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Propagate muted parameter to VoE::Channel

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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-Commit-Position: refs/heads/master@{#12779}
This commit is contained in:
henrik.lundin 2016-05-17 12:21:55 -07:00 committed by Commit bot
parent e305d956c0
commit d4ccb00b9e
15 changed files with 88 additions and 27 deletions
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18
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -205,7 +205,9 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
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, MAYBE_NetEqCalls) {
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, VerifyOutputFrame) {
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 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
// 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_);

6
webrtc/modules/audio_coding/test/APITest.cc
View File

@ -323,7 +323,8 @@ bool APITest::APIThreadB(void* obj) {
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::PullAudioRunA() {
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);

8
webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -208,8 +208,12 @@ bool Receiver::IncomingPacket() {
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;

4
webrtc/modules/audio_coding/test/TestAllCodecs.cc
View File

@ -452,7 +452,9 @@ void TestAllCodecs::Run(TestPack* channel) {
}
// 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_,

4
webrtc/modules/audio_coding/test/TestRedFec.cc
View File

@ -461,7 +461,9 @@ void TestRedFec::Run() {
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();

4
webrtc/modules/audio_coding/test/TestStereo.cc
View File

@ -792,7 +792,9 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
}
// 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(

4
webrtc/modules/audio_coding/test/TestVADDTX.cc
View File

@ -108,7 +108,9 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
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);
}

13
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
View File

@ -261,13 +261,18 @@ void TwoWayCommunication::Perform() {
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.

5
webrtc/modules/audio_coding/test/delay_test.cc
View File

@ -204,7 +204,10 @@ class DelayTest {
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_);

7
webrtc/modules/audio_coding/test/iSACTest.cc
View File

@ -198,9 +198,12 @@ void ISACTest::Run10ms() {
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);
}

4
webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
View File

@ -141,8 +141,10 @@ class InsertPacketWithTiming {
// 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;

5
webrtc/modules/audio_coding/test/opus_test.cc
View File

@ -336,7 +336,10 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
}
// 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(

4
webrtc/modules/audio_coding/test/target_delay_unittest.cc
View File

@ -150,8 +150,10 @@ class TargetDelayTest : public ::testing::Test {
// 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_);

22
webrtc/modules/utility/source/coder.cc
View File

@ -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 @@ int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
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,

7
webrtc/voice_engine/channel.cc
View File

@ -483,8 +483,9 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
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 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
// irrelevant.
return -1;
}
RTC_DCHECK(!muted);
if (_RxVadDetection) {
UpdateRxVadDetection(*audioFrame);
@ -811,6 +813,7 @@ Channel::Channel(int32_t channelId,
}
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();