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WebRtc_Word32 => int32_t for rtp_rtcp/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-08 11:08:41 +00:00
parent 367804cce2
commit 2f44673d66
86 changed files with 3934 additions and 3956 deletions
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56
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
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@ -20,7 +20,7 @@
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id,
RTPReceiverAudio::RTPReceiverAudio(const int32_t id,
RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback)
: RTPReceiverStrategy(data_callback),
@ -41,7 +41,7 @@ RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id,
last_payload_.Audio.channels = 1;
}
WebRtc_UWord32 RTPReceiverAudio::AudioFrequency() const {
uint32_t RTPReceiverAudio::AudioFrequency() const {
CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
if (last_received_g722_) {
return 8000;
@ -64,13 +64,13 @@ bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
}
bool RTPReceiverAudio::TelephoneEventPayloadType(
const WebRtc_Word8 payload_type) const {
const int8_t payload_type) const {
CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
return (telephone_event_payload_type_ == payload_type) ? true : false;
}
bool RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payload_type,
WebRtc_UWord32* frequency,
bool RTPReceiverAudio::CNGPayloadType(const int8_t payload_type,
uint32_t* frequency,
bool* cng_payload_type_has_changed) {
CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
*cng_payload_type_has_changed = false;
@ -119,7 +119,7 @@ bool RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payload_type,
}
bool RTPReceiverAudio::ShouldReportCsrcChanges(
WebRtc_UWord8 payload_type) const {
uint8_t payload_type) const {
// Don't do this for DTMF packets, otherwise it's fine.
return !TelephoneEventPayloadType(payload_type);
}
@ -156,10 +156,10 @@ bool RTPReceiverAudio::ShouldReportCsrcChanges(
// - MPA frame N/A var. var.
// -
// - G7221 frame N/A
WebRtc_Word32 RTPReceiverAudio::OnNewPayloadTypeCreated(
int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payload_type,
const WebRtc_UWord32 frequency) {
const int8_t payload_type,
const uint32_t frequency) {
CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
if (ModuleRTPUtility::StringCompare(payload_name, "telephone-event", 15)) {
@ -183,21 +183,21 @@ WebRtc_Word32 RTPReceiverAudio::OnNewPayloadTypeCreated(
return 0;
}
WebRtc_Word32 RTPReceiverAudio::ParseRtpPacket(
int32_t RTPReceiverAudio::ParseRtpPacket(
WebRtcRTPHeader* rtp_header,
const ModuleRTPUtility::PayloadUnion& specific_payload,
const bool is_red,
const WebRtc_UWord8* packet,
const WebRtc_UWord16 packet_length,
const WebRtc_Word64 timestamp_ms,
const uint8_t* packet,
const uint16_t packet_length,
const int64_t timestamp_ms,
const bool is_first_packet) {
TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPReceiverAudio::ParseRtpPacket",
"seqnum", rtp_header->header.sequenceNumber,
"timestamp", rtp_header->header.timestamp);
const WebRtc_UWord8* payload_data =
const uint8_t* payload_data =
ModuleRTPUtility::GetPayloadData(rtp_header, packet);
const WebRtc_UWord16 payload_data_length =
const uint16_t payload_data_length =
ModuleRTPUtility::GetPayloadDataLength(rtp_header, packet_length);
return ParseAudioCodecSpecific(rtp_header,
@ -207,12 +207,12 @@ WebRtc_Word32 RTPReceiverAudio::ParseRtpPacket(
is_red);
}
WebRtc_Word32 RTPReceiverAudio::GetFrequencyHz() const {
int32_t RTPReceiverAudio::GetFrequencyHz() const {
return AudioFrequency();
}
RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
WebRtc_UWord16 last_payload_length) const {
uint16_t last_payload_length) const {
// Our CNG is 9 bytes; if it's a likely CNG the receiver needs to check
// kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG.
@ -224,7 +224,7 @@ RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
}
void RTPReceiverAudio::CheckPayloadChanged(
const WebRtc_Word8 payload_type,
const int8_t payload_type,
ModuleRTPUtility::PayloadUnion* specific_payload,
bool* should_reset_statistics,
bool* should_discard_changes) {
@ -251,10 +251,10 @@ void RTPReceiverAudio::CheckPayloadChanged(
}
}
WebRtc_Word32 RTPReceiverAudio::InvokeOnInitializeDecoder(
int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
RtpFeedback* callback,
const WebRtc_Word32 id,
const WebRtc_Word8 payload_type,
const int32_t id,
const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const ModuleRTPUtility::PayloadUnion& specific_payload) const {
if (-1 == callback->OnInitializeDecoder(id,
@ -274,10 +274,10 @@ WebRtc_Word32 RTPReceiverAudio::InvokeOnInitializeDecoder(
}
// We are not allowed to have any critsects when calling data_callback.
WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
WebRtcRTPHeader* rtp_header,
const WebRtc_UWord8* payload_data,
const WebRtc_UWord16 payload_length,
const uint8_t* payload_data,
const uint16_t payload_length,
const ModuleRTPUtility::AudioPayload& audio_specific,
const bool is_red) {
@ -300,7 +300,7 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
if (payload_length % 4 != 0) {
return -1;
}
WebRtc_UWord8 number_of_events = payload_length / 4;
uint8_t number_of_events = payload_length / 4;
// sanity
if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
@ -309,7 +309,7 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
for (int n = 0; n < number_of_events; ++n) {
bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
std::set<WebRtc_UWord8>::iterator event =
std::set<uint8_t>::iterator event =
telephone_event_reported_.find(payload_data[4 * n]);
if (event != telephone_event_reported_.end()) {
@ -340,7 +340,7 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
}
// Check if this is a CNG packet, receiver might want to know
WebRtc_UWord32 ignored;
uint32_t ignored;
bool also_ignored;
if (CNGPayloadType(rtp_header->header.payloadType,
&ignored,
@ -358,7 +358,7 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
// don't forward event to decoder
return 0;
}
std::set<WebRtc_UWord8>::iterator first =
std::set<uint8_t>::iterator first =
telephone_event_reported_.begin();
if (first != telephone_event_reported_.end() && *first > 15) {
// don't forward non DTMF events