Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

 /*
  *  Copyright (c) 2012 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 117 matching lines @@
 
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
                                    int sample_rate_hz,
                                    int num_channels,
                                    const std::string& name) override;
 
   // 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,
                      const WebRtcRTPHeader& rtp_info) override;
 
   // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
   // One usage for this API is when pre-encoded files are pushed in ACM.
   int IncomingPayload(const uint8_t* incoming_payload,
                       const size_t payload_length,
                       uint8_t payload_type,
@@ skipping 914 matching lines @@
   return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
                             sample_rate_hz, external_decoder, name);
 }
 
 // Get current received codec.
 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
   rtc::CritScope lock(&acm_crit_sect_);
   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,
                                           const WebRtcRTPHeader& rtp_header) {
   return receiver_.InsertPacket(
       rtp_header,
       rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
 }
 
 // Minimum playout delay (Used for lip-sync).
@@ skipping 274 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc