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Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData

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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}
This commit is contained in:
Niels Möller
2019-04-25 16:31:18 +02:00
committed by Commit Bot
parent e670fd9795
commit c35b6e675a
17 changed files with 54 additions and 121 deletions
Download Patch File Download Diff File Expand all files Collapse all files

49
modules/audio_coding/test/Channel.cc
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@ -22,8 +22,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
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 @@ int32_t Channel::SendData(AudioFrameType frameType,
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;
}
}

3
modules/audio_coding/test/Channel.h
View File

@ -51,8 +51,7 @@ class Channel : public AudioPacketizationCallback {
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);

12
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -32,13 +32,11 @@ TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
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;

3
modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -32,8 +32,7 @@ class TestPacketization : public AudioPacketizationCallback {
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,

3
modules/audio_coding/test/TestAllCodecs.cc
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@ -64,8 +64,7 @@ int32_t TestPack::SendData(AudioFrameType frame_type,
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;

3
modules/audio_coding/test/TestAllCodecs.h
View File

@ -29,8 +29,7 @@ class TestPack : public AudioPacketizationCallback {
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();

3
modules/audio_coding/test/TestStereo.cc
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@ -44,8 +44,7 @@ int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
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;

3
modules/audio_coding/test/TestStereo.h
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@ -35,8 +35,7 @@ class TestPackStereo : public AudioPacketizationCallback {
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();

2
modules/audio_coding/test/opus_test.cc
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@ -316,7 +316,7 @@ void OpusTest::Run(TestPackStereo* channel,
// 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_;