In ChannelReceive, use AcmReceiver directly, not AudioCodingModule
Bug: webrtc:9801
Change-Id: I02d76bc89c363247c8dc782db316a9f87a2b93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/111504
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28766}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index cf8af3d..3cbfa14 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -59,6 +59,7 @@
"../logging:rtc_event_log_api",
"../logging:rtc_stream_config",
"../modules/audio_coding",
+ "../modules/audio_coding:audio_coding_module_typedefs",
"../modules/audio_coding:audio_encoder_cng",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/audio_device",
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index aa60438..2554476 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -23,8 +23,8 @@
#include "audio/utility/audio_frame_operations.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
@@ -77,6 +77,21 @@
return rtp_header;
}
+AudioCodingModule::Config AcmConfig(
+ rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+ absl::optional<AudioCodecPairId> codec_pair_id,
+ size_t jitter_buffer_max_packets,
+ bool jitter_buffer_fast_playout) {
+ AudioCodingModule::Config acm_config;
+ acm_config.decoder_factory = decoder_factory;
+ acm_config.neteq_config.codec_pair_id = codec_pair_id;
+ acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
+ acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
+ acm_config.neteq_config.enable_muted_state = true;
+
+ return acm_config;
+}
+
class ChannelReceive : public ChannelReceiveInterface,
public MediaTransportAudioSinkInterface {
public:
@@ -167,7 +182,7 @@
}
private:
- bool ReceivePacket(const uint8_t* packet,
+ void ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header);
int ResendPackets(const uint16_t* sequence_numbers, int length);
@@ -180,9 +195,8 @@
void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) override;
- int32_t OnReceivedPayloadData(const uint8_t* payloadData,
- size_t payloadSize,
- const RTPHeader& rtpHeader);
+ void OnReceivedPayloadData(rtc::ArrayView<const uint8_t> payload,
+ const RTPHeader& rtpHeader);
bool Playing() const {
rtc::CritScope lock(&playing_lock_);
@@ -224,7 +238,8 @@
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(&sync_info_lock_);
- std::unique_ptr<AudioCodingModule> audio_coding_;
+ // The AcmReceiver is thread safe, using its own lock.
+ acm2::AcmReceiver acm_receiver_;
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
@@ -269,36 +284,34 @@
const bool use_standard_bytes_stats_;
};
-int32_t ChannelReceive::OnReceivedPayloadData(const uint8_t* payloadData,
- size_t payloadSize,
- const RTPHeader& rtp_header) {
+void ChannelReceive::OnReceivedPayloadData(
+ rtc::ArrayView<const uint8_t> payload,
+ const RTPHeader& rtpHeader) {
// We should not be receiving any RTP packets if media_transport is set.
RTC_CHECK(!media_transport());
if (!Playing()) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
- return 0;
+ return;
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
- if (audio_coding_->IncomingPacket(payloadData, payloadSize, rtp_header) !=
- 0) {
+ if (acm_receiver_.InsertPacket(rtpHeader, payload) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
"push data to the ACM";
- return -1;
+ return;
}
int64_t round_trip_time = 0;
_rtpRtcpModule->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL);
- std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time);
+ std::vector<uint16_t> nack_list = acm_receiver_.GetNackList(round_trip_time);
if (!nack_list.empty()) {
// Can't use nack_list.data() since it's not supported by all
// compilers.
ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size()));
}
- return 0;
}
// MediaTransportAudioSinkInterface override.
@@ -313,9 +326,9 @@
}
// Send encoded audio frame to Decoder / NetEq.
- if (audio_coding_->IncomingPacket(
- frame.encoded_data().data(), frame.encoded_data().size(),
- CreateRTPHeaderForMediaTransportFrame(frame, channel_id)) != 0) {
+ if (acm_receiver_.InsertPacket(
+ CreateRTPHeaderForMediaTransportFrame(frame, channel_id),
+ frame.encoded_data()) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
"push data to the ACM";
}
@@ -331,8 +344,8 @@
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
- if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
- &muted) == -1) {
+ if (acm_receiver_.GetAudio(audio_frame->sample_rate_hz_, audio_frame,
+ &muted) == -1) {
RTC_DLOG(LS_ERROR)
<< "ChannelReceive::GetAudioFrame() PlayoutData10Ms() failed!";
// In all likelihood, the audio in this frame is garbage. We return an
@@ -414,8 +427,8 @@
{
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs",
- audio_coding_->TargetDelayMs());
- const int jitter_buffer_delay = audio_coding_->FilteredCurrentDelayMs();
+ acm_receiver_.TargetDelayMs());
+ const int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs();
rtc::CritScope lock(&video_sync_lock_);
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs",
jitter_buffer_delay + playout_delay_ms_);
@@ -432,8 +445,8 @@
int ChannelReceive::PreferredSampleRate() const {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
// Return the bigger of playout and receive frequency in the ACM.
- return std::max(audio_coding_->ReceiveFrequency(),
- audio_coding_->PlayoutFrequency());
+ return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
+ acm_receiver_.last_output_sample_rate_hz());
}
ChannelReceive::ChannelReceive(
@@ -455,6 +468,10 @@
: event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
remote_ssrc_(remote_ssrc),
+ acm_receiver_(AcmConfig(decoder_factory,
+ codec_pair_id,
+ jitter_buffer_max_packets,
+ jitter_buffer_fast_playout)),
_outputAudioLevel(),
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
@@ -476,16 +493,11 @@
RTC_DCHECK(module_process_thread);
RTC_DCHECK(audio_device_module);
- AudioCodingModule::Config acm_config;
- acm_config.decoder_factory = decoder_factory;
- acm_config.neteq_config.codec_pair_id = codec_pair_id;
- acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
- acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
- acm_config.neteq_config.min_delay_ms = jitter_buffer_min_delay_ms;
- acm_config.neteq_config.enable_muted_state = true;
- acm_config.neteq_config.enable_rtx_handling =
- jitter_buffer_enable_rtx_handling;
- audio_coding_.reset(AudioCodingModule::Create(acm_config));
+
+ acm_receiver_.ResetInitialDelay();
+ acm_receiver_.SetMinimumDelay(0);
+ acm_receiver_.SetMaximumDelay(0);
+ acm_receiver_.FlushBuffers();
_outputAudioLevel.ResetLevelFullRange();
@@ -527,9 +539,6 @@
StopPlayout();
- int error = audio_coding_->RegisterTransportCallback(NULL);
- RTC_DCHECK_EQ(0, error);
-
if (_moduleProcessThreadPtr)
_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
}
@@ -556,7 +565,7 @@
absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
- return audio_coding_->ReceiveCodec();
+ return acm_receiver_.LastDecoder();
}
void ChannelReceive::SetReceiveCodecs(
@@ -566,7 +575,7 @@
RTC_DCHECK_GE(kv.second.clockrate_hz, 1000);
payload_type_frequencies_[kv.first] = kv.second.clockrate_hz;
}
- audio_coding_->SetReceiveCodecs(codecs);
+ acm_receiver_.SetCodecs(codecs);
}
// May be called on either worker thread or network thread.
@@ -597,7 +606,7 @@
ReceivePacket(packet_copy.data(), packet_copy.size(), header);
}
-bool ChannelReceive::ReceivePacket(const uint8_t* packet,
+void ChannelReceive::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header) {
const uint8_t* payload = packet + header.headerLength;
@@ -638,10 +647,8 @@
payload_data_length = 0;
}
- if (payload_data_length == 0) {
- return OnReceivedPayloadData(nullptr, 0, header);
- }
- return OnReceivedPayloadData(payload, payload_data_length, header);
+ OnReceivedPayloadData(
+ rtc::ArrayView<const uint8_t>(payload, payload_data_length), header);
}
// May be called on either worker thread or network thread.
@@ -770,13 +777,12 @@
RTC_DCHECK(worker_thread_checker_.IsCurrent());
// None of these functions can fail.
if (enable) {
- rtp_receive_statistics_->SetMaxReorderingThreshold(remote_ssrc_,
- max_packets);
- audio_coding_->EnableNack(max_packets);
+ rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
+ acm_receiver_.EnableNack(max_packets);
} else {
rtp_receive_statistics_->SetMaxReorderingThreshold(
- remote_ssrc_, kDefaultMaxReorderingThreshold);
- audio_coding_->DisableNack();
+ kDefaultMaxReorderingThreshold);
+ acm_receiver_.DisableNack();
}
}
@@ -796,15 +802,14 @@
NetworkStatistics ChannelReceive::GetNetworkStatistics() const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
NetworkStatistics stats;
- int error = audio_coding_->GetNetworkStatistics(&stats);
- RTC_DCHECK_EQ(0, error);
+ acm_receiver_.GetNetworkStatistics(&stats);
return stats;
}
AudioDecodingCallStats ChannelReceive::GetDecodingCallStatistics() const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
AudioDecodingCallStats stats;
- audio_coding_->GetDecodingCallStatistics(&stats);
+ acm_receiver_.GetDecodingCallStatistics(&stats);
return stats;
}
@@ -812,7 +817,7 @@
RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
module_process_thread_checker_.IsCurrent());
rtc::CritScope lock(&video_sync_lock_);
- return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
+ return acm_receiver_.FilteredCurrentDelayMs() + playout_delay_ms_;
}
void ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) {
@@ -821,7 +826,7 @@
// close as possible, instead of failing.
delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs,
kVoiceEngineMaxMinPlayoutDelayMs);
- if (audio_coding_->SetMinimumPlayoutDelay(delay_ms) != 0) {
+ if (acm_receiver_.SetMinimumDelay(delay_ms) != 0) {
RTC_DLOG(LS_ERROR)
<< "SetMinimumPlayoutDelay() failed to set min playout delay";
}
@@ -836,11 +841,11 @@
}
bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
- return audio_coding_->SetBaseMinimumPlayoutDelayMs(delay_ms);
+ return acm_receiver_.SetBaseMinimumDelayMs(delay_ms);
}
int ChannelReceive::GetBaseMinimumPlayoutDelayMs() const {
- return audio_coding_->GetBaseMinimumPlayoutDelayMs();
+ return acm_receiver_.GetBaseMinimumDelayMs();
}
absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
@@ -863,7 +868,7 @@
}
void ChannelReceive::UpdatePlayoutTimestamp(bool rtcp) {
- jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp();
+ jitter_buffer_playout_timestamp_ = acm_receiver_.GetPlayoutTimestamp();
if (!jitter_buffer_playout_timestamp_) {
// This can happen if this channel has not received any RTP packets. In
@@ -895,14 +900,14 @@
}
int ChannelReceive::GetRtpTimestampRateHz() const {
- const auto decoder = audio_coding_->ReceiveCodec();
+ const auto decoder = acm_receiver_.LastDecoder();
// Default to the playout frequency if we've not gotten any packets yet.
// TODO(ossu): Zero clockrate can only happen if we've added an external
// decoder for a format we don't support internally. Remove once that way of
// adding decoders is gone!
return (decoder && decoder->second.clockrate_hz != 0)
? decoder->second.clockrate_hz
- : audio_coding_->PlayoutFrequency();
+ : acm_receiver_.last_output_sample_rate_hz();
}
int64_t ChannelReceive::GetRTT() const {
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index d0dea67..1fe64b9 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -27,7 +27,8 @@
#include "api/rtp_receiver_interface.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "system_wrappers/include/clock.h"
// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
// warnings about use of unsigned short.
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 6de45e7..6e7e631 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -228,7 +228,7 @@
return last_decoder_;
}
-void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
+void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) const {
NetEqNetworkStatistics neteq_stat;
// NetEq function always returns zero, so we don't check the return value.
neteq_->NetworkStatistics(&neteq_stat);
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 8d62312..8d54cf7 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -138,7 +138,7 @@
// Output:
// - statistics : The current network statistics.
//
- void GetNetworkStatistics(NetworkStatistics* statistics);
+ void GetNetworkStatistics(NetworkStatistics* statistics) const;
//
// Flushes the NetEq packet and speech buffers.
