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Add muted_output parameter to ACM

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The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.

A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
This commit is contained in:
henrik.lundin
2016-05-13 03:45:24 -07:00
committed by Commit bot
parent 29dca2ce95
commit 834a6ea12b
7 changed files with 52 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
View File

@ -158,8 +158,11 @@ void AcmReceiveTestOldApi::Run() {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
bool muted;
EXPECT_EQ(0,
acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
ASSERT_FALSE(muted);
const size_t samples_per_block =
static_cast<size_t>(output_freq_hz_ * 10 / 1000);
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);

8
webrtc/modules/audio_coding/acm2/acm_receiver.cc
View File

@ -132,16 +132,16 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
return 0;
}
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
int AcmReceiver::GetAudio(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) {
// Accessing members, take the lock.
rtc::CritScope lock(&crit_sect_);
bool muted;
if (neteq_->GetAudio(audio_frame, &muted) != NetEq::kOK) {
if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
return -1;
}
RTC_DCHECK(!muted);
const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();

4
webrtc/modules/audio_coding/acm2/acm_receiver.h
View File

@ -82,11 +82,13 @@ class AcmReceiver {
// Output:
// -audio_frame : an audio frame were output data and
// associated parameters are written to.
// -muted : if true, the sample data in audio_frame is not
// populated, and must be interpreted as all zero.
//
// Return value : 0 if OK.
// -1 if NetEq returned an error.
//
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
//
// Adds a new codec to the NetEq codec database.

22
webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -285,7 +285,8 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k) {
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
}
EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
}
@ -326,13 +327,15 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
AudioFrame frame;
EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
// Expect timestamp = 0 before first packet is inserted.
EXPECT_EQ(0u, frame.timestamp_);
for (int i = 0; i < 5; ++i) {
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k) {
EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
EXPECT_EQ(0,
receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
EXPECT_EQ(expected_output_ts, frame.timestamp_);
expected_output_ts += 10 * samples_per_ms;
EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
@ -340,6 +343,7 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
EXPECT_EQ(output_channels, frame.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
EXPECT_FALSE(muted);
}
}
}
@ -388,8 +392,10 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k)
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame, &muted));
}
}
EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
}
@ -417,8 +423,10 @@ TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
InsertOnePacketOfSilence(codec.id);
for (int k = 0; k < num_10ms_frames; ++k)
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
for (int k = 0; k < num_10ms_frames; ++k) {
bool muted;
ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame, &muted));
}
}
EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
}

13
webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
View File

@ -791,9 +791,10 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame) {
AudioFrame* audio_frame,
bool* muted) {
// GetAudio always returns 10 ms, at the requested sample rate.
if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"PlayoutData failed, RecOut Failed");
return -1;
@ -802,6 +803,14 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
return 0;
}
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame) {
bool muted;
int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
RTC_DCHECK(!muted);
return ret;
}
/////////////////////////////////////////
// Statistics
//

3
webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
View File

@ -163,6 +163,9 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
int PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) override;
int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
/////////////////////////////////////////

11
webrtc/modules/audio_coding/include/audio_coding_module.h
View File

@ -687,11 +687,22 @@ class AudioCodingModule {
// and other relevant parameters, c.f.
// module_common_types.h for the definition of
// AudioFrame.
// -muted : if true, the sample data in audio_frame is not
// populated, and must be interpreted as all zero.
//
// Return value:
// -1 if the function fails,
// 0 if the function succeeds.
//
virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) = 0;
/////////////////////////////////////////////////////////////////////////////
// Same as above, but without the muted parameter. This methods should not be
// used if enable_fast_accelerate was set to true in NetEq::Config.
// TODO(henrik.lundin) Remove this method when downstream dependencies are
// ready.
virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
AudioFrame* audio_frame) = 0;