Delete a chain of methods in ViE, VoE and ACM

The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):

ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay

The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.

This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1421013006

Cr-Commit-Position: refs/heads/master@{#10471}

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

@@ -765,17 +765,6 @@ int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
   return receiver_.RemoveCodec(payload_type);
 }
 
-int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
-  {
-    CriticalSectionScoped lock(acm_crit_sect_.get());
-    // Initialize receiver, if it is not initialized. Otherwise, initial delay
-    // is reset upon initialization of the receiver.
-    if (!receiver_initialized_)
-      InitializeReceiverSafe();
-  }
-  return receiver_.SetInitialDelay(delay_ms);
-}
-
 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
   return receiver_.EnableNack(max_nack_list_size);
 }

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

@@ -150,10 +150,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Smallest latency NetEq will maintain.
   int LeastRequiredDelayMs() const override;
 
-  // Impose an initial delay on playout. ACM plays silence until |delay_ms|
-  // audio is accumulated in NetEq buffer, then starts decoding payloads.
-  int SetInitialPlayoutDelay(int delay_ms) override;
-
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 

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

@@ -249,30 +249,6 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
   EXPECT_EQ(0, stats.decoded_plc_cng);
 }
 
-// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
-// should result in generating silence, check the associated field.
-TEST_F(AudioCodingModuleTestOldApi,
-       DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
-  RegisterCodec();
-  AudioDecodingCallStats stats;
-  const int kInitialDelay = 100;
-
-  acm_->SetInitialPlayoutDelay(kInitialDelay);
-
-  int num_calls = 0;
-  for (int time_ms = 0; time_ms < kInitialDelay;
-       time_ms += kFrameSizeMs, ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
 // Insert some packets and pull audio. Check statistics are valid. Then,
 // simulate packet loss and check if PLC and PLC-to-CNG statistics are
 // correctly updated.

webrtc/modules/audio_coding/main/include/audio_coding_module.h

@@ -704,23 +704,6 @@ class AudioCodingModule {
   virtual int32_t GetNetworkStatistics(
       NetworkStatistics* network_statistics) = 0;
 
-  //
-  // Set an initial delay for playout.
-  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
-  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
-  // from NetEq in its regular fashion, and the given delay is maintained
-  // through out the call, unless channel conditions yield to a higher jitter
-  // buffer delay.
-  //
-  // Input:
-  //   -delay_ms           : delay in milliseconds.
-  //
-  // Return values:
-  //   -1 if failed to set the delay.
-  //    0 if delay is set successfully.
-  //
-  virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
-
   //
   // Enable NACK and set the maximum size of the NACK list. If NACK is already
   // enable then the maximum NACK list size is modified accordingly.

webrtc/modules/audio_coding/main/test/delay_test.cc

@@ -33,7 +33,6 @@ DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
 DEFINE_int32(num_channels, 1, "Number of Channels.");
 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
 DEFINE_int32(delay, 0, "Delay in millisecond.");
-DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
@@ -89,10 +88,6 @@ class DelayTest {
         "Couldn't initialize receiver.\n";
     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
-    if (FLAGS_init_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
-          "Failed to set initial delay.\n";
-    }
 
     if (FLAGS_delay > 0) {
       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
@@ -172,7 +167,7 @@ class DelayTest {
   void OpenOutFile(const char* output_id) {
     std::stringstream file_stream;
     file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
+        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
     out_file_b_.Open(file_name.c_str(), 32000, "wb");

webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc

@@ -1,175 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-
-#include <assert.h>
-#include <math.h>
-
-#include <iostream>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-namespace {
-
-double FrameRms(AudioFrame& frame) {
-  size_t samples = frame.num_channels_ * frame.samples_per_channel_;
-  double rms = 0;
-  for (size_t n = 0; n < samples; ++n)
-    rms += frame.data_[n] * frame.data_[n];
-  rms /= samples;
-  rms = sqrt(rms);
-  return rms;
-}
-
-}
-
-class InitialPlayoutDelayTest : public ::testing::Test {
- protected:
-  InitialPlayoutDelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
-        channel_a2b_(NULL) {}
-
-  ~InitialPlayoutDelayTest() {
-    if (channel_a2b_ != NULL) {
-      delete channel_a2b_;
-      channel_a2b_ = NULL;
-    }
-  }
-
-  void SetUp() {
-    ASSERT_TRUE(acm_a_.get() != NULL);
-    ASSERT_TRUE(acm_b_.get() != NULL);
-
-    EXPECT_EQ(0, acm_b_->InitializeReceiver());
-    EXPECT_EQ(0, acm_a_->InitializeReceiver());
-
-    // Register all L16 codecs in receiver.
-    CodecInst codec;
-    const int kFsHz[3] = { 8000, 16000, 32000 };
-    const int kChannels[2] = { 1, 2 };
-    for (int n = 0; n < 3; ++n) {
-      for (int k = 0; k < 2; ++k) {
-        AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
-        acm_b_->RegisterReceiveCodec(codec);
-      }
-    }
-
-    // Create and connect the channel
-    channel_a2b_ = new Channel;
-    acm_a_->RegisterTransportCallback(channel_a2b_);
-    channel_a2b_->RegisterReceiverACM(acm_b_.get());
-  }
-
-  void NbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 1);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
-  void NbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 2);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
- private:
-  void Run(CodecInst codec, int initial_delay_ms) {
-    AudioFrame in_audio_frame;
-    AudioFrame out_audio_frame;
-    int num_frames = 0;
-    const int kAmp = 10000;
-    in_audio_frame.sample_rate_hz_ = codec.plfreq;
-    in_audio_frame.num_channels_ = codec.channels;
-    in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    size_t samples = in_audio_frame.num_channels_ *
-        in_audio_frame.samples_per_channel_;
-    for (size_t n = 0; n < samples; ++n) {
-      in_audio_frame.data_[n] = kAmp;
-    }
-
-    uint32_t timestamp = 0;
-    double rms = 0;
-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
-    acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
-    while (rms < kAmp / 2) {
-      in_audio_frame.timestamp_ = timestamp;
-      timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
-      ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
-      rms = FrameRms(out_audio_frame);
-      ++num_frames;
-    }
-
-    ASSERT_GE(num_frames * 10, initial_delay_ms);
-    ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
-  }
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-  Channel* channel_a2b_;
-};
-
-TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
-
-}  // namespace webrtc

webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc

@@ -42,7 +42,6 @@ DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
 DEFINE_string(delay, "", "Log for delay.");
 
 // Other setups
-DEFINE_int32(init_delay, 0, "Initial delay.");
 DEFINE_bool(verbose, false, "Verbosity.");
 DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
 
@@ -122,9 +121,6 @@ class InsertPacketWithTiming {
         << " Hz." << std::endl;
 
     // Other setups
-    if (FLAGS_init_delay > 0)
-      EXPECT_EQ(0, receive_acm_->SetInitialPlayoutDelay(FLAGS_init_delay));
-
     if (FLAGS_loss_rate > 0)
       loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
     else

webrtc/modules/modules.gyp

@@ -432,7 +432,6 @@
             'audio_coding/main/test/TimedTrace.cc',
             'audio_coding/main/test/TwoWayCommunication.cc',
             'audio_coding/main/test/iSACTest.cc',
-            'audio_coding/main/test/initial_delay_unittest.cc',
             'audio_coding/main/test/opus_test.cc',
             'audio_coding/main/test/target_delay_unittest.cc',
             'audio_coding/main/test/utility.cc',

webrtc/test/fake_voice_engine.h

@@ -441,7 +441,6 @@ class FakeVoiceEngine final : public VoiceEngineImpl {
   // VoEVideoSync
   int GetPlayoutBufferSize(int& buffer_ms) override { return -1; }
   int SetMinimumPlayoutDelay(int channel, int delay_ms) override { return -1; }
-  int SetInitialPlayoutDelay(int channel, int delay_ms) override { return -1; }
   int GetDelayEstimate(int channel,
                        int* jitter_buffer_delay_ms,
                        int* playout_buffer_delay_ms) override {

webrtc/video_engine/vie_channel.cc

@@ -40,7 +40,6 @@ namespace webrtc {
 
 const int kMaxDecodeWaitTimeMs = 50;
 static const int kMaxTargetDelayMs = 10000;
-static const float kMaxIncompleteTimeMultiplier = 3.5f;
 
 // Helper class receiving statistics callbacks.
 class ChannelStatsObserver : public CallStatsObserver {
@@ -575,33 +574,6 @@ int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
   return 0;
 }
 
-int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
-  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
-    LOG(LS_ERROR) << "Invalid receive buffer delay value.";
-    return -1;
-  }
-  int max_nack_list_size;
-  int max_incomplete_time_ms;
-  if (target_delay_ms == 0) {
-    // Real-time mode - restore default settings.
-    max_nack_reordering_threshold_ = kMaxPacketAgeToNack;
-    max_nack_list_size = kMaxNackListSize;
-    max_incomplete_time_ms = 0;
-  } else {
-    max_nack_list_size =  3 * GetRequiredNackListSize(target_delay_ms) / 4;
-    max_nack_reordering_threshold_ = max_nack_list_size;
-    // Calculate the max incomplete time and round to int.
-    max_incomplete_time_ms = static_cast<int>(kMaxIncompleteTimeMultiplier *
-        target_delay_ms + 0.5f);
-  }
-  vcm_->SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_,
-                       max_incomplete_time_ms);
-  vcm_->SetMinReceiverDelay(target_delay_ms);
-  if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0)
-    return -1;
-  return 0;
-}
-
 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
   // The max size of the nack list should be large enough to accommodate the
   // the number of packets (frames) resulting from the increased delay.

webrtc/video_engine/vie_channel.h

@@ -108,7 +108,6 @@ class ViEChannel : public VCMFrameTypeCallback,
                          int payload_type_fec);
   bool IsSendingFecEnabled();
   int SetSenderBufferingMode(int target_delay_ms);
-  int SetReceiverBufferingMode(int target_delay_ms);
   int SetSendTimestampOffsetStatus(bool enable, int id);
   int SetReceiveTimestampOffsetStatus(bool enable, int id);
   int SetSendAbsoluteSendTimeStatus(bool enable, int id);

webrtc/video_engine/vie_sync_module.cc

@@ -171,18 +171,4 @@ int32_t ViESyncModule::Process() {
   return 0;
 }
 
-int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
-  CriticalSectionScoped cs(data_cs_.get());
-  if (!voe_sync_interface_) {
-    LOG(LS_ERROR) << "voe_sync_interface_ NULL, can't set playout delay.";
-    return -1;
-  }
-  sync_->SetTargetBufferingDelay(target_delay_ms);
-  // Setting initial playout delay to voice engine (video engine is updated via
-  // the VCM interface).
-  voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
-                                              target_delay_ms);
-  return 0;
-}
-
 }  // namespace webrtc

webrtc/video_engine/vie_sync_module.h

@@ -40,9 +40,6 @@ class ViESyncModule : public Module {
 
   int VoiceChannel();
 
-  // Set target delay for buffering mode (0 = real-time mode).
-  int SetTargetBufferingDelay(int target_delay_ms);
-
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
   int32_t Process() override;

webrtc/voice_engine/channel.cc

@@ -3414,29 +3414,6 @@ int Channel::LeastRequiredDelayMs() const {
   return audio_coding_->LeastRequiredDelayMs();
 }
 
-int Channel::SetInitialPlayoutDelay(int delay_ms)
-{
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::SetInitialPlayoutDelay()");
-  if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) ||
-      (delay_ms > kVoiceEngineMaxMinPlayoutDelayMs))
-  {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetInitialPlayoutDelay() invalid min delay");
-    return -1;
-  }
-  if (audio_coding_->SetInitialPlayoutDelay(delay_ms) != 0)
-  {
-    _engineStatisticsPtr->SetLastError(
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "SetInitialPlayoutDelay() failed to set min playout delay");
-    return -1;
-  }
-  return 0;
-}
-
-
 int
 Channel::SetMinimumPlayoutDelay(int delayMs)
 {

webrtc/voice_engine/channel.h

@@ -280,7 +280,6 @@ public:
     bool GetDelayEstimate(int* jitter_buffer_delay_ms,
                           int* playout_buffer_delay_ms) const;
     int LeastRequiredDelayMs() const;
-    int SetInitialPlayoutDelay(int delay_ms);
     int SetMinimumPlayoutDelay(int delayMs);
     int GetPlayoutTimestamp(unsigned int& timestamp);
     int SetInitTimestamp(unsigned int timestamp);

webrtc/voice_engine/include/voe_video_sync.h

@@ -64,13 +64,6 @@ class WEBRTC_DLLEXPORT VoEVideoSync {
   // computes based on inter-arrival times and its playout mode.
   virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
 
-  // Sets an initial delay for the playout jitter buffer. The playout of the
-  // audio is delayed by |delay_ms| in milliseconds. Thereafter, the delay is
-  // maintained, unless NetEq's internal mechanism requires a higher latency.
-  // Such a latency is computed based on inter-arrival times and NetEq's
-  // playout mode.
-  virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0;
-
   // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and
   // the |playout_buffer_delay_ms| for a specified |channel|.
   virtual int GetDelayEstimate(int channel,

webrtc/voice_engine/voe_video_sync_impl.cc

@@ -116,25 +116,6 @@ int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel, int delayMs) {
   return channelPtr->SetMinimumPlayoutDelay(delayMs);
 }
 
-int VoEVideoSyncImpl::SetInitialPlayoutDelay(int channel, int delay_ms) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetInitialPlayoutDelay(channel=%d, delay_ms=%d)", channel,
-               delay_ms);
-
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetInitialPlayoutDelay() failed to locate channel");
-    return -1;
-  }
-  return channelPtr->SetInitialPlayoutDelay(delay_ms);
-}
-
 int VoEVideoSyncImpl::GetDelayEstimate(int channel,
                                        int* jitter_buffer_delay_ms,
                                        int* playout_buffer_delay_ms) {

webrtc/voice_engine/voe_video_sync_impl.h

@@ -23,8 +23,6 @@ class VoEVideoSyncImpl : public VoEVideoSync {
 
   int SetMinimumPlayoutDelay(int channel, int delayMs) override;
 
-  int SetInitialPlayoutDelay(int channel, int delay_ms) override;
-
   int GetDelayEstimate(int channel,
                        int* jitter_buffer_delay_ms,
                        int* playout_buffer_delay_ms) override;