Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 99b539ab6440745769efcd58606f9fc6823cf4ca..9d9e74444fc1ec5b68474e9facf9a2d437e6ec23 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -135,6 +135,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Get current received codec.
   int ReceiveCodec(CodecInst* current_codec) const override;
 
+  rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
+
   // Incoming packet from network parsed and ready for decode.
   int IncomingPacket(const uint8_t* incoming_payload,
                      const size_t payload_length,
@@ -1069,6 +1071,11 @@ int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
   return receiver_.LastAudioCodec(current_codec);
 }
 
+rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
+  rtc::CritScope lock(&acm_crit_sect_);
+  return receiver_.LastAudioFormat();
+}
+
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,