Support N unsignaled audio streams

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  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.
  */
 
@@ skipping 170 matching lines @@
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
                     const AudioOptions* options,
                     AudioSource* source) override;
   bool AddSendStream(const StreamParams& sp) override;
   bool RemoveSendStream(uint32_t ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool RemoveRecvStream(uint32_t ssrc) override;
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
+  // SSRC=0 will apply the new volume to current and future unsignaled streams.
   bool SetOutputVolume(uint32_t ssrc, double volume) override;
 
   bool CanInsertDtmf() override;
   bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
 
   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
   void OnReadyToSend(bool ready) override;
   void OnTransportOverheadChanged(int transport_overhead_per_packet) override;
   bool GetStats(VoiceMediaInfo* info) override;
 
+  // SSRC=0 will set the audio sink on the latest unsignaled stream, future or
+  // current. Only one stream at a time will use the sink.
   void SetRawAudioSink(
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override {
     rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
     rtc::PacketOptions rtc_options;
@@ skipping 15 matching lines @@
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   void ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
-  bool IsDefaultRecvStream(uint32_t ssrc) {
-    return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
-  }
   bool SetMaxSendBitrate(int bps);
   bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
   void SetupRecording();
+  // Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
+  // unsignaled anymore (i.e. it is now removed, or signaled), and return true.
+  bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
   int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   int dtmf_payload_freq_ = -1;
   bool recv_transport_cc_enabled_ = false;
   bool recv_nack_enabled_ = false;
   bool desired_playout_ = false;
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
   webrtc::Call::Config::BitrateConfig bitrate_config_;
 
-  // SSRC of unsignalled receive stream, or -1 if there isn't one.
-  int64_t default_recv_ssrc_ = -1;
-  // Volume for unsignalled stream, which may be set before the stream exists.
+  // Queue of unsignaled SSRCs; oldest at the beginning.
+  std::vector<uint32_t> unsignaled_recv_ssrcs_;
+
+  // Volume for unsignaled streams, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
-  // Sink for unsignalled stream, which may be set before the stream exists.
+  // Sink for latest unsignaled stream - may be set before the stream exists.
   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661
   uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_