commit 5ceb4ac5eda1a5ce9ac67a17f5c3c37e3a81be28
author Niels Möller <nisse@webrtc.org> Tue Aug 13 15:54:15 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Aug 20 13:42:36 2019 +0000
tree 228e7450afc09b9959dec48c279e64391f131fd8
parent 728a0ee4594752cddf13bf561edb9b7a1ffbc2ea

Delete some unused AudioCodingModule methods

Methods deleted:

  ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
  SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
  SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
  PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.

Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}

modules/audio_coding/test/target_delay_unittest.cc

diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 2fb59d1..77a2e5a 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -12,6 +12,7 @@
 
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/include/module_common_types.h"
@@ -23,19 +24,16 @@
 class TargetDelayTest : public ::testing::Test {
  protected:
   TargetDelayTest()
-      : acm_(AudioCodingModule::Create(
-            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
+      : receiver_(
+            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())) {}
 
   ~TargetDelayTest() {}
 
   void SetUp() {
-    EXPECT_TRUE(acm_.get() != NULL);
-
-    ASSERT_EQ(0, acm_->InitializeReceiver());
     constexpr int pltype = 108;
     std::map<int, SdpAudioFormat> receive_codecs = {
         {pltype, {"L16", kSampleRateHz, 1}}};
-    acm_->SetReceiveCodecs(receive_codecs);
+    receiver_.SetCodecs(receive_codecs);
 
     rtp_header_.payloadType = pltype;
     rtp_header_.timestamp = 0;
@@ -99,8 +97,9 @@
   void Push() {
     rtp_header_.timestamp += kFrameSizeSamples;
     rtp_header_.sequenceNumber++;
-    ASSERT_EQ(
-        0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_header_));
+    ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
+                                        rtc::ArrayView<const uint8_t>(
+                                            payload_, kFrameSizeSamples * 2)));
   }
 
   // Pull audio equivalent to the amount of audio in one RTP packet.
@@ -108,7 +107,7 @@
     AudioFrame frame;
     bool muted;
     for (int k = 0; k < kNum10msPerFrame; ++k) {  // Pull one frame.
-      ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted));
+      ASSERT_EQ(0, receiver_.GetAudio(-1, &frame, &muted));
       ASSERT_FALSE(muted);
       // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
       ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
@@ -135,20 +134,20 @@
   }
 
   int SetMinimumDelay(int delay_ms) {
-    return acm_->SetMinimumPlayoutDelay(delay_ms);
+    return receiver_.SetMinimumDelay(delay_ms);
   }
 
   int SetMaximumDelay(int delay_ms) {
-    return acm_->SetMaximumPlayoutDelay(delay_ms);
+    return receiver_.SetMaximumDelay(delay_ms);
   }
 
   int GetCurrentOptimalDelayMs() {
     NetworkStatistics stats;
-    acm_->GetNetworkStatistics(&stats);
+    receiver_.GetNetworkStatistics(&stats);
     return stats.preferredBufferSize;
   }
 
-  std::unique_ptr<AudioCodingModule> acm_;
+  acm2::AcmReceiver receiver_;
   RTPHeader rtp_header_;
   uint8_t payload_[kPayloadLenBytes];
 };