Insert audio frame transformer between encoder and packetizer.

The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in ChannelSend, where the frame
transformation will occur in the follow-up CL.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30946}

audio/channel_send.cc

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index dd866f3..d8ac39c 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -76,7 +76,8 @@
               const webrtc::CryptoOptions& crypto_options,
               bool extmap_allow_mixed,
               int rtcp_report_interval_ms,
-              uint32_t ssrc);
+              uint32_t ssrc,
+              rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
 
   ~ChannelSend() override;
 
@@ -142,6 +143,12 @@
   void SetFrameEncryptor(
       rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
 
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  void SetEncoderToPacketizerFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
  private:
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(AudioFrameType frameType,
@@ -217,6 +224,10 @@
   // E2EE Frame Encryption Options
   const webrtc::CryptoOptions crypto_options_;
 
+  // Frame transformer used by insertable streams to transform encoded frames.
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
+      RTC_GUARDED_BY(encoder_queue_);
+
   rtc::CriticalSection bitrate_crit_section_;
   int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_crit_section_) = 0;
 
@@ -452,18 +463,20 @@
   return 0;
 }
 
-ChannelSend::ChannelSend(Clock* clock,
-                         TaskQueueFactory* task_queue_factory,
-                         ProcessThread* module_process_thread,
-                         OverheadObserver* overhead_observer,
-                         Transport* rtp_transport,
-                         RtcpRttStats* rtcp_rtt_stats,
-                         RtcEventLog* rtc_event_log,
-                         FrameEncryptorInterface* frame_encryptor,
-                         const webrtc::CryptoOptions& crypto_options,
-                         bool extmap_allow_mixed,
-                         int rtcp_report_interval_ms,
-                         uint32_t ssrc)
+ChannelSend::ChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    ProcessThread* module_process_thread,
+    OverheadObserver* overhead_observer,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
     : event_log_(rtc_event_log),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
@@ -478,6 +491,7 @@
           new RateLimiter(clock, kMaxRetransmissionWindowMs)),
       frame_encryptor_(frame_encryptor),
       crypto_options_(crypto_options),
+      frame_transformer_(std::move(frame_transformer)),
       encoder_queue_(task_queue_factory->CreateTaskQueue(
           "AudioEncoder",
           TaskQueueFactory::Priority::NORMAL)) {
@@ -898,6 +912,16 @@
   });
 }
 
+void ChannelSend::SetEncoderToPacketizerFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  encoder_queue_.PostTask(
+      [this, frame_transformer = std::move(frame_transformer)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        frame_transformer_ = std::move(frame_transformer);
+      });
+}
+
 void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
   // Invoke audio encoders OnReceivedRtt().
   CallEncoder(
@@ -918,11 +942,13 @@
     const webrtc::CryptoOptions& crypto_options,
     bool extmap_allow_mixed,
     int rtcp_report_interval_ms,
-    uint32_t ssrc) {
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
   return std::make_unique<ChannelSend>(
       clock, task_queue_factory, module_process_thread, overhead_observer,
       rtp_transport, rtcp_rtt_stats, rtc_event_log, frame_encryptor,
-      crypto_options, extmap_allow_mixed, rtcp_report_interval_ms, ssrc);
+      crypto_options, extmap_allow_mixed, rtcp_report_interval_ms, ssrc,
+      std::move(frame_transformer));
 }
 
 }  // namespace voe