Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index a9550fb..6415074 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -209,6 +209,8 @@
//
int LastAudioCodec(CodecInst* codec) const;
+ rtc::Optional<SdpAudioFormat> LastAudioFormat() const;
+
//
// Get a decoder given its registered payload-type.
//
@@ -273,6 +275,7 @@
rtc::CriticalSection crit_sect_;
rtc::Optional<CodecInst> last_audio_decoder_ GUARDED_BY(crit_sect_);
+ rtc::Optional<SdpAudioFormat> last_audio_format_ GUARDED_BY(crit_sect_);
ACMResampler resampler_ GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);