Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 582bde5f26216f645340ef2102d87e485ff2fafb..b802d7bab616d2ffc1b4ca2cff154a02b8ff4495 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -674,7 +674,7 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
         rtp_ts_wraparound_handler_->Unwrap(audioFrame->timestamp_);
     audioFrame->elapsed_time_ms_ =
         (unwrap_timestamp - capture_start_rtp_time_stamp_) /
-        (GetPlayoutFrequency() / 1000);
+        (GetPayloadFrequency() / 1000);
 
     {
       rtc::CritScope lock(&ts_stats_lock_);
@@ -3080,7 +3080,7 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
   uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
 
   // Remove the playout delay.
-  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+  playout_timestamp -= (delay_ms * (GetPayloadFrequency() / 1000));
 
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
@@ -3140,25 +3140,12 @@ int Channel::SetSendRtpHeaderExtension(bool enable,
   return error;
 }
 
-int32_t Channel::GetPlayoutFrequency() const {
-  int32_t playout_frequency = audio_coding_->PlayoutFrequency();
-  CodecInst current_recive_codec;
-  if (audio_coding_->ReceiveCodec(&current_recive_codec) == 0) {
-    if (STR_CASE_CMP("G722", current_recive_codec.plname) == 0) {
-      // Even though the actual sampling rate for G.722 audio is
-      // 16,000 Hz, the RTP clock rate for the G722 payload format is
-      // 8,000 Hz because that value was erroneously assigned in
-      // RFC 1890 and must remain unchanged for backward compatibility.
-      playout_frequency = 8000;
-    } else if (STR_CASE_CMP("opus", current_recive_codec.plname) == 0) {
-      // We are resampling Opus internally to 32,000 Hz until all our
-      // DSP routines can operate at 48,000 Hz, but the RTP clock
-      // rate for the Opus payload format is standardized to 48,000 Hz,
-      // because that is the maximum supported decoding sampling rate.
-      playout_frequency = 48000;
-    }
-  }
-  return playout_frequency;
+int32_t Channel::GetPayloadFrequency() const {
+  CodecInst current_receive_codec;
+  RTC_CHECK_EQ(0, audio_coding_->ReceiveCodec(&current_receive_codec))
+      << "GetPayloadFrequency cannot be called before we have a received a "
+         "payload";
+  return current_receive_codec.plfreq;

[kwiberg-webrtc, 2016/09/20 15:34:54]

CodecInst::plfreq is the actual sample rate of the codec, not the RTP timestamp
rate (which is what the old code was computing). Is this what you intended?

[ossu, 2016/09/20 15:43:51]

Hmm ... no. I was quite sure of this when I made this change, but it's been
sitting around as part of another set of CLs for quite a while. I'll re-examine
this one and get back to you. I seem to recall I wanted to get the value
directly from the SdpAudioFormat (in DecoderDatabase) but this doesn't look like
it does anymore.

 }
 
 int64_t Channel::GetRTT(bool allow_associate_channel) const {