Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData
It appears unused everywhere. It will be deleted in a followup cl.
Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 63d61cf..e8360cb 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -185,8 +185,7 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) override;
+ size_t payloadSize) override;
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
@@ -196,15 +195,13 @@
int32_t SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload,
- const RTPFragmentationHeader* fragmentation)
+ rtc::ArrayView<const uint8_t> payload)
RTC_RUN_ON(encoder_queue_);
int32_t SendMediaTransportAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload,
- const RTPFragmentationHeader* fragmentation)
+ rtc::ArrayView<const uint8_t> payload)
RTC_RUN_ON(encoder_queue_);
// Return media transport or nullptr if using RTP.
@@ -477,8 +474,7 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) {
+ size_t payloadSize) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
@@ -489,19 +485,16 @@
return 0;
}
- return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload,
- fragmentation);
+ return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload);
} else {
- return SendRtpAudio(frameType, payloadType, timeStamp, payload,
- fragmentation);
+ return SendRtpAudio(frameType, payloadType, timeStamp, payload);
}
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload,
- const RTPFragmentationHeader* fragmentation) {
+ rtc::ArrayView<const uint8_t> payload) {
if (_includeAudioLevelIndication) {
// Store current audio level in the RTP sender.
// The level will be used in combination with voice-activity state
@@ -572,8 +565,7 @@
AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload,
- const RTPFragmentationHeader* fragmentation) {
+ rtc::ArrayView<const uint8_t> payload) {
// TODO(nisse): Use null _transportPtr for MediaTransport.
// RTC_DCHECK(_transportPtr == nullptr);
uint64_t channel_id;
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 747d4a3..780026d 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -107,8 +107,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) override {
+ size_t payload_len_bytes) override {
if (frame_type == AudioFrameType::kEmptyFrame)
return 0;
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index c558f7b..55552ca 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -122,13 +122,11 @@
}
// This method receives the callback from ACM when a new packet is produced.
-int32_t AcmSendTestOldApi::SendData(
- AudioFrameType frame_type,
- uint8_t payload_type,
- uint32_t timestamp,
- const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) {
+int32_t AcmSendTestOldApi::SendData(AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes) {
// Store the packet locally.
frame_type_ = frame_type;
payload_type_ = payload_type;
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index 744d015..f4a6fc4 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -54,8 +54,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) override;
+ size_t payload_len_bytes) override;
AudioCodingModule* acm() { return acm_.get(); }
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b5c5973..0dc4fcf 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -282,28 +282,6 @@
return 0;
}
-void ConvertEncodedInfoToFragmentationHeader(
- const AudioEncoder::EncodedInfo& info,
- RTPFragmentationHeader* frag) {
- if (info.redundant.empty()) {
- frag->fragmentationVectorSize = 0;
- return;
- }
-
- frag->VerifyAndAllocateFragmentationHeader(
- static_cast<uint16_t>(info.redundant.size()));
- frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size());
- size_t offset = 0;
- for (size_t i = 0; i < info.redundant.size(); ++i) {
- frag->fragmentationOffset[i] = offset;
- offset += info.redundant[i].encoded_bytes;
- frag->fragmentationLength[i] = info.redundant[i].encoded_bytes;
- frag->fragmentationTimeDiff[i] = rtc::dchecked_cast<uint16_t>(
- info.encoded_timestamp - info.redundant[i].encoded_timestamp);
- frag->fragmentationPlType[i] = info.redundant[i].payload_type;
- }
-}
-
void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
if (value != last_value_ || first_time_) {
first_time_ = false;
@@ -391,8 +369,6 @@
}
}
- RTPFragmentationHeader my_fragmentation;
- ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
AudioFrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
frame_type = AudioFrameType::kEmptyFrame;
@@ -408,9 +384,7 @@
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
- encode_buffer_.data(), encode_buffer_.size(),
- my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
- : nullptr);
+ encode_buffer_.data(), encode_buffer_.size());
}
if (vad_callback_) {
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 8e85249..68ed6b6 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -108,8 +108,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) override {
+ size_t payload_len_bytes) override {
rtc::CritScope lock(&crit_sect_);
++num_calls_;
last_frame_type_ = frame_type;
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 8e0c4d5..17ad71d 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -44,8 +44,23 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
+ size_t payload_len_bytes) {
+ return SendData(frame_type, payload_type, timestamp, payload_data,
+ payload_len_bytes, nullptr);
+ }
+
+ // TODO(bugs.webrtc.org/6471) Deprecated, delete as soon as downstream
+ // implementations are updated. Then make above method pure virtual, and
+ // delete forward declaration of RTPFragmentationHeader.
+ virtual int32_t SendData(AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) = 0;
+ const RTPFragmentationHeader* fragmentation) {
+ return SendData(frame_type, payload_type, timestamp, payload_data,
+ payload_len_bytes);
+ }
};
// Callback class used for reporting VAD decision
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index 443dfd8..a75802b 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -111,9 +111,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_len_bytes,
- const RTPFragmentationHeader* fragmentation) override {
- RTC_CHECK(!fragmentation);
+ size_t payload_len_bytes) override {
if (payload_len_bytes == 0) {
return 0;
}
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index ba9c95e..8ad0e00 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -22,8 +22,7 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) {
+ size_t payloadSize) {
RTPHeader rtp_header;
int32_t status;
size_t payloadDataSize = payloadSize;
@@ -46,46 +45,14 @@
return 0;
}
- // Treat fragmentation separately
- if (fragmentation != NULL) {
- // If silence for too long, send only new data.
- if ((fragmentation->fragmentationVectorSize == 2) &&
- (fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
- // only 0x80 if we have multiple blocks
- _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
- size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
- fragmentation->fragmentationLength[1];
- _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
- _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
- _payloadData[3] = uint8_t(REDheader & 0x000000FF);
-
- _payloadData[4] = fragmentation->fragmentationPlType[0];
- // copy the RED data
- memcpy(_payloadData + 5,
- payloadData + fragmentation->fragmentationOffset[1],
- fragmentation->fragmentationLength[1]);
- // copy the normal data
- memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
- payloadDataSize += 5;
+ memcpy(_payloadData, payloadData, payloadDataSize);
+ if (_isStereo) {
+ if (_leftChannel) {
+ _rtp_header = rtp_header;
+ _leftChannel = false;
} else {
- // single block (newest one)
- memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
- payloadDataSize = fragmentation->fragmentationLength[0];
- rtp_header.payloadType = fragmentation->fragmentationPlType[0];
- }
- } else {
- memcpy(_payloadData, payloadData, payloadDataSize);
- if (_isStereo) {
- if (_leftChannel) {
- _rtp_header = rtp_header;
- _leftChannel = false;
- } else {
- rtp_header = _rtp_header;
- _leftChannel = true;
- }
+ rtp_header = _rtp_header;
+ _leftChannel = true;
}
}
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index 6a55b06..0b248c8 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -51,8 +51,7 @@
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) override;
+ size_t payloadSize) override;
void RegisterReceiverACM(AudioCodingModule* acm);
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index c961fe5..25e273a 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -32,13 +32,11 @@
TestPacketization::~TestPacketization() {
}
-int32_t TestPacketization::SendData(
- const AudioFrameType /* frameType */,
- const uint8_t payloadType,
- const uint32_t timeStamp,
- const uint8_t* payloadData,
- const size_t payloadSize,
- const RTPFragmentationHeader* /* fragmentation */) {
+int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
+ const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const uint8_t* payloadData,
+ const size_t payloadSize) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
_frequency);
return 1;
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 6dc7bc9..ec95766 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -32,8 +32,7 @@
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
- const size_t payloadSize,
- const RTPFragmentationHeader* fragmentation) override;
+ const size_t payloadSize) override;
private:
static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index ad742ce..a3f0964 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -64,8 +64,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) {
+ size_t payload_size) {
RTPHeader rtp_header;
int32_t status;
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index d8a7711..ef56661 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -29,8 +29,7 @@
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) override;
+ size_t payload_size) override;
size_t payload_size();
uint32_t timestamp_diff();
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 9564af7e..ea8735b 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -44,8 +44,7 @@
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
- const size_t payload_size,
- const RTPFragmentationHeader* fragmentation) {
+ const size_t payload_size) {
RTPHeader rtp_header;
int32_t status = 0;
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index 9a44a10..e950840 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -35,8 +35,7 @@
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
- const size_t payload_size,
- const RTPFragmentationHeader* fragmentation) override;
+ const size_t payload_size) override;
uint16_t payload_size();
uint32_t timestamp_diff();
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 55f7af0..38b2362 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -316,7 +316,7 @@
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
- rtp_timestamp_, bitstream, bitstream_len_byte, NULL);
+ rtp_timestamp_, bitstream, bitstream_len_byte);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;
