NetEq: Add new method last_output_sample_rate_hz
This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467163002
Cr-Commit-Position: refs/heads/master@{#10754}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index b4dfe3a..1775029 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -123,7 +123,6 @@
id_(config.id),
last_audio_decoder_(nullptr),
previous_audio_activity_(AudioFrame::kVadPassive),
- current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
neteq_(NetEq::Create(config.neteq_config)),
@@ -157,9 +156,8 @@
return neteq_->LeastRequiredDelayMs();
}
-int AcmReceiver::current_sample_rate_hz() const {
- CriticalSectionScoped lock(crit_sect_.get());
- return current_sample_rate_hz_;
+int AcmReceiver::last_output_sample_rate_hz() const {
+ return neteq_->last_output_sample_rate_hz();
}
int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
@@ -224,23 +222,18 @@
return -1;
}
- // NetEq always returns 10 ms of audio.
- current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
+ const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
// Update if resampling is required.
- bool need_resampling = (desired_freq_hz != -1) &&
- (current_sample_rate_hz_ != desired_freq_hz);
+ const bool need_resampling =
+ (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
- int samples_per_channel_int =
- resampler_.Resample10Msec(last_audio_buffer_.get(),
- current_sample_rate_hz_,
- desired_freq_hz,
- num_channels,
- AudioFrame::kMaxDataSizeSamples,
- temp_output);
+ int samples_per_channel_int = resampler_.Resample10Msec(
+ last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+ num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
@@ -254,13 +247,9 @@
// TODO(henrik.lundin) Glitches in the output may appear if the output rate
// from NetEq changes. See WebRTC issue 3923.
if (need_resampling) {
- int samples_per_channel_int =
- resampler_.Resample10Msec(audio_buffer_.get(),
- current_sample_rate_hz_,
- desired_freq_hz,
- num_channels,
- AudioFrame::kMaxDataSizeSamples,
- audio_frame->data_);
+ int samples_per_channel_int = resampler_.Resample10Msec(
+ audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+ num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return -1;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 7dc851a..f02605b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -154,12 +154,8 @@
//
void ResetInitialDelay();
- //
- // Get the current sampling frequency in Hz.
- //
- // Return value : Sampling frequency in Hz.
- //
- int current_sample_rate_hz() const;
+ // Returns last_output_sample_rate_hz from the NetEq instance.
+ int last_output_sample_rate_hz() const;
//
// Get the current network statistics from NetEq.
@@ -287,7 +283,6 @@
int id_; // TODO(henrik.lundin) Make const.
const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
- int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
ACMResampler resampler_ GUARDED_BY(crit_sect_);
// Used in GetAudio, declared as member to avoid allocating every 10ms.
// TODO(henrik.lundin) Stack-allocate in GetAudio instead?
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
index 37bb131..3bda116 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -261,8 +261,7 @@
for (int k = 0; k < num_10ms_frames; ++k) {
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
}
- EXPECT_EQ(std::min(32000, codec.inst.plfreq),
- receiver_->current_sample_rate_hz());
+ EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
}
}
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 d4d5853..19ae4cb 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
@@ -532,17 +532,14 @@
auto codec_id = RentACodec::CodecIdFromIndex(receiver_.last_audio_codec_id());
return codec_id ? RentACodec::CodecInstById(*codec_id)->plfreq
- : receiver_.current_sample_rate_hz();
+ : receiver_.last_output_sample_rate_hz();
}
// Get current playout frequency.
int AudioCodingModuleImpl::PlayoutFrequency() const {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"PlayoutFrequency()");
-
- CriticalSectionScoped lock(acm_crit_sect_.get());
-
- return receiver_.current_sample_rate_hz();
+ return receiver_.last_output_sample_rate_hz();
}
// Register possible receive codecs, can be called multiple times,
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 9a1bc17..88677d8 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -251,6 +251,11 @@
// Returns true if the RTP timestamp is valid, otherwise false.
virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
+ // Returns the sample rate in Hz of the audio produced in the last GetAudio
+ // call. If GetAudio has not been called yet, the configured sample rate
+ // (Config::sample_rate_hz) is returned.
+ virtual int last_output_sample_rate_hz() const = 0;
+
// Not implemented.
virtual int SetTargetNumberOfChannels() = 0;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 5fa2cbd..ed0c83f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -106,6 +106,7 @@
}
fs_hz_ = fs;
fs_mult_ = fs / 8000;
+ last_output_sample_rate_hz_ = fs;
output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
decoder_frame_length_ = 3 * output_size_samples_;
WebRtcSpl_Init();
@@ -160,6 +161,13 @@
if (type) {
*type = LastOutputType();
}
+ last_output_sample_rate_hz_ =
+ rtc::checked_cast<int>(*samples_per_channel * 100);
+ RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+ last_output_sample_rate_hz_ == 16000 ||
+ last_output_sample_rate_hz_ == 32000 ||
+ last_output_sample_rate_hz_ == 48000)
+ << "Unexpected sample rate " << last_output_sample_rate_hz_;
return kOK;
}
@@ -359,6 +367,11 @@
return true;
}
+int NetEqImpl::last_output_sample_rate_hz() const {
+ CriticalSectionScoped lock(crit_sect_.get());
+ return last_output_sample_rate_hz_;
+}
+
int NetEqImpl::SetTargetNumberOfChannels() {
return kNotImplemented;
}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 60c846d..0fc204f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -168,6 +168,8 @@
bool GetPlayoutTimestamp(uint32_t* timestamp) override;
+ int last_output_sample_rate_hz() const override;
+
int SetTargetNumberOfChannels() override;
int SetTargetSampleRate() override;
@@ -375,6 +377,7 @@
StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
int fs_hz_ GUARDED_BY(crit_sect_);
int fs_mult_ GUARDED_BY(crit_sect_);
+ int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
size_t output_size_samples_ GUARDED_BY(crit_sect_);
size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
Modes last_mode_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 11fdfe9..5d1dc0c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -1230,4 +1230,13 @@
EXPECT_CALL(mock_decoder, Die());
}
+// Tests that the return value from last_output_sample_rate_hz() is equal to the
+// configured inital sample rate.
+TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
+ UseNoMocks();
+ config_.sample_rate_hz = 48000;
+ CreateInstance();
+ EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+}
+
}// namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 2088cd1..5d75a4f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -362,6 +362,7 @@
(*out_len == kBlockSize16kHz) ||
(*out_len == kBlockSize32kHz));
output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
+ EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
// Increase time.
sim_clock_ += kTimeStepMs;
@@ -895,6 +896,8 @@
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, out_data_[i]);
}
+ // Verify that the sample rate did not change from the initial configuration.
+ EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
}
class NetEqBgnTest : public NetEqDecodingTest {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index dcf5f61..13b4185 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -56,6 +56,7 @@
EXPECT_EQ(channels_, num_channels);
EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
samples_per_channel);
+ EXPECT_EQ(sample_rate_hz_, neteq_->last_output_sample_rate_hz());
return samples_per_channel;
}
