Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 80 matching lines @@
     receive_timestamp = NowInTimestamp(ci->plfreq);
 
     if (STR_CASE_CMP(ci->plname, "cn") == 0) {
       if (last_audio_decoder_ && last_audio_decoder_->channels > 1) {
         // This is a CNG and the audio codec is not mono, so skip pushing in
         // packets into NetEq.
         return 0;
       }
     } else {
       last_audio_decoder_ = ci;
+      last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
+      RTC_DCHECK(last_audio_format_);
       last_packet_sample_rate_hz_ = rtc::Optional<int>(ci->plfreq);
     }
 
   }  // |crit_sect_| is released.
 
   if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
       0) {
     LOG(LERROR) << "AcmReceiver::InsertPacket "
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
@@ skipping 120 matching lines @@
 }
 
 void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
 void AcmReceiver::RemoveAllCodecs() {
   rtc::CritScope lock(&crit_sect_);
   neteq_->RemoveAllPayloadTypes();
   last_audio_decoder_ = rtc::Optional<CodecInst>();
+  last_audio_format_ = rtc::Optional<SdpAudioFormat>();
   last_packet_sample_rate_hz_ = rtc::Optional<int>();
 }
 
 int AcmReceiver::RemoveCodec(uint8_t payload_type) {
   rtc::CritScope lock(&crit_sect_);
   if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK &&
       neteq_->LastError() != NetEq::kDecoderNotFound) {
     LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
     return -1;
   }
   if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
     last_audio_decoder_ = rtc::Optional<CodecInst>();
+    last_audio_format_ = rtc::Optional<SdpAudioFormat>();
     last_packet_sample_rate_hz_ = rtc::Optional<int>();
   }
   return 0;
 }
 
 rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
   return neteq_->GetPlayoutTimestamp();
 }
 
 int AcmReceiver::FilteredCurrentDelayMs() const {
   return neteq_->FilteredCurrentDelayMs();
 }
 
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {
     return -1;
   }
   *codec = *last_audio_decoder_;
   return 0;
 }
 
+rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
+  rtc::CritScope lock(&crit_sect_);
+  return last_audio_format_;
+}
+
 void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
   NetEqNetworkStatistics neteq_stat;
   // NetEq function always returns zero, so we don't check the return value.
   neteq_->NetworkStatistics(&neteq_stat);
 
   acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
   acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
   acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
   acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
   acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
@@ skipping 69 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc