Adds OnReceivedUplinkAllocation method to AudioEncoder.

This allows sending the full BitrateAllocationUpdate to the encoder.
This will be used in a later CL to use the link capacity field in the
update to control the Opus decoder.

Bug: webrtc:9718
Change-Id: I1c228cc318c7f9f1b0fec232e27732177b80705a
Reviewed-on: https://webrtc-review.googlesource.com/c/111509
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25739}

api/audio_codecs/BUILD.gn

@@ -30,6 +30,7 @@ rtc_source_set("audio_codecs_api") {
   ]
   deps = [
     "..:array_view",
+    "..:bitrate_allocation",
     "../..:webrtc_common",
     "../../rtc_base:checks",
     "../../rtc_base:deprecation",

api/audio_codecs/audio_encoder.cc

@@ -92,6 +92,11 @@ void AudioEncoder::OnReceivedUplinkBandwidth(
     int target_audio_bitrate_bps,
     absl::optional<int64_t> bwe_period_ms) {}
 
+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
+                            update.bwe_period.ms());
+}
+
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 
 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}

api/audio_codecs/audio_encoder.h

@@ -17,6 +17,7 @@
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/deprecation.h"
 
@@ -221,6 +222,10 @@ class AudioEncoder {
   virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
                                          absl::optional<int64_t> bwe_period_ms);
 
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update);
+
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);