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Remove redundant LastDecodedTimestamps.

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The same information can be found in `AudioFrame.packet_infos_`.

Bug: none
Change-Id: Ib63bc41ffb896677a445d875afce0a98acea6999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37153}
This commit is contained in:
Jakob Ivarsson
2022-06-06 17:42:54 +02:00
committed by WebRTC LUCI CQ
parent 640b51eeae
commit 664e30ff57
5 changed files with 2 additions and 84 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
api/neteq/neteq.h
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@ -312,12 +312,6 @@ class NetEq {
virtual std::vector<uint16_t> GetNackList(
int64_t round_trip_time_ms) const = 0;
// Returns a vector containing the timestamps of the packets that were decoded
// in the last GetAudio call. If no packets were decoded in the last call, the
// vector is empty.
// Mainly intended for testing.
virtual std::vector<uint32_t> LastDecodedTimestamps() const = 0;
// Returns the length of the audio yet to play in the sync buffer.
// Mainly intended for testing.
virtual int SyncBufferSizeMs() const = 0;

8
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -476,11 +476,6 @@ std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
return nack_->GetNackList(round_trip_time_ms);
}
std::vector<uint32_t> NetEqImpl::LastDecodedTimestamps() const {
MutexLock lock(&mutex_);
return last_decoded_timestamps_;
}
int NetEqImpl::SyncBufferSizeMs() const {
MutexLock lock(&mutex_);
return rtc::dchecked_cast<int>(sync_buffer_->FutureLength() /
@ -779,7 +774,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
Operation operation;
bool play_dtmf;
*muted = false;
last_decoded_timestamps_.clear();
last_decoded_packet_infos_.clear();
tick_timer_->Increment();
stats_->IncreaseCounter(output_size_samples_, fs_hz_);
@ -1464,7 +1458,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
RTC_DCHECK(last_decoded_timestamps_.empty());
RTC_DCHECK(last_decoded_packet_infos_.empty());
// Do decoding.
@ -1484,7 +1477,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
auto opt_result = packet_list->front().frame->Decode(
rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
decoded_buffer_length_ - *decoded_length));
last_decoded_timestamps_.push_back(packet_list->front().timestamp);
last_decoded_packet_infos_.push_back(
std::move(packet_list->front().packet_info));
packet_list->pop_front();

3
modules/audio_coding/neteq/neteq_impl.h
View File

@ -194,8 +194,6 @@ class NetEqImpl : public webrtc::NetEq {
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
std::vector<uint32_t> LastDecodedTimestamps() const override;
int SyncBufferSizeMs() const override;
// This accessor method is only intended for testing purposes.
@ -395,7 +393,6 @@ class NetEqImpl : public webrtc::NetEq {
AudioFrame::kVadPassive;
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
RTC_GUARDED_BY(mutex_);
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(mutex_);
std::vector<RtpPacketInfo> last_decoded_packet_infos_ RTC_GUARDED_BY(mutex_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(mutex_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);

61
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -842,67 +842,6 @@ TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
EXPECT_FALSE(muted);
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsEmpty) {
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsOneDecoded) {
// Insert one packet with PCM16b WB data (this is what PopulateRtpInfo does by
// default). Make the length 10 ms.
constexpr size_t kPayloadSamples = 16 * 10;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp}),
neteq_->LastDecodedTimestamps());
// Nothing decoded on the second call.
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsTwoDecoded) {
// Insert two packets with PCM16b WB data (this is what PopulateRtpInfo does
// by default). Make the length 5 ms so that NetEq must decode them both in
// the same GetAudio call.
constexpr size_t kPayloadSamples = 16 * 5;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp1 = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp1, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + kPayloadSamples;
PopulateRtpInfo(1, kRtpTimestamp2, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
neteq_->LastDecodedTimestamps());
}
TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
const int kNumConcealmentEvents = 19;
const size_t kSamples = 10 * 16;

8
modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
View File

@ -102,12 +102,8 @@ void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
bool /*muted*/,
NetEq* neteq) {
get_audio_time_ms_.push_back(time_now_ms);
// Check what timestamps were decoded in the last GetAudio call.
std::vector<uint32_t> dec_ts = neteq->LastDecodedTimestamps();
// Find those timestamps in data_, insert their decoding time and sync
// delay.
for (uint32_t ts : dec_ts) {
auto it = data_.find(ts);
for (const RtpPacketInfo& info : audio_frame.packet_infos_) {
auto it = data_.find(info.rtp_timestamp());
if (it == data_.end()) {
// This is a packet that was split out from another packet. Skip it.
continue;