Update ACM to use RTPHeader instead of WebRtcRTPHeader
Bug: webrtc:5876
Change-Id: Id3311dcf508cca34495349197eeac2edf8783772
Reviewed-on: https://webrtc-review.googlesource.com/c/123188
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26729}diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index b899dcf..0e218ed 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -57,36 +57,19 @@
constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
-webrtc::FrameType WebrtcFrameTypeForMediaTransportFrameType(
- MediaTransportEncodedAudioFrame::FrameType frame_type) {
- switch (frame_type) {
- case MediaTransportEncodedAudioFrame::FrameType::kSpeech:
- return kAudioFrameSpeech;
- break;
-
- case MediaTransportEncodedAudioFrame::FrameType::
- kDiscountinuousTransmission:
- return kAudioFrameCN;
- break;
- }
-}
-
-WebRtcRTPHeader CreateWebrtcRTPHeaderForMediaTransportFrame(
+RTPHeader CreateRTPHeaderForMediaTransportFrame(
const MediaTransportEncodedAudioFrame& frame,
uint64_t channel_id) {
- webrtc::WebRtcRTPHeader webrtc_header = {};
- webrtc_header.header.payloadType = frame.payload_type();
- webrtc_header.header.payload_type_frequency = frame.sampling_rate_hz();
- webrtc_header.header.timestamp = frame.starting_sample_index();
- webrtc_header.header.sequenceNumber = frame.sequence_number();
+ webrtc::RTPHeader rtp_header;
+ rtp_header.payloadType = frame.payload_type();
+ rtp_header.payload_type_frequency = frame.sampling_rate_hz();
+ rtp_header.timestamp = frame.starting_sample_index();
+ rtp_header.sequenceNumber = frame.sequence_number();
- webrtc_header.frameType =
- WebrtcFrameTypeForMediaTransportFrameType(frame.frame_type());
-
- webrtc_header.header.ssrc = static_cast<uint32_t>(channel_id);
+ rtp_header.ssrc = static_cast<uint32_t>(channel_id);
// The rest are initialized by the RTPHeader constructor.
- return webrtc_header;
+ return rtp_header;
}
class ChannelReceive : public ChannelReceiveInterface,
@@ -189,7 +172,7 @@
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
- const WebRtcRTPHeader* rtpHeader);
+ const RTPHeader& rtpHeader);
bool Playing() const {
rtc::CritScope lock(&playing_lock_);
@@ -277,10 +260,9 @@
webrtc::CryptoOptions crypto_options_;
};
-int32_t ChannelReceive::OnReceivedPayloadData(
- const uint8_t* payloadData,
- size_t payloadSize,
- const WebRtcRTPHeader* rtpHeader) {
+int32_t ChannelReceive::OnReceivedPayloadData(const uint8_t* payloadData,
+ size_t payloadSize,
+ const RTPHeader& rtp_header) {
// We should not be receiving any RTP packets if media_transport is set.
RTC_CHECK(!media_transport_);
@@ -291,7 +273,7 @@
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
- if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
+ if (audio_coding_->IncomingPacket(payloadData, payloadSize, rtp_header) !=
0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
"push data to the ACM";
@@ -324,8 +306,7 @@
// Send encoded audio frame to Decoder / NetEq.
if (audio_coding_->IncomingPacket(
frame.encoded_data().data(), frame.encoded_data().size(),
- CreateWebrtcRTPHeaderForMediaTransportFrame(frame, channel_id)) !=
- 0) {
+ CreateRTPHeaderForMediaTransportFrame(frame, channel_id)) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
"push data to the ACM";
}
@@ -637,8 +618,6 @@
const uint8_t* payload = packet + header.headerLength;
assert(packet_length >= header.headerLength);
size_t payload_length = packet_length - header.headerLength;
- WebRtcRTPHeader webrtc_rtp_header = {};
- webrtc_rtp_header.header = header;
size_t payload_data_length = payload_length - header.paddingLength;
@@ -677,11 +656,9 @@
}
if (payload_data_length == 0) {
- webrtc_rtp_header.frameType = kEmptyFrame;
- return OnReceivedPayloadData(nullptr, 0, &webrtc_rtp_header);
+ return OnReceivedPayloadData(nullptr, 0, header);
}
- return OnReceivedPayloadData(payload, payload_data_length,
- &webrtc_rtp_header);
+ return OnReceivedPayloadData(payload, payload_data_length, header);
}
// May be called on either worker thread or network thread.
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 76b0506..078c991 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -120,18 +120,15 @@
AfterGetAudio();
}
- // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
- WebRtcRTPHeader header;
- header.header = packet->header();
- header.frameType = kAudioFrameSpeech;
- EXPECT_EQ(0,
- acm_->IncomingPacket(
- packet->payload(),
- static_cast<int32_t>(packet->payload_length_bytes()), header))
+ EXPECT_EQ(0, acm_->IncomingPacket(
+ packet->payload(),
+ static_cast<int32_t>(packet->payload_length_bytes()),
+ packet->header()))
<< "Failure when inserting packet:" << std::endl
- << " PT = " << static_cast<int>(header.header.payloadType) << std::endl
- << " TS = " << header.header.timestamp << std::endl
- << " SN = " << header.header.sequenceNumber;
+ << " PT = " << static_cast<int>(packet->header().payloadType)
+ << std::endl
+ << " TS = " << packet->header().timestamp << std::endl
+ << " SN = " << packet->header().sequenceNumber;
}
}
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 9eb59ec..d296a57 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -78,15 +78,14 @@
return neteq_->last_output_sample_rate_hz();
}
-int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
+int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> incoming_payload) {
if (incoming_payload.empty()) {
- neteq_->InsertEmptyPacket(rtp_header.header);
+ neteq_->InsertEmptyPacket(rtp_header);
return 0;
}
- const RTPHeader& header = rtp_header.header; // Just a shorthand.
- int payload_type = header.payloadType;
+ int payload_type = rtp_header.payloadType;
auto format = neteq_->GetDecoderFormat(payload_type);
if (format && absl::EqualsIgnoreCase(format->name, "red")) {
// This is a RED packet. Get the format of the audio codec.
@@ -115,9 +114,10 @@
} // |crit_sect_| is released.
uint32_t receive_timestamp = NowInTimestamp(format->clockrate_hz);
- if (neteq_->InsertPacket(header, incoming_payload, receive_timestamp) < 0) {
+ if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
+ 0) {
RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
- << static_cast<int>(header.payloadType)
+ << static_cast<int>(rtp_header.payloadType)
<< " Failed to insert packet";
return -1;
}
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 3e30b86..1f449a3 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -33,7 +33,6 @@
class Clock;
class NetEq;
struct RTPHeader;
-struct WebRtcRTPHeader;
namespace acm2 {
@@ -58,7 +57,7 @@
// Return value : 0 if OK.
// <0 if NetEq returned an error.
//
- int InsertPacket(const WebRtcRTPHeader& rtp_header,
+ int InsertPacket(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> incoming_payload);
//
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index d5392dc..e5a7684 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -50,13 +50,12 @@
acm_->InitializeReceiver();
acm_->RegisterTransportCallback(this);
- rtp_header_.header.sequenceNumber = 0;
- rtp_header_.header.timestamp = 0;
- rtp_header_.header.markerBit = false;
- rtp_header_.header.ssrc = 0x12345678; // Arbitrary.
- rtp_header_.header.numCSRCs = 0;
- rtp_header_.header.payloadType = 0;
- rtp_header_.frameType = kAudioFrameSpeech;
+ rtp_header_.sequenceNumber = 0;
+ rtp_header_.timestamp = 0;
+ rtp_header_.markerBit = false;
+ rtp_header_.ssrc = 0x12345678; // Arbitrary.
+ rtp_header_.numCSRCs = 0;
+ rtp_header_.payloadType = 0;
}
void TearDown() override {}
@@ -113,9 +112,8 @@
if (frame_type == kEmptyFrame)
return 0;
- rtp_header_.header.payloadType = payload_type;
- rtp_header_.frameType = frame_type;
- rtp_header_.header.timestamp = timestamp;
+ rtp_header_.payloadType = payload_type;
+ rtp_header_.timestamp = timestamp;
int ret_val = receiver_->InsertPacket(
rtp_header_,
@@ -124,7 +122,7 @@
assert(false);
return -1;
}
- rtp_header_.header.sequenceNumber++;
+ rtp_header_.sequenceNumber++;
packet_sent_ = true;
last_frame_type_ = frame_type;
return 0;
@@ -137,7 +135,7 @@
AudioCodingModule::Config config_;
std::unique_ptr<AcmReceiver> receiver_;
std::unique_ptr<AudioCodingModule> acm_;
- WebRtcRTPHeader rtp_header_;
+ RTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
uint32_t last_packet_send_timestamp_;
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 9e4ac89..014aede 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -92,7 +92,7 @@
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
- const WebRtcRTPHeader& rtp_info) override;
+ const RTPHeader& rtp_info) override;
// Minimum playout delay.
int SetMinimumPlayoutDelay(int time_ms) override;
@@ -688,7 +688,7 @@
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
- const WebRtcRTPHeader& rtp_header) {
+ const RTPHeader& rtp_header) {
RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
return receiver_.InsertPacket(
rtp_header,
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 5037bc6..c9a03a1 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -71,21 +71,20 @@
virtual ~RtpUtility() {}
- void Populate(WebRtcRTPHeader* rtp_header) {
- rtp_header->header.sequenceNumber = 0xABCD;
- rtp_header->header.timestamp = 0xABCDEF01;
- rtp_header->header.payloadType = payload_type_;
- rtp_header->header.markerBit = false;
- rtp_header->header.ssrc = 0x1234;
- rtp_header->header.numCSRCs = 0;
- rtp_header->frameType = kAudioFrameSpeech;
+ void Populate(RTPHeader* rtp_header) {
+ rtp_header->sequenceNumber = 0xABCD;
+ rtp_header->timestamp = 0xABCDEF01;
+ rtp_header->payloadType = payload_type_;
+ rtp_header->markerBit = false;
+ rtp_header->ssrc = 0x1234;
+ rtp_header->numCSRCs = 0;
- rtp_header->header.payload_type_frequency = kSampleRateHz;
+ rtp_header->payload_type_frequency = kSampleRateHz;
}
- void Forward(WebRtcRTPHeader* rtp_header) {
- ++rtp_header->header.sequenceNumber;
- rtp_header->header.timestamp += samples_per_packet_;
+ void Forward(RTPHeader* rtp_header) {
+ ++rtp_header->sequenceNumber;
+ rtp_header->timestamp += samples_per_packet_;
}
private:
@@ -237,7 +236,7 @@
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
- WebRtcRTPHeader rtp_header_;
+ RTPHeader rtp_header_;
AudioFrame input_frame_;
absl::optional<SdpAudioFormat> audio_format_;
@@ -792,16 +791,15 @@
++receive_packet_count_;
// Encode new frame.
- uint32_t input_timestamp = rtp_header_.header.timestamp;
+ uint32_t input_timestamp = rtp_header_.timestamp;
while (info.encoded_bytes == 0) {
info = isac_encoder_->Encode(input_timestamp,
audio_loop_.GetNextBlock(), &encoded);
input_timestamp += 160; // 10 ms at 16 kHz.
}
- EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
- input_timestamp);
- EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
- EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
+ EXPECT_EQ(rtp_header_.timestamp + kPacketSizeSamples, input_timestamp);
+ EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp);
+ EXPECT_EQ(rtp_header_.payloadType, info.payload_type);
}
// Now we're not holding the crit sect when calling ACM.
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 7d11d70..d5af571 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -21,17 +21,19 @@
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/deprecation.h"
#include "rtc_base/function_view.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
// forward declarations
-struct WebRtcRTPHeader;
class AudioDecoder;
class AudioEncoder;
class AudioFrame;
class RTPFragmentationHeader;
+struct RTPHeader;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
@@ -246,7 +248,13 @@
//
virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_len_bytes,
- const WebRtcRTPHeader& rtp_info) = 0;
+ const RTPHeader& rtp_header) = 0;
+ RTC_DEPRECATED
+ int32_t IncomingPacket(const uint8_t* incoming_payload,
+ const size_t payload_len_bytes,
+ const WebRtcRTPHeader& rtp_info) {
+ return IncomingPacket(incoming_payload, payload_len_bytes, rtp_info.header);
+ }
///////////////////////////////////////////////////////////////////////////
// int SetMinimumPlayoutDelay()
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 4cb6c35..e54a29a 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -116,7 +116,8 @@
return 0;
}
- status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
+ status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize,
+ rtpInfo.header);
return status;
}
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index fbbc9d3..cd57ecd 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -168,7 +168,7 @@
}
EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
- _rtpInfo));
+ _rtpInfo.header));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 000041b..fd76224 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -57,7 +57,8 @@
}
if (!PacketLost()) {
- _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
+ _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
+ _rtpInfo.header);
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 1d0ddf8..141075b 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -15,6 +15,7 @@
#include <queue>
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 75ba60a..81b83c0 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -66,14 +66,14 @@
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
- WebRtcRTPHeader rtp_info;
+ RTPHeader rtp_header;
int32_t status;
- rtp_info.header.markerBit = false;
- rtp_info.header.ssrc = 0;
- rtp_info.header.sequenceNumber = sequence_number_++;
- rtp_info.header.payloadType = payload_type;
- rtp_info.header.timestamp = timestamp;
+ rtp_header.markerBit = false;
+ rtp_header.ssrc = 0;
+ rtp_header.sequenceNumber = sequence_number_++;
+ rtp_header.payloadType = payload_type;
+ rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
// Skip this frame.
@@ -83,7 +83,8 @@
// Only run mono for all test cases.
memcpy(payload_data_, payload_data, payload_size);
- status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
+ status =
+ receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header);
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 70065fb..2c71f46 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -46,14 +46,14 @@
const uint8_t* payload_data,
const size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
- WebRtcRTPHeader rtp_info;
+ RTPHeader rtp_header;
int32_t status = 0;
- rtp_info.header.markerBit = false;
- rtp_info.header.ssrc = 0;
- rtp_info.header.sequenceNumber = seq_no_++;
- rtp_info.header.payloadType = payload_type;
- rtp_info.header.timestamp = timestamp;
+ rtp_header.markerBit = false;
+ rtp_header.ssrc = 0;
+ rtp_header.sequenceNumber = seq_no_++;
+ rtp_header.payloadType = payload_type;
+ rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
// Skip this frame
return 0;
@@ -61,7 +61,7 @@
if (lost_packet_ == false) {
status =
- receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_info);
+ receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
if (frame_type != kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index c972e62..6f7c6cf 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -37,12 +37,11 @@
{{pltype, {"L16", kSampleRateHz, 1}}};
acm_->SetReceiveCodecs(receive_codecs);
- rtp_info_.header.payloadType = pltype;
- rtp_info_.header.timestamp = 0;
- rtp_info_.header.ssrc = 0x12345678;
- rtp_info_.header.markerBit = false;
- rtp_info_.header.sequenceNumber = 0;
- rtp_info_.frameType = kAudioFrameSpeech;
+ rtp_header_.payloadType = pltype;
+ rtp_header_.timestamp = 0;
+ rtp_header_.ssrc = 0x12345678;
+ rtp_header_.markerBit = false;
+ rtp_header_.sequenceNumber = 0;
int16_t audio[kFrameSizeSamples];
const int kRange = 0x7FF; // 2047, easy for masking.
@@ -98,10 +97,10 @@
static const int kInterarrivalJitterPacket = 2;
void Push() {
- rtp_info_.header.timestamp += kFrameSizeSamples;
- rtp_info_.header.sequenceNumber++;
- ASSERT_EQ(0,
- acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_info_));
+ rtp_header_.timestamp += kFrameSizeSamples;
+ rtp_header_.sequenceNumber++;
+ ASSERT_EQ(
+ 0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_header_));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@@ -150,7 +149,7 @@
}
std::unique_ptr<AudioCodingModule> acm_;
- WebRtcRTPHeader rtp_info_;
+ RTPHeader rtp_header_;
uint8_t payload_[kPayloadLenBytes];
};
