Allow applications to control audio send bitrate through RtpParameters.

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 128 matching lines @@
                           const AudioOptions& options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
 
   rtc::DiffServCodePoint PreferredDscp() const override;
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
+  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override;
+  bool SetRtpParameters(uint32_t ssrc,
+                        const webrtc::RtpParameters& parameters) override;
+
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
   void SetSend(bool send) override;
   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;
@@ skipping 40 matching lines @@
     return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
   }
 
   int GetReceiveChannelId(uint32_t ssrc) const;
   int GetSendChannelId(uint32_t ssrc) const;
 
  private:
   bool SetOptions(const AudioOptions& options);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetSendCodecs(int channel);
+  bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
   void SetNack(int channel, bool nack_enabled);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
-  bool SetMaxSendBandwidth(int bps);
   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);
   bool SetPlayout(int channel, bool playout);
   bool ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendBitrateInternal(int bps);
+  bool SetSendBitrate(int bps);
+  bool SetSendBitrate(int channel, int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
+  bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
-  bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
   bool recv_transport_cc_enabled_ = false;
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
@@ skipping 30 matching lines @@
     int cng_payload_type = -1;
     int cng_plfreq = -1;
     webrtc::CodecInst codec_inst;
   } send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_