VoE2 API draft

BUG=4690
R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50029004

Cr-Commit-Position: refs/heads/master@{#9392}

talk/media/webrtc/fakewebrtccall.cc

@@ -39,6 +39,10 @@ FakeAudioReceiveStream::FakeAudioReceiveStream(
     : config_(config), received_packets_(0) {
 }
 
+webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
+  return webrtc::AudioReceiveStream::Stats();
+}
+
 const webrtc::AudioReceiveStream::Config&
     FakeAudioReceiveStream::GetConfig() const {
   return config_;
@@ -230,6 +234,14 @@ webrtc::Call::NetworkState FakeCall::GetNetworkState() const {
   return network_state_;
 }
 
+webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  return nullptr;
+}
+
+void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+}
+
 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));

talk/media/webrtc/fakewebrtccall.h

@@ -42,6 +42,8 @@ class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
 
   int received_packets() const { return received_packets_; }
@@ -137,6 +139,10 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
   void SetStats(const webrtc::Call::Stats& stats);
 
  private:
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(

webrtc/audio_receive_stream.h

@@ -11,37 +11,50 @@
 #ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
+#include <map>
 #include <string>
 #include <vector>
 
-#include "webrtc/common_types.h"
 #include "webrtc/config.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+class AudioDecoder;
+
 class AudioReceiveStream {
  public:
+  struct Stats {};
+
   struct Config {
-    Config() {}
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
-      Rtp() : remote_ssrc(0) {}
       std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
-      uint32_t remote_ssrc;
+      uint32_t remote_ssrc = 0;
+
+      // Sender SSRC used for sending RTCP (such as receiver reports).
+      uint32_t local_ssrc = 0;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
+
+    // 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;
   };
 
+  virtual Stats GetStats() const = 0;
+
  protected:
   virtual ~AudioReceiveStream() {}
 };
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_

webrtc/audio_send_stream.h

@@ -0,0 +1,54 @@
+/*
+ *  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_AUDIO_SEND_STREAM_H_
+#define WEBRTC_AUDIO_SEND_STREAM_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioSendStream {
+ public:
+  struct Stats {};
+
+  struct Config {
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Sender SSRC.
+      uint32_t ssrc = 0;
+
+      // RTP header extensions used for the received stream.
+      std::vector<RtpExtension> extensions;
+    } rtp;
+
+    rtc::scoped_ptr<AudioEncoder> encoder;
+    int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
+    int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
+  };
+
+  virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~AudioSendStream() {}
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_SEND_STREAM_H_

webrtc/call.h

@@ -15,12 +15,16 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio_send_stream.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
+class AudioDeviceModule;
+class AudioProcessing;
 class VoiceEngine;
+class VoiceEngineObserver;
 
 const char* Version();
 
@@ -76,6 +80,8 @@ class Call {
 
     static const int kDefaultStartBitrateBps;
 
+    // TODO(solenberg): Need to add media type to the interface for outgoing
+    // packets too.
     newapi::Transport* send_transport;
 
     // VoiceEngine used for audio/video synchronization for this Call.
@@ -96,6 +102,12 @@ class Call {
       int start_bitrate_bps;
       int max_bitrate_bps;
     } bitrate_config;
+
+    struct AudioConfig {
+      AudioDeviceModule* audio_device_manager;
+      AudioProcessing* audio_processing;
+      VoiceEngineObserver* voice_engine_observer;
+    } audio_config;
   };
 
   struct Stats {
@@ -113,6 +125,10 @@ class Call {
 
   static Call* Create(const Call::Config& config);
 
+  virtual AudioSendStream* CreateAudioSendStream(
+      const AudioSendStream::Config& config) = 0;
+  virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
+
   virtual AudioReceiveStream* CreateAudioReceiveStream(
       const AudioReceiveStream::Config& config) = 0;
   virtual void DestroyAudioReceiveStream(

webrtc/video/audio_receive_stream.cc

@@ -63,6 +63,10 @@ AudioReceiveStream::AudioReceiveStream(
   }
 }
 
+webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
+  return webrtc::AudioReceiveStream::Stats();
+}
+
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }

webrtc/video/audio_receive_stream.h

@@ -26,6 +26,8 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream {
                      const webrtc::AudioReceiveStream::Config& config);
   ~AudioReceiveStream() override {}
 
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
   bool DeliverRtcp(const uint8_t* packet, size_t length);
   bool DeliverRtp(const uint8_t* packet, size_t length);
 

webrtc/video/call.cc

@@ -72,6 +72,10 @@ class Call : public webrtc::Call, public PacketReceiver {
 
   PacketReceiver* Receiver() override;
 
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
@@ -196,6 +200,14 @@ Call::~Call() {
 
 PacketReceiver* Call::Receiver() { return this; }
 
+webrtc::AudioSendStream* Call::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  return nullptr;
+}
+
+void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+}
+
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");