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Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
View File

@ -113,10 +113,9 @@ void AcmReceiveTest::Run() {
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_TRUE(
acm_->InsertPacket(packet->payload(),
static_cast<int32_t>(packet->payload_length_bytes()),
header))
EXPECT_TRUE(acm_->InsertPacket(packet->payload(),
packet->payload_length_bytes(),
header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl

2
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
View File

@ -261,7 +261,7 @@ AudioPlayoutMode AcmReceiver::PlayoutMode() const {
int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* incoming_payload,
int length_payload) {
size_t length_payload) {
uint32_t receive_timestamp = 0;
InitialDelayManager::PacketType packet_type =
InitialDelayManager::kUndefinedPacket;

2
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@ -67,7 +67,7 @@ class AcmReceiver {
//
int InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* incoming_payload,
int length_payload);
size_t length_payload);
//
// Asks NetEq for 10 milliseconds of decoded audio.

4
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
View File

@ -115,12 +115,12 @@ class AcmReceiverTest : public AudioPacketizationCallback,
}
}
virtual int SendData(
virtual int32_t SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
if (frame_type == kFrameEmpty)
return 0;

2
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -124,7 +124,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
if (frame_type == kFrameEmpty)
return 0;

2
webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
View File

@ -94,7 +94,7 @@ int32_t AcmSendTest::SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;

2
webrtc/modules/audio_coding/main/acm2/acm_send_test.h
View File

@ -49,7 +49,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:

2
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
View File

@ -98,7 +98,7 @@ int32_t AcmSendTestOldApi::SendData(
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;

2
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
View File

@ -51,7 +51,7 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:

27
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -314,7 +314,7 @@ int AudioCodingModuleImpl::EncodeFragmentation(int fragmentation_index,
int AudioCodingModuleImpl::ProcessDualStream() {
uint8_t stream[kMaxNumFragmentationVectors * MAX_PAYLOAD_SIZE_BYTE];
uint32_t current_timestamp;
int16_t length_bytes = 0;
size_t length_bytes = 0;
RTPFragmentationHeader my_fragmentation;
uint8_t my_red_payload_type;
@ -336,8 +336,7 @@ int AudioCodingModuleImpl::ProcessDualStream() {
// Nothing to send.
return 0;
}
int len_bytes_previous_secondary = static_cast<int>(
fragmentation_.fragmentationLength[2]);
size_t len_bytes_previous_secondary = fragmentation_.fragmentationLength[2];
assert(len_bytes_previous_secondary <= MAX_PAYLOAD_SIZE_BYTE);
bool has_previous_payload = len_bytes_previous_secondary > 0;
@ -1689,13 +1688,8 @@ int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const int payload_length,
const size_t payload_length,
const WebRtcRTPHeader& rtp_header) {
if (payload_length < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"IncomingPacket() Error, payload-length cannot be negative");
return -1;
}
int last_audio_pltype = receiver_.last_audio_payload_type();
if (receiver_.InsertPacket(rtp_header, incoming_payload, payload_length) <
0) {
@ -1797,16 +1791,9 @@ int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
// TODO(tlegrand): Modify this function to work for stereo, and add tests.
int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
int payload_length,
size_t payload_length,
uint8_t payload_type,
uint32_t timestamp) {
if (payload_length < 0) {
// Log error in trace file.
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"IncomingPacket() Error, payload-length cannot be negative");
return -1;
}
// We are not acquiring any lock when interacting with |aux_rtp_header_| no
// other method uses this member variable.
if (aux_rtp_header_ == NULL) {
@ -1960,7 +1947,7 @@ int AudioCodingModuleImpl::REDPayloadISAC(int isac_rate,
}
void AudioCodingModuleImpl::ResetFragmentation(int vector_size) {
for (int n = 0; n < kMaxNumFragmentationVectors; n++) {
for (size_t n = 0; n < kMaxNumFragmentationVectors; n++) {
fragmentation_.fragmentationOffset[n] = n * MAX_PAYLOAD_SIZE_BYTE;
}
memset(fragmentation_.fragmentationLength, 0, kMaxNumFragmentationVectors *
@ -2116,14 +2103,14 @@ bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type,
}
bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload,
int32_t payload_len_bytes,
size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) {
return acm_old_->IncomingPacket(
incoming_payload, payload_len_bytes, rtp_info) == 0;
}
bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload,
int32_t payload_len_byte,
size_t payload_len_byte,
uint8_t payload_type,
uint32_t timestamp) {
FATAL() << "Not implemented yet.";

8
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -156,13 +156,13 @@ class AudioCodingModuleImpl : public AudioCodingModule {
// Incoming packet from network parsed and ready for decode.
virtual int IncomingPacket(const uint8_t* incoming_payload,
int payload_length,
const size_t payload_length,
const WebRtcRTPHeader& rtp_info) OVERRIDE;
// Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
// One usage for this API is when pre-encoded files are pushed in ACM.
virtual int IncomingPayload(const uint8_t* incoming_payload,
int payload_length,
const size_t payload_length,
uint8_t payload_type,
uint32_t timestamp) OVERRIDE;
@ -423,11 +423,11 @@ class AudioCodingImpl : public AudioCoding {
uint8_t payload_type) OVERRIDE;
virtual bool InsertPacket(const uint8_t* incoming_payload,
int32_t payload_len_bytes,
size_t payload_len_bytes,
const WebRtcRTPHeader& rtp_info) OVERRIDE;
virtual bool InsertPayload(const uint8_t* incoming_payload,
int32_t payload_len_byte,
size_t payload_len_byte,
uint8_t payload_type,
uint32_t timestamp) OVERRIDE;

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -42,7 +42,7 @@ const int kSampleRateHz = 16000;
const int kNumSamples10ms = kSampleRateHz / 100;
const int kFrameSizeMs = 10; // Multiple of 10.
const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const size_t kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
const uint8_t kPayloadType = 111;
class RtpUtility {
@ -87,7 +87,7 @@ class PacketizationCallbackStub : public AudioPacketizationCallback {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;

2
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -87,7 +87,7 @@ class PacketizationCallbackStub : public AudioPacketizationCallback {
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
uint16_t payload_len_bytes,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
++num_calls_;