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}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index b36c064..dfd777c 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -765,17 +765,6 @@
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);
}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index f208613..a512cc7 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -150,10 +150,6 @@
// 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;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 879fb83..dbbf8f8 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -249,30 +249,6 @@
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.
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module.h b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
index b145cf4..2b23eb0 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
@@ -705,23 +705,6 @@
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.
//
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
index 6186d67..ce08c0f 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -33,7 +33,6 @@
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 @@
"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 @@
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");
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
deleted file mode 100644
index 8495e0e..0000000
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ /dev/null
@@ -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
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
index ea72665..5b1e07e 100644
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
@@ -42,7 +42,6 @@
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 @@
<< " 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
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index f3ac454..de00c95 100644
--- a/webrtc/modules/modules.gyp
+++ b/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',
diff --git a/webrtc/test/fake_voice_engine.h b/webrtc/test/fake_voice_engine.h
index 8f08929..a67f9bf 100644
--- a/webrtc/test/fake_voice_engine.h
+++ b/webrtc/test/fake_voice_engine.h
@@ -441,7 +441,6 @@
// 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 {
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 147ecb1..9140cf1 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -40,7 +40,6 @@
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 @@
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.
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 41c679a..b1c1bba 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -108,7 +108,6 @@
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);
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc
index 1c5d877c..e7327eb 100644
--- a/webrtc/video_engine/vie_sync_module.cc
+++ b/webrtc/video_engine/vie_sync_module.cc
@@ -171,18 +171,4 @@
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
diff --git a/webrtc/video_engine/vie_sync_module.h b/webrtc/video_engine/vie_sync_module.h
index ea2ae0b..2a343b8 100644
--- a/webrtc/video_engine/vie_sync_module.h
+++ b/webrtc/video_engine/vie_sync_module.h
@@ -40,9 +40,6 @@
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;
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 013f908..65cf7a6 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -3414,29 +3414,6 @@
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)
{
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 46e6750..0f7b543 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -280,7 +280,6 @@
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);
diff --git a/webrtc/voice_engine/include/voe_video_sync.h b/webrtc/voice_engine/include/voe_video_sync.h
index 1143cef..655ba63 100644
--- a/webrtc/voice_engine/include/voe_video_sync.h
+++ b/webrtc/voice_engine/include/voe_video_sync.h
@@ -64,13 +64,6 @@
// 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,
diff --git a/webrtc/voice_engine/voe_video_sync_impl.cc b/webrtc/voice_engine/voe_video_sync_impl.cc
index 811bb4e..77517c6 100644
--- a/webrtc/voice_engine/voe_video_sync_impl.cc
+++ b/webrtc/voice_engine/voe_video_sync_impl.cc
@@ -116,25 +116,6 @@
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) {
diff --git a/webrtc/voice_engine/voe_video_sync_impl.h b/webrtc/voice_engine/voe_video_sync_impl.h
index aac575c..8b367ee 100644
--- a/webrtc/voice_engine/voe_video_sync_impl.h
+++ b/webrtc/voice_engine/voe_video_sync_impl.h
@@ -23,8 +23,6 @@
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;
