
-  return video_send_streams_;
-}
-
-const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
-  return video_receive_streams_;
-}
-
-const std::vector<FakeAudioSendStream*>& FakeCall::GetAudioSendStreams() {
-  return audio_send_streams_;
-}
-
-const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) {
-  for (const auto* p : GetAudioSendStreams()) {
-    if (p->GetConfig().rtp.ssrc == ssrc) {
-      return p;
-    }
-  }
-  return nullptr;
-}
-
-const std::vector<FakeAudioReceiveStream*>& FakeCall::GetAudioReceiveStreams() {
-  return audio_receive_streams_;
-}
-
-const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
-  for (const auto* p : GetAudioReceiveStreams()) {
-    if (p->GetConfig().rtp.remote_ssrc == ssrc) {
-      return p;
-    }
-  }
-  return nullptr;
-}
-
-webrtc::NetworkState FakeCall::GetNetworkState() const {
-  return network_state_;
-}
-
-webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
-    const webrtc::AudioSendStream::Config& config) {
-  FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config);
-  audio_send_streams_.push_back(fake_stream);
-  ++num_created_send_streams_;
-  return fake_stream;
-}
-
-void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
-  auto it = std::find(audio_send_streams_.begin(),
-                      audio_send_streams_.end(),
-                      static_cast<FakeAudioSendStream*>(send_stream));
-  if (it == audio_send_streams_.end()) {
-    ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter.";
-  } else {
-    delete *it;
-    audio_send_streams_.erase(it);
-  }
-}
-
-webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
-    const webrtc::AudioReceiveStream::Config& config) {
-  audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
-  ++num_created_receive_streams_;
-  return audio_receive_streams_.back();
-}
-
-void FakeCall::DestroyAudioReceiveStream(
-    webrtc::AudioReceiveStream* receive_stream) {
-  auto it = std::find(audio_receive_streams_.begin(),
-                      audio_receive_streams_.end(),
-                      static_cast<FakeAudioReceiveStream*>(receive_stream));
-  if (it == audio_receive_streams_.end()) {
-    ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown paramter.";
-  } else {
-    delete *it;
-    audio_receive_streams_.erase(it);
-  }
-}
-
-webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
-    const webrtc::VideoSendStream::Config& config,
-    const webrtc::VideoEncoderConfig& encoder_config) {
-  FakeVideoSendStream* fake_stream =
-      new FakeVideoSendStream(config, encoder_config);
-  video_send_streams_.push_back(fake_stream);
-  ++num_created_send_streams_;
-  return fake_stream;
-}
-
-void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
-  auto it = std::find(video_send_streams_.begin(),
-                      video_send_streams_.end(),
-                      static_cast<FakeVideoSendStream*>(send_stream));
-  if (it == video_send_streams_.end()) {
-    ADD_FAILURE() << "DestroyVideoSendStream called with unknown paramter.";
-  } else {
-    delete *it;
-    video_send_streams_.erase(it);
-  }
-}
-
-webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream(
-    const webrtc::VideoReceiveStream::Config& config) {
-  video_receive_streams_.push_back(new FakeVideoReceiveStream(config));
-  ++num_created_receive_streams_;
-  return video_receive_streams_.back();
-}
-
-void FakeCall::DestroyVideoReceiveStream(
-    webrtc::VideoReceiveStream* receive_stream) {
-  auto it = std::find(video_receive_streams_.begin(),
-                      video_receive_streams_.end(),
-                      static_cast<FakeVideoReceiveStream*>(receive_stream));
-  if (it == video_receive_streams_.end()) {
-    ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown paramter.";
-  } else {
-    delete *it;
-    video_receive_streams_.erase(it);
-  }
-}
-
-webrtc::PacketReceiver* FakeCall::Receiver() {
-  return this;
-}
-
-FakeCall::DeliveryStatus FakeCall::DeliverPacket(
-    webrtc::MediaType media_type,
-    const uint8_t* packet,
-    size_t length,
-    const webrtc::PacketTime& packet_time) {
-  EXPECT_GE(length, 12u);
-  uint32_t ssrc;
-  if (!GetRtpSsrc(packet, length, &ssrc))
-    return DELIVERY_PACKET_ERROR;
-
-  if (media_type == webrtc::MediaType::ANY ||
-      media_type == webrtc::MediaType::VIDEO) {
-    for (auto receiver : video_receive_streams_) {
-      if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
-        return DELIVERY_OK;
-    }
-  }
-  if (media_type == webrtc::MediaType::ANY ||
-      media_type == webrtc::MediaType::AUDIO) {
-    for (auto receiver : audio_receive_streams_) {
-      if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
-        receiver->IncrementReceivedPackets();
-        return DELIVERY_OK;
-      }
-    }
-  }
-  return DELIVERY_UNKNOWN_SSRC;
-}
-
-void FakeCall::SetStats(const webrtc::Call::Stats& stats) {
-  stats_ = stats;
-}
-
-int FakeCall::GetNumCreatedSendStreams() const {
-  return num_created_send_streams_;
-}
-
-int FakeCall::GetNumCreatedReceiveStreams() const {
-  return num_created_receive_streams_;
-}
-
-webrtc::Call::Stats FakeCall::GetStats() const {
-  return stats_;
-}
-
-void FakeCall::SetBitrateConfig(
-    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
-  config_.bitrate_config = bitrate_config;
-}
-
-void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
-  network_state_ = state;
-}
-
-void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
-  last_sent_packet_ = sent_packet;
-}
-}  // namespace cricket