Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/call/audio_send_stream.h

Index: webrtc/call/audio_send_stream.h
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 42914301911968878d4559a91ac7336c7d622c47..7575834048a8df6a5d0bc1fc9fa4d2153d625cde 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -15,10 +15,11 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -102,7 +103,8 @@ class AudioSendStream {
     rtc::Optional<std::string> audio_network_adaptor_config;
 
     struct SendCodecSpec {
-      SendCodecSpec();
+      SendCodecSpec(int payload_type, const SdpAudioFormat& format);
+      ~SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
@@ -112,15 +114,15 @@ class AudioSendStream {
 
       bool nack_enabled = false;
       bool transport_cc_enabled = false;
-      bool enable_codec_fec = false;
-      bool enable_opus_dtx = false;
-      int opus_max_playback_rate = 0;
       int cng_payload_type = -1;
-      int cng_plfreq = -1;
-      int max_ptime_ms = -1;
-      int min_ptime_ms = -1;
-      webrtc::CodecInst codec_inst;
-    } send_codec_spec;
+      int payload_type;
+      SdpAudioFormat format;
+      // If unset, use the encoder's default target bitrate.
+      rtc::Optional<int> target_bitrate_bps;
+    };
+
+    rtc::Optional<SendCodecSpec> send_codec_spec;
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
   };
 
   // Starts stream activity.