Add new method AcmReceiver::last_packet_sample_rate_hz()
This change allows us to delete AcmReceiver::last_audio_codec_id().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467183002
Cr-Commit-Position: refs/heads/master@{#10756}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 1775029..6c28933 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -156,6 +156,11 @@
return neteq_->LeastRequiredDelayMs();
}
+rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+ CriticalSectionScoped lock(crit_sect_.get());
+ return last_packet_sample_rate_hz_;
+}
+
int AcmReceiver::last_output_sample_rate_hz() const {
return neteq_->last_output_sample_rate_hz();
}
@@ -190,6 +195,7 @@
decoder->acm_codec_id !=
*RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
last_audio_decoder_ = decoder;
+ last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
}
} // |crit_sect_| is released.
@@ -392,6 +398,7 @@
// No codec is registered, invalidate last audio decoder.
last_audio_decoder_ = nullptr;
+ last_packet_sample_rate_hz_ = rtc::Optional<int>();
return ret_val;
}
@@ -405,8 +412,10 @@
LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
return -1;
}
- if (last_audio_decoder_ == &it->second)
+ if (last_audio_decoder_ == &it->second) {
last_audio_decoder_ = nullptr;
+ last_packet_sample_rate_hz_ = rtc::Optional<int>();
+ }
decoders_.erase(it);
return 0;
}
@@ -420,11 +429,6 @@
return neteq_->GetPlayoutTimestamp(timestamp);
}
-int AcmReceiver::last_audio_codec_id() const {
- CriticalSectionScoped lock(crit_sect_.get());
- return last_audio_decoder_ ? last_audio_decoder_->acm_codec_id : -1;
-}
-
int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
CriticalSectionScoped lock(crit_sect_.get());
if (!last_audio_decoder_) {
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index f02605b..bcedacd 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -15,6 +15,7 @@
#include <vector>
#include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
@@ -154,6 +155,12 @@
//
void ResetInitialDelay();
+ // Returns the sample rate of the decoder associated with the last incoming
+ // packet. If no packet of a registered non-CNG codec has been received, the
+ // return value is empty. Also, if the decoder was unregistered since the last
+ // packet was inserted, the return value is empty.
+ rtc::Optional<int> last_packet_sample_rate_hz() const;
+
// Returns last_output_sample_rate_hz from the NetEq instance.
int last_output_sample_rate_hz() const;
@@ -213,13 +220,6 @@
bool GetPlayoutTimestamp(uint32_t* timestamp);
//
- // Return the index of the codec associated with the last non-CNG/non-DTMF
- // received payload. If no non-CNG/non-DTMF payload is received -1 is
- // returned.
- //
- int last_audio_codec_id() const; // TODO(turajs): can be inline.
-
- //
// Get the audio codec associated with the last non-CNG/non-DTMF received
// payload. If no non-CNG/non-DTMF packet is received -1 is returned,
// otherwise return 0.
@@ -295,6 +295,7 @@
bool vad_enabled_;
Clock* clock_; // TODO(henrik.lundin) Make const if possible.
bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
+ rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
};
} // namespace acm2
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 3bda116..8f43ac4 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
@@ -330,7 +330,7 @@
// Has received, only, DTX. Last Audio codec is undefined.
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
- EXPECT_EQ(-1, receiver_->last_audio_codec_id());
+ EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());
for (auto id : kCodecId) {
const CodecIdInst c(id);
@@ -344,7 +344,8 @@
// of type "speech."
ASSERT_TRUE(packet_sent_);
ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
- EXPECT_EQ(c.id, receiver_->last_audio_codec_id());
+ EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+ receiver_->last_packet_sample_rate_hz());
// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
// the expected codec.
@@ -356,7 +357,8 @@
InsertOnePacketOfSilence(c.id);
ASSERT_TRUE(packet_sent_);
}
- EXPECT_EQ(c.id, receiver_->last_audio_codec_id());
+ EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+ receiver_->last_packet_sample_rate_hz());
EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
EXPECT_TRUE(CodecsEqual(c.inst, codec));
}
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 19ae4cb..5d18bda 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
@@ -525,14 +525,9 @@
// Get current receive frequency.
int AudioCodingModuleImpl::ReceiveFrequency() const {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
- "ReceiveFrequency()");
-
- CriticalSectionScoped lock(acm_crit_sect_.get());
-
- auto codec_id = RentACodec::CodecIdFromIndex(receiver_.last_audio_codec_id());
- return codec_id ? RentACodec::CodecInstById(*codec_id)->plfreq
- : receiver_.last_output_sample_rate_hz();
+ const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
+ return last_packet_sample_rate ? *last_packet_sample_rate
+ : receiver_.last_output_sample_rate_hz();
}
// Get current playout frequency.
