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Let AudioCodingModule::SendCodec return Maybe<CodecInst>

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And deal with the consequences thereof...

Review URL: https://codereview.webrtc.org/1406123011

Cr-Commit-Position: refs/heads/master@{#10497}
This commit is contained in:
kwiberg
2015-11-03 11:20:50 -08:00
committed by Commit bot
parent 969aeb1910
commit 1fd4a4ab35
15 changed files with 72 additions and 78 deletions
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6
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
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@ -93,9 +93,9 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
if (timestamp_ == 0) { // This is the first time inserting audio.
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
} else {
CodecInst current_codec;
ASSERT_EQ(0, acm_->SendCodec(&current_codec));
if (!CodecsEqual(codec, current_codec))
auto current_codec = acm_->SendCodec();
ASSERT_TRUE(current_codec);
if (!CodecsEqual(codec, *current_codec))
ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
}
AudioFrame frame;

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

@ -206,9 +206,9 @@ void AudioCodingModuleImpl::RegisterExternalSendCodec(
}
// Get current send codec.
int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const {
rtc::Maybe<CodecInst> AudioCodingModuleImpl::SendCodec() const {
CriticalSectionScoped lock(acm_crit_sect_.get());
return codec_manager_.GetCodecInst(current_codec);
return codec_manager_.GetCodecInst();
}
// Get current send frequency.

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

@ -48,7 +48,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
AudioEncoder* external_speech_encoder) override;
// Get current send codec.
int SendCodec(CodecInst* current_codec) const override;
rtc::Maybe<CodecInst> SendCodec() const override;
// Get current send frequency.
int SendFrequency() const override;

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

@ -343,7 +343,7 @@ void CodecManager::RegisterEncoder(AudioEncoder* external_speech_encoder) {
codec_owner_.SetEncoders(external_speech_encoder, cng_pt, vad_mode_, red_pt);
}
int CodecManager::GetCodecInst(CodecInst* current_codec) const {
rtc::Maybe<CodecInst> CodecManager::GetCodecInst() const {
int dummy_id = 0;
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
"SendCodec()");
@ -351,10 +351,9 @@ int CodecManager::GetCodecInst(CodecInst* current_codec) const {
if (!codec_owner_.Encoder()) {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
"SendCodec Failed, no codec is registered");
return -1;
return rtc::Maybe<CodecInst>();
}
*current_codec = send_codec_inst_;
return 0;
return rtc::Maybe<CodecInst>(send_codec_inst_);
}
bool CodecManager::SetCopyRed(bool enable) {

3
webrtc/modules/audio_coding/main/acm2/codec_manager.h
View File

@ -12,6 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/maybe.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_coding/main/acm2/codec_owner.h"
@ -34,7 +35,7 @@ class CodecManager final {
void RegisterEncoder(AudioEncoder* external_speech_encoder);
int GetCodecInst(CodecInst* current_codec) const;
rtc::Maybe<CodecInst> GetCodecInst() const;
bool SetCopyRed(bool enable);

9
webrtc/modules/audio_coding/main/include/audio_coding_module.h
View File

@ -13,6 +13,7 @@
#include <vector>
#include "webrtc/base/maybe.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
@ -210,14 +211,10 @@ class AudioCodingModule {
// int32_t SendCodec()
// Get parameters for the codec currently registered as send codec.
//
// Output:
// -current_send_codec : parameters of the send codec.
//
// Return value:
// -1 if failed to get send codec,
// 0 if succeeded.
// The send codec, or nothing if we don't have one
//
virtual int32_t SendCodec(CodecInst* current_send_codec) const = 0;
virtual rtc::Maybe<CodecInst> SendCodec() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SendFrequency()

53
webrtc/modules/audio_coding/main/test/APITest.cc
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@ -823,9 +823,11 @@ void APITest::TestRegisteration(char sendSide) {
exit(-1);
}
CodecInst myCodec;
if (sendACM->SendCodec(&myCodec) < 0) {
AudioCodingModule::Codec(_codecCntrA, &myCodec);
auto myCodec = sendACM->SendCodec();
if (!myCodec) {
CodecInst ci;
AudioCodingModule::Codec(_codecCntrA, &ci);
myCodec = rtc::Maybe<CodecInst>(ci);
}
if (!_randomTest) {
@ -837,12 +839,12 @@ void APITest::TestRegisteration(char sendSide) {
*thereIsDecoder = false;
}
//myEvent->Wait(20);
CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
Wait(1000);
int currentPayload = myCodec.pltype;
int currentPayload = myCodec->pltype;
if (!FixedPayloadTypeCodec(myCodec.plname)) {
if (!FixedPayloadTypeCodec(myCodec->plname)) {
int32_t i;
for (i = 0; i < 32; i++) {
if (!_payloadUsed[i]) {
@ -850,9 +852,9 @@ void APITest::TestRegisteration(char sendSide) {
fprintf(stdout,
"Register receive codec with new Payload, AUDIO BACK.\n");
}
//myCodec.pltype = i + 96;
//CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
//CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
//myCodec->pltype = i + 96;
//CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
//CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
//myEvent->Wait(20);
//{
// WriteLockScoped wl(_apiTestRWLock);
@ -868,17 +870,17 @@ void APITest::TestRegisteration(char sendSide) {
// *thereIsDecoder = false;
//}
//myEvent->Wait(20);
//CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
//CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
Wait(1000);
myCodec.pltype = currentPayload;
myCodec->pltype = currentPayload;
if (!_randomTest) {
fprintf(stdout,
"Register receive codec with default Payload, AUDIO BACK.\n");
fflush (stdout);
}
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
//CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
//CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
myEvent->Wait(20);
{
WriteLockScoped wl(_apiTestRWLock);
@ -890,7 +892,7 @@ void APITest::TestRegisteration(char sendSide) {
}
}
if (i == 32) {
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
{
WriteLockScoped wl(_apiTestRWLock);
*thereIsDecoder = true;
@ -902,9 +904,9 @@ void APITest::TestRegisteration(char sendSide) {
"Register receive codec with fixed Payload, AUDIO BACK.\n");
fflush (stdout);
}
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
//CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
//CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
//CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
//CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
myEvent->Wait(20);
{
WriteLockScoped wl(_apiTestRWLock);
@ -1001,22 +1003,17 @@ void APITest::TestSendVAD(char side) {
}
void APITest::CurrentCodec(char side) {
CodecInst myCodec;
if (side == 'A') {
_acmA->SendCodec(&myCodec);
} else {
_acmB->SendCodec(&myCodec);
}
auto myCodec = (side == 'A' ? _acmA : _acmB)->SendCodec();
if (!_randomTest) {
fprintf(stdout, "\n\n");
fprintf(stdout, "Send codec in Side A\n");
fprintf(stdout, "----------------------------\n");
fprintf(stdout, "Name................. %s\n", myCodec.plname);
fprintf(stdout, "Sampling Frequency... %d\n", myCodec.plfreq);
fprintf(stdout, "Rate................. %d\n", myCodec.rate);
fprintf(stdout, "Payload-type......... %d\n", myCodec.pltype);
fprintf(stdout, "Packet-size.......... %d\n", myCodec.pacsize);
fprintf(stdout, "Name................. %s\n", myCodec->plname);
fprintf(stdout, "Sampling Frequency... %d\n", myCodec->plfreq);
fprintf(stdout, "Rate................. %d\n", myCodec->rate);
fprintf(stdout, "Payload-type......... %d\n", myCodec->pltype);
fprintf(stdout, "Packet-size.......... %d\n", myCodec->pacsize);
}
Wait(100);

3
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -339,8 +339,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
_sender.codeId = codeId;
_sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
struct CodecInst sendCodecInst;
if (acm->SendCodec(&sendCodecInst) >= 0) {
if (acm->SendCodec()) {
_sender.Run();
}
_sender.Teardown();

3
webrtc/modules/audio_coding/main/test/PacketLossTest.cc
View File

@ -143,8 +143,7 @@ void PacketLossTest::Perform() {
sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
expected_loss_rate_);
struct CodecInst sendCodecInst;
if (acm->SendCodec(&sendCodecInst) >= 0) {
if (acm->SendCodec()) {
sender_->Run();
}
sender_->Teardown();

3
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
View File

@ -477,8 +477,7 @@ void TestAllCodecs::OpenOutFile(int test_number) {
void TestAllCodecs::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(&my_codec_param);
printf("%s -> ", my_codec_param.plname);
printf("%s -> ", acm_a_->SendCodec()->plname);
acm_b_->ReceiveCodec(&my_codec_param);
printf("%s\n", my_codec_param.plname);
}

11
webrtc/modules/audio_coding/main/test/TestStereo.cc
View File

@ -823,14 +823,15 @@ void TestStereo::OpenOutFile(int16_t test_number) {
}
void TestStereo::DisplaySendReceiveCodec() {
CodecInst my_codec_param;
acm_a_->SendCodec(&my_codec_param);
auto send_codec = acm_a_->SendCodec();
if (test_mode_ != 0) {
printf("%s -> ", my_codec_param.plname);
ASSERT_TRUE(send_codec);
printf("%s -> ", send_codec->plname);
}
acm_b_->ReceiveCodec(&my_codec_param);
CodecInst receive_codec;
acm_b_->ReceiveCodec(&receive_codec);
if (test_mode_ != 0) {
printf("%s\n", my_codec_param.plname);
printf("%s\n", receive_codec.plname);
}
}

6
webrtc/modules/audio_coding/main/test/TestVADDTX.cc
View File

@ -209,9 +209,9 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
EXPECT_EQ(0, acm_send_->VAD(&dtx_enabled_, &vad_enabled_, &mode));
CodecInst codec_param;
acm_send_->SendCodec(&codec_param);
if (STR_CASE_CMP(codec_param.plname, "opus") == 0) {
auto codec_param = acm_send_->SendCodec();
ASSERT_TRUE(codec_param);
if (STR_CASE_CMP(codec_param->plname, "opus") == 0) {
// If send codec is Opus, WebRTC VAD/DTX cannot be used.
enable_dtx = enable_vad = false;
}

12
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
View File

@ -250,10 +250,8 @@ void TwoWayCommunication::Perform() {
AudioFrame audioFrame;
CodecInst codecInst_B;
CodecInst dummy;
EXPECT_EQ(0, _acmB->SendCodec(&codecInst_B));
auto codecInst_B = _acmB->SendCodec();
ASSERT_TRUE(codecInst_B);
// In the following loop we tests that the code can handle misuse of the APIs.
// In the middle of a session with data flowing between two sides, called A
@ -285,15 +283,15 @@ void TwoWayCommunication::Perform() {
}
// Re-register send codec on side B.
if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
EXPECT_EQ(0, _acmB->SendCodec(&dummy));
EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B));
EXPECT_TRUE(_acmB->SendCodec());
}
// Initialize receiver on side A.
if (((secPassed % 7) == 6) && (msecPassed == 0))
EXPECT_EQ(0, _acmA->InitializeReceiver());
// Re-register codec on side A.
if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
}
}
}

21
webrtc/modules/audio_coding/main/test/iSACTest.cc
View File

@ -47,21 +47,21 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
if ((isacConfig.currentRateBitPerSec != 0)
|| (isacConfig.currentFrameSizeMsec != 0)) {
CodecInst sendCodec;
EXPECT_EQ(0, acm->SendCodec(&sendCodec));
auto sendCodec = acm->SendCodec();
EXPECT_TRUE(sendCodec);
if (isacConfig.currentRateBitPerSec < 0) {
// Register iSAC in adaptive (channel-dependent) mode.
sendCodec.rate = -1;
EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
sendCodec->rate = -1;
EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
} else {
if (isacConfig.currentRateBitPerSec != 0) {
sendCodec.rate = isacConfig.currentRateBitPerSec;
sendCodec->rate = isacConfig.currentRateBitPerSec;
}
if (isacConfig.currentFrameSizeMsec != 0) {
sendCodec.pacsize = isacConfig.currentFrameSizeMsec
* (sendCodec.plfreq / 1000);
sendCodec->pacsize = isacConfig.currentFrameSizeMsec
* (sendCodec->plfreq / 1000);
}
EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
}
}
@ -238,7 +238,6 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
_channel_B2A->ResetStats();
char currentTime[500];
CodecInst sendCodec;
EventTimerWrapper* myEvent = EventTimerWrapper::Create();
EXPECT_TRUE(myEvent->StartTimer(true, 10));
while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
@ -248,8 +247,8 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
if ((adaptiveMode) && (_testMode != 0)) {
myEvent->Wait(5000);
EXPECT_EQ(0, _acmA->SendCodec(&sendCodec));
EXPECT_EQ(0, _acmB->SendCodec(&sendCodec));
EXPECT_TRUE(_acmA->SendCodec());
EXPECT_TRUE(_acmB->SendCodec());
}
}

7
webrtc/voice_engine/channel.cc
View File

@ -1242,7 +1242,12 @@ Channel::DeRegisterVoiceEngineObserver()
int32_t
Channel::GetSendCodec(CodecInst& codec)
{
return (audio_coding_->SendCodec(&codec));
auto send_codec = audio_coding_->SendCodec();
if (send_codec) {
codec = *send_codec;
return 0;
}
return -1;
}
int32_t