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Switching to stl instead of WebRTC wrappers.

Browse Source 
BUG=C-10436, C-10437, C-10438

Review URL: https://webrtc-codereview.appspot.com/343012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1468 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-01-19 12:48:53 +00:00
parent 8e50693736
commit 70111e69bf
4 changed files with 35 additions and 33 deletions
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7
src/video_engine/test/auto_test/interface/tb_external_transport.h
View File

@ -15,8 +15,9 @@
#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
#include <list>
#include "common_types.h"
#include "list_wrapper.h"
namespace webrtc
{
@ -85,8 +86,8 @@ private:
WebRtc_Word32 _rtcpCount;
WebRtc_Word32 _dropCount;
webrtc::ListWrapper _rtpPackets;
webrtc::ListWrapper _rtcpPackets;
std::list<VideoPacket*> _rtpPackets;
std::list<VideoPacket*> _rtcpPackets;
unsigned char _temporalLayers;
unsigned short _seqNum;

18
src/video_engine/test/auto_test/source/tb_external_transport.cc
View File

@ -48,8 +48,6 @@ TbExternalTransport::TbExternalTransport(webrtc::ViENetwork& vieNetwork) :
_rtpCount(0),
_rtcpCount(0),
_dropCount(0),
_rtpPackets(),
_rtcpPackets(),
_temporalLayers(0),
_seqNum(0),
_sendPID(0),
@ -196,7 +194,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
_crit.Enter();
newPacket->receiveTime = NowMs() + _networkDelayMs;
_rtpPackets.PushBack(newPacket);
_rtpPackets.push_back(newPacket);
_event.Set();
_crit.Leave();
return len;
@ -221,7 +219,7 @@ int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
_crit.Enter();
newPacket->receiveTime = NowMs() + _networkDelayMs;
_rtcpPackets.PushBack(newPacket);
_rtcpPackets.push_back(newPacket);
_event.Set();
_crit.Leave();
return len;
@ -300,11 +298,11 @@ bool TbExternalTransport::ViEExternalTransportProcess()
VideoPacket* packet = NULL;
while (!_rtpPackets.Empty())
while (!_rtpPackets.empty())
{
// Take first packet in queue
_crit.Enter();
packet = static_cast<VideoPacket*> ((_rtpPackets.First())->GetItem());
packet = _rtpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
if (timeToReceive > 0)
{
@ -316,7 +314,7 @@ bool TbExternalTransport::ViEExternalTransportProcess()
_crit.Leave();
break;
}
_rtpPackets.PopFront();
_rtpPackets.pop_front();
_crit.Leave();
// Send to ViE
@ -347,11 +345,11 @@ bool TbExternalTransport::ViEExternalTransportProcess()
packet = NULL;
}
}
while (!_rtcpPackets.Empty())
while (!_rtcpPackets.empty())
{
// Take first packet in queue
_crit.Enter();
packet = static_cast<VideoPacket*> ((_rtcpPackets.First())->GetItem());
packet = _rtcpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
if (timeToReceive > 0)
{
@ -363,7 +361,7 @@ bool TbExternalTransport::ViEExternalTransportProcess()
_crit.Leave();
break;
}
_rtcpPackets.PopFront();
_rtcpPackets.pop_front();
_crit.Leave();
// Send to ViE

35
src/video_engine/vie_file_player.cc
View File

@ -270,8 +270,12 @@ int ViEFilePlayer::StopPlay() {
int ViEFilePlayer::StopPlayAudio() {
// Stop sending audio.
while (MapItem* audio_item = audio_channels_sending_.First()) {
StopSendAudioOnChannel(audio_item->GetId());
std::set<int>::iterator it = audio_channels_sending_.begin();
while (it != audio_channels_sending_.end()) {
StopSendAudioOnChannel(*it);
// StopSendAudioOnChannel erases the item from the map.
it = audio_channels_sending_.begin();
}
// Stop local audio playback.
@ -279,10 +283,7 @@ int ViEFilePlayer::StopPlayAudio() {
StopPlayAudioLocally(local_audio_channel_);
}
local_audio_channel_ = -1;
while (audio_channel_buffers_.PopFront() != -1) {
}
while (audio_channels_sending_.Erase(audio_channels_sending_.First()) != -1) {
}
audio_channel_buffers_.clear();
audio_clients_ = 0;
return 0;
}
@ -300,8 +301,8 @@ int ViEFilePlayer::Read(void* buf, int len) {
}
// 2 bytes per sample.
decoded_audio_length_ *= 2;
if (buf != 0) {
audio_channel_buffers_.PushBack(buf);
if (buf) {
audio_channel_buffers_.push_back(buf);
}
} else {
// No need for new audiobuffer from file, ie the buffer read from file has
@ -315,16 +316,16 @@ int ViEFilePlayer::Read(void* buf, int len) {
bool ViEFilePlayer::NeedsAudioFromFile(void* buf) {
bool needs_new_audio = false;
if (audio_channel_buffers_.GetSize() == 0) {
if (audio_channel_buffers_.size() == 0) {
return true;
}
// Check if we the buf already have read the current audio.
for (ListItem* item = audio_channel_buffers_.First(); item != NULL;
item = audio_channel_buffers_.Next(item)) {
if (item->GetItem() == buf) {
for (std::list<void*>::iterator it = audio_channel_buffers_.begin();
it != audio_channel_buffers_.end(); ++it) {
if (*it == buf) {
needs_new_audio = true;
audio_channel_buffers_.Erase(item);
audio_channel_buffers_.erase(it);
break;
}
}
@ -381,7 +382,7 @@ int ViEFilePlayer::SendAudioOnChannel(const int audio_channel,
audio_channel, mix_microphone, volume_scaling);
return -1;
}
audio_channels_sending_.Insert(audio_channel, NULL);
audio_channels_sending_.insert(audio_channel);
CriticalSectionScoped lock(*audio_cs_);
audio_clients_++;
@ -390,8 +391,8 @@ int ViEFilePlayer::SendAudioOnChannel(const int audio_channel,
int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) {
int result = 0;
MapItem* audio_item = audio_channels_sending_.Find(audio_channel);
if (!audio_item) {
std::set<int>::iterator it = audio_channels_sending_.find(audio_channel);
if (it == audio_channels_sending_.end()) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
"ViEFilePlayer::StopSendAudioOnChannel AudioChannel %d not "
"sending", audio_channel);
@ -404,7 +405,7 @@ int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) {
"VE_StopPlayingFileAsMicrophone failed. audio_channel %d",
audio_channel);
}
audio_channels_sending_.Erase(audio_item);
audio_channels_sending_.erase(audio_channel);
CriticalSectionScoped lock(*audio_cs_);
audio_clients_--;
assert(audio_clients_ >= 0);

8
src/video_engine/vie_file_player.h
View File

@ -11,10 +11,12 @@
#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
#define WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
#include <list>
#include <set>
#include "common_types.h"
#include "modules/media_file/interface/media_file_defines.h"
#include "system_wrappers/interface/file_wrapper.h"
#include "system_wrappers/interface/list_wrapper.h"
#include "typedefs.h"
#include "video_engine/vie_frame_provider_base.h"
@ -124,10 +126,10 @@ class ViEFilePlayer
// Trick - list containing VoE buffer reading this file. Used if multiple
// audio channels are sending.
ListWrapper audio_channel_buffers_;
std::list<void*> audio_channel_buffers_;
// AudioChannels sending audio from this file.
MapWrapper audio_channels_sending_;
std::set<int> audio_channels_sending_;
// Frame receiving decoded video from file.
VideoFrame decoded_video_;