Remove redundant LastDecodedTimestamps.

The same information can be found in `AudioFrame.packet_infos_`.

Bug: none
Change-Id: Ib63bc41ffb896677a445d875afce0a98acea6999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37153}

api/neteq/neteq.h

@@ -312,12 +312,6 @@ class NetEq {
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
-  // Returns a vector containing the timestamps of the packets that were decoded
-  // in the last GetAudio call. If no packets were decoded in the last call, the
-  // vector is empty.
-  // Mainly intended for testing.
-  virtual std::vector<uint32_t> LastDecodedTimestamps() const = 0;
-
   // Returns the length of the audio yet to play in the sync buffer.
   // Mainly intended for testing.
   virtual int SyncBufferSizeMs() const = 0;

modules/audio_coding/neteq/neteq_impl.cc

@@ -476,11 +476,6 @@ std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
   return nack_->GetNackList(round_trip_time_ms);
 }
 
-std::vector<uint32_t> NetEqImpl::LastDecodedTimestamps() const {
-  MutexLock lock(&mutex_);
-  return last_decoded_timestamps_;
-}
-
 int NetEqImpl::SyncBufferSizeMs() const {
   MutexLock lock(&mutex_);
   return rtc::dchecked_cast<int>(sync_buffer_->FutureLength() /
@@ -779,7 +774,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
   Operation operation;
   bool play_dtmf;
   *muted = false;
-  last_decoded_timestamps_.clear();
   last_decoded_packet_infos_.clear();
   tick_timer_->Increment();
   stats_->IncreaseCounter(output_size_samples_, fs_hz_);
@@ -1464,7 +1458,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
                           AudioDecoder* decoder,
                           int* decoded_length,
                           AudioDecoder::SpeechType* speech_type) {
-  RTC_DCHECK(last_decoded_timestamps_.empty());
   RTC_DCHECK(last_decoded_packet_infos_.empty());
 
   // Do decoding.
@@ -1484,7 +1477,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
     auto opt_result = packet_list->front().frame->Decode(
         rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
                                 decoded_buffer_length_ - *decoded_length));
-    last_decoded_timestamps_.push_back(packet_list->front().timestamp);
     last_decoded_packet_infos_.push_back(
         std::move(packet_list->front().packet_info));
     packet_list->pop_front();

modules/audio_coding/neteq/neteq_impl.h

@@ -194,8 +194,6 @@ class NetEqImpl : public webrtc::NetEq {
 
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
-  std::vector<uint32_t> LastDecodedTimestamps() const override;
-
   int SyncBufferSizeMs() const override;
 
   // This accessor method is only intended for testing purposes.
@@ -395,7 +393,6 @@ class NetEqImpl : public webrtc::NetEq {
       AudioFrame::kVadPassive;
   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
       RTC_GUARDED_BY(mutex_);
-  std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(mutex_);
   std::vector<RtpPacketInfo> last_decoded_packet_infos_ RTC_GUARDED_BY(mutex_);
   ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(mutex_);
   ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);

modules/audio_coding/neteq/neteq_unittest.cc

@@ -842,67 +842,6 @@ TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
   EXPECT_FALSE(muted);
 }
 
-TEST_F(NetEqDecodingTest, LastDecodedTimestampsEmpty) {
-  EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
-
-  // Pull out data once.
-  AudioFrame output;
-  bool muted;
-  ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-
-  EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
-}
-
-TEST_F(NetEqDecodingTest, LastDecodedTimestampsOneDecoded) {
-  // Insert one packet with PCM16b WB data (this is what PopulateRtpInfo does by
-  // default). Make the length 10 ms.
-  constexpr size_t kPayloadSamples = 16 * 10;
-  constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
-  uint8_t payload[kPayloadBytes] = {0};
-
-  RTPHeader rtp_info;
-  constexpr uint32_t kRtpTimestamp = 0x1234;
-  PopulateRtpInfo(0, kRtpTimestamp, &rtp_info);
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
-
-  // Pull out data once.
-  AudioFrame output;
-  bool muted;
-  ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-
-  EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp}),
-            neteq_->LastDecodedTimestamps());
-
-  // Nothing decoded on the second call.
-  ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-  EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
-}
-
-TEST_F(NetEqDecodingTest, LastDecodedTimestampsTwoDecoded) {
-  // Insert two packets with PCM16b WB data (this is what PopulateRtpInfo does
-  // by default). Make the length 5 ms so that NetEq must decode them both in
-  // the same GetAudio call.
-  constexpr size_t kPayloadSamples = 16 * 5;
-  constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
-  uint8_t payload[kPayloadBytes] = {0};
-
-  RTPHeader rtp_info;
-  constexpr uint32_t kRtpTimestamp1 = 0x1234;
-  PopulateRtpInfo(0, kRtpTimestamp1, &rtp_info);
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
-  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + kPayloadSamples;
-  PopulateRtpInfo(1, kRtpTimestamp2, &rtp_info);
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
-
-  // Pull out data once.
-  AudioFrame output;
-  bool muted;
-  ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
-
-  EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
-            neteq_->LastDecodedTimestamps());
-}
-
 TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
   const int kNumConcealmentEvents = 19;
   const size_t kSamples = 10 * 16;

modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc

@@ -102,12 +102,8 @@ void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
                                        bool /*muted*/,
                                        NetEq* neteq) {
   get_audio_time_ms_.push_back(time_now_ms);
-  // Check what timestamps were decoded in the last GetAudio call.
+  for (const RtpPacketInfo& info : audio_frame.packet_infos_) {
-  std::vector<uint32_t> dec_ts = neteq->LastDecodedTimestamps();
+    auto it = data_.find(info.rtp_timestamp());
-  // Find those timestamps in data_, insert their decoding time and sync
-  // delay.
-  for (uint32_t ts : dec_ts) {
-    auto it = data_.find(ts);
     if (it == data_.end()) {
       // This is a packet that was split out from another packet. Skip it.
       continue;