Add new method AcmReceiver::last_packet_sample_rate_hz()

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 17750291dad5ed15c675ea4f2ab27124f08ed27f..c441827600a5d66edc5fc9858c64717afd3b7849 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -128,7 +128,8 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
       neteq_(NetEq::Create(config.neteq_config)),
       vad_enabled_(config.neteq_config.enable_post_decode_vad),
       clock_(config.clock),
-      resampled_last_output_frame_(true) {
+      resampled_last_output_frame_(true),
+      last_packet_sample_rate_hz_() {

[kwiberg-webrtc, 2015/11/23 13:03:50]

No need to call the default constructor explicitly.

[hlundin-webrtc, 2015/11/23 13:50:45]

Done.

   assert(clock_);
   memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
@@ -156,6 +157,11 @@ int AcmReceiver::LeastRequiredDelayMs() const {
   return neteq_->LeastRequiredDelayMs();
 }
 
+rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  return last_packet_sample_rate_hz_;
+}
+
 int AcmReceiver::last_output_sample_rate_hz() const {
   return neteq_->last_output_sample_rate_hz();
 }
@@ -190,6 +196,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
         decoder->acm_codec_id !=
             *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
       last_audio_decoder_ = decoder;
+      last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
     }
 
   }  // |crit_sect_| is released.
@@ -392,6 +399,7 @@ int AcmReceiver::RemoveAllCodecs() {
 
   // No codec is registered, invalidate last audio decoder.
   last_audio_decoder_ = nullptr;
+  last_packet_sample_rate_hz_ = rtc::Optional<int>();
   return ret_val;
 }
 
@@ -405,8 +413,10 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
     LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
     return -1;
   }
-  if (last_audio_decoder_ == &it->second)
+  if (last_audio_decoder_ == &it->second) {
     last_audio_decoder_ = nullptr;
+    last_packet_sample_rate_hz_ = rtc::Optional<int>();
+  }
   decoders_.erase(it);
   return 0;
 }
@@ -420,11 +430,6 @@ bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
   return neteq_->GetPlayoutTimestamp(timestamp);
 }
 
-int AcmReceiver::last_audio_codec_id() const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  return last_audio_decoder_ ? last_audio_decoder_->acm_codec_id : -1;
-}
-
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   CriticalSectionScoped lock(crit_sect_.get());
   if (!last_audio_decoder_) {