Propagate muted parameter to VoE::Channel
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.
BUG=webrtc:5609
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1985743002
Cr-Original-Commit-Position: refs/heads/master@{#12779}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: d4ccb00b9e636cc8b4297db5555654b2546681e0
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index a2ef5b0..470f690 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -205,7 +205,9 @@
virtual void PullAudio() {
AudioFrame audio_frame;
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
+ bool muted;
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
}
virtual void InsertAudio() {
@@ -296,7 +298,9 @@
TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
AudioFrame audio_frame;
const int kSampleRateHz = 32000;
- EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
+ bool muted;
+ EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0u);
@@ -307,7 +311,8 @@
TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
- EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
+ bool muted;
+ EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame, &muted));
}
// Checks that the transport callback is invoked once for each speech packet.
@@ -806,8 +811,13 @@
// Pull audio.
for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
AudioFrame audio_frame;
+ bool muted;
EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
- &audio_frame));
+ &audio_frame, &muted));
+ if (muted) {
+ ADD_FAILURE();
+ return false;
+ }
fake_clock_->AdvanceTimeMilliseconds(10);
}
rtp_utility_->Forward(&rtp_header_);
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 25bacfd..833398a 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -323,7 +323,8 @@
bool APITest::PullAudioRunA() {
_pullEventA->Wait(100);
AudioFrame audioFrame;
- if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0) {
+ bool muted;
+ if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame, &muted) < 0) {
bool thereIsDecoder;
{
ReadLockScoped rl(_apiTestRWLock);
@@ -343,7 +344,8 @@
bool APITest::PullAudioRunB() {
_pullEventB->Wait(100);
AudioFrame audioFrame;
- if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0) {
+ bool muted;
+ if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame, &muted) < 0) {
bool thereIsDecoder;
{
ReadLockScoped rl(_apiTestRWLock);
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index e063224..7245023 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -208,8 +208,12 @@
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
-
- int32_t ok =_acm->PlayoutData10Ms(_frequency, &audioFrame);
+ bool muted;
+ int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
+ if (muted) {
+ ADD_FAILURE();
+ return false;
+ }
EXPECT_EQ(0, ok);
if (ok < 0){
return false;
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index bacfd37..80a0464 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -452,7 +452,9 @@
}
// Run received side of ACM.
- CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
+ bool muted;
+ CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
// Write output speech to file.
outfile_b_.Write10MsData(audio_frame.data_,
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index a1bdc04..24cda11 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -461,7 +461,9 @@
while (!_inFileA.EndOfFile()) {
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+ bool muted;
+ EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
}
_inFileA.Rewind();
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 9bf560d..85a2ae2 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -792,7 +792,9 @@
}
// Run received side of ACM
- EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+ bool muted;
+ EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
// Write output speech to file
out_file_.Write10MsData(
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 229dc2d..4f53e47c 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -108,7 +108,9 @@
audio_frame.timestamp_ = time_stamp;
time_stamp += frame_size_samples;
EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
- acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame);
+ bool muted;
+ acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
+ ASSERT_FALSE(muted);
out_file.Write10MsData(audio_frame);
}
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 161491b..b59ff1f 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -261,13 +261,18 @@
EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+ bool muted;
+ EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+ EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileRefA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+ EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+ EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileRefB.Write10MsData(audioFrame);
// Update time counters each time a second of data has passed.
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 8fa1fb1..50702f9 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -204,7 +204,10 @@
in_file_a_.Read10MsData(audio_frame);
ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
- ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+ bool muted;
+ ASSERT_EQ(0,
+ acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+ RTC_DCHECK(!muted);
out_file_b_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index abc84b2..f06833c 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -198,9 +198,12 @@
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
- EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
+ bool muted;
+ EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
+ EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame, &muted));
+ ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame);
}
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index 966f4c6..27a8833 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -141,8 +141,10 @@
// Is it time to pull audio?
if (time_to_playout_audio_ms_ == 0) {
time_to_playout_audio_ms_ = kPlayoutPeriodMs;
+ bool muted;
receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
- &frame_);
+ &frame_, &muted);
+ ASSERT_FALSE(muted);
fwrite(frame_.data_, sizeof(frame_.data_[0]),
frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
*action |= kAudioPlayedOut;
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 104b5e5..5d25028 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -336,7 +336,10 @@
}
// Run received side of ACM.
- ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+ bool muted;
+ ASSERT_EQ(
+ 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
// Write output speech to file.
out_file_.Write10MsData(
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 99c1c2d..5de5bf2 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -150,8 +150,10 @@
// Pull audio equivalent to the amount of audio in one RTP packet.
void Pull() {
AudioFrame frame;
+ bool muted;
for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame.
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted));
+ ASSERT_FALSE(muted);
// Had to use ASSERT_TRUE, ASSERT_EQ generated error.
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
ASSERT_EQ(1u, frame.num_channels_);
diff --git a/modules/utility/source/coder.cc b/modules/utility/source/coder.cc
index 9e43ca8..3c065e7 100644
--- a/modules/utility/source/coder.cc
+++ b/modules/utility/source/coder.cc
@@ -13,9 +13,18 @@
#include "webrtc/modules/utility/source/coder.h"
namespace webrtc {
+namespace {
+AudioCodingModule::Config GetAcmConfig(uint32_t id) {
+ AudioCodingModule::Config config;
+ // This class does not handle muted output.
+ config.neteq_config.enable_muted_state = false;
+ config.id = id;
+ return config;
+}
+} // namespace
AudioCoder::AudioCoder(uint32_t instance_id)
- : acm_(AudioCodingModule::Create(instance_id)),
+ : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
receive_codec_(),
encode_timestamp_(0),
encoded_data_(nullptr),
@@ -54,12 +63,19 @@
return -1;
}
}
- return acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio);
+ bool muted;
+ int32_t ret =
+ acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
+ RTC_DCHECK(!muted);
+ return ret;
}
int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
uint16_t& samp_freq_hz) {
- return acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio);
+ bool muted;
+ int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
+ RTC_DCHECK(!muted);
+ return ret;
}
int32_t AudioCoder::Encode(const AudioFrame& audio,
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 37b0270..339a6a5 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -483,8 +483,9 @@
event_log_->LogAudioPlayout(ssrc);
}
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
- if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
- -1) {
+ bool muted;
+ if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
+ &muted) == -1) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::GetAudioFrame() PlayoutData10Ms() failed!");
// In all likelihood, the audio in this frame is garbage. We return an
@@ -493,6 +494,7 @@
// irrelevant.
return -1;
}
+ RTC_DCHECK(!muted);
if (_RxVadDetection) {
UpdateRxVadDetection(*audioFrame);
@@ -811,6 +813,7 @@
}
acm_config.neteq_config.enable_fast_accelerate =
config.Get<NetEqFastAccelerate>().enabled;
+ acm_config.neteq_config.enable_muted_state = false;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_outputAudioLevel.Clear();
