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/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;