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Re-land "Switch to using AudioEncoderIsac instead of ACMISAC"

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It should work now, after the fix in r8431.

Previously committed in r8342, reverted in r8372, committed in r8378,
and reverted in r8412.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34279004

Cr-Commit-Position: refs/heads/master@{#8433}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8433 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org 2015-02-19 15:10:20 +00:00
parent 1ed6224eaf
commit be96bfb179
5 changed files with 26 additions and 27 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

14
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View File

@ -23,6 +23,7 @@ namespace webrtc {
const int kIsacPayloadType = 103;
const int kInvalidPayloadType = -1;
const int kDefaultBitRate = 32000;
template <typename T>
AudioEncoderDecoderIsacT<T>::Config::Config()
@ -30,7 +31,7 @@ AudioEncoderDecoderIsacT<T>::Config::Config()
red_payload_type(kInvalidPayloadType),
sample_rate_hz(16000),
frame_size_ms(30),
bit_rate(32000),
bit_rate(kDefaultBitRate),
max_bit_rate(-1),
max_payload_size_bytes(-1) {
}
@ -48,7 +49,7 @@ bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
if (max_payload_size_bytes > 400)
return false;
return (frame_size_ms == 30 || frame_size_ms == 60) &&
bit_rate >= 10000 && bit_rate <= 32000;
((bit_rate >= 10000 && bit_rate <= 32000) || bit_rate == 0);
case 32000:
case 48000:
if (max_bit_rate > 160000)
@ -56,7 +57,8 @@ bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
if (max_payload_size_bytes > 600)
return false;
return T::has_swb &&
(frame_size_ms == 30 && bit_rate >= 10000 && bit_rate <= 56000);
(frame_size_ms == 30 &&
((bit_rate >= 10000 && bit_rate <= 56000) || bit_rate == 0));
default:
return false;
}
@ -68,7 +70,7 @@ AudioEncoderDecoderIsacT<T>::ConfigAdaptive::ConfigAdaptive()
red_payload_type(kInvalidPayloadType),
sample_rate_hz(16000),
initial_frame_size_ms(30),
initial_bit_rate(32000),
initial_bit_rate(kDefaultBitRate),
max_bit_rate(-1),
enforce_frame_size(false),
max_payload_size_bytes(-1) {
@ -114,7 +116,9 @@ AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
CHECK_EQ(0, T::Create(&isac_state_));
CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
CHECK_EQ(0, T::Control(isac_state_, config.bit_rate, config.frame_size_ms));
CHECK_EQ(0, T::Control(isac_state_, config.bit_rate == 0 ? kDefaultBitRate
: config.bit_rate,
config.frame_size_ms));
// When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is
// still set to 32000 Hz, since there is no full-band mode in the decoder.
CHECK_EQ(0, T::SetDecSampRate(isac_state_,

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

@ -575,7 +575,9 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst& codec_inst,
// All we have support for right now.
if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
return new ACMISAC(kISAC, enable_red);
return new ACMGenericCodecWrapper(codec_inst, cng_pt_nb, cng_pt_wb,
cng_pt_swb, cng_pt_fb, enable_red,
red_payload_type);
#endif
} else if (!STR_CASE_CMP(codec_inst.plname, "PCMU") ||
!STR_CASE_CMP(codec_inst.plname, "PCMA")) {

13
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -270,7 +270,6 @@ int32_t AudioCodingModuleImpl::Process() {
FrameType frame_type = kAudioFrameSpeech;
uint8_t current_payload_type = 0;
bool has_data_to_send = false;
bool red_active = false;
RTPFragmentationHeader my_fragmentation;
// Keep the scope of the ACM critical section limited.
@ -302,36 +301,32 @@ int32_t AudioCodingModuleImpl::Process() {
}
case kActiveNormalEncoded:
case kPassiveNormalEncoded: {
current_payload_type = static_cast<uint8_t>(send_codec_inst_.pltype);
frame_type = kAudioFrameSpeech;
break;
}
case kPassiveDTXNB: {
current_payload_type = cng_nb_pltype_;
frame_type = kAudioFrameCN;
is_first_red_ = true;
break;
}
case kPassiveDTXWB: {
current_payload_type = cng_wb_pltype_;
frame_type = kAudioFrameCN;
is_first_red_ = true;
break;
}
case kPassiveDTXSWB: {
current_payload_type = cng_swb_pltype_;
frame_type = kAudioFrameCN;
is_first_red_ = true;
break;
}
case kPassiveDTXFB: {
current_payload_type = cng_fb_pltype_;
frame_type = kAudioFrameCN;
is_first_red_ = true;
break;
}
}
has_data_to_send = true;
current_payload_type = encoded_info.payload_type;
previous_pltype_ = current_payload_type;
ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
@ -348,8 +343,9 @@ int32_t AudioCodingModuleImpl::Process() {
// have been switched to the new AudioEncoder interface.
if ((codecs_[current_send_codec_idx_]->ExternalRedNeeded()) &&
((encoding_type == kActiveNormalEncoded) ||
(encoding_type == kPassiveNormalEncoded))) {
(encoding_type == kPassiveNormalEncoded))) {
DCHECK(encoded_info.redundant.empty());
FATAL() << "Don't go here!";
// RED is enabled within this scope.
//
// Note that, a special solution exists for iSAC since it is the only
@ -389,7 +385,6 @@ int32_t AudioCodingModuleImpl::Process() {
//
// Hence, even if every second packet is dropped, perfect
// reconstruction is possible.
red_active = true;
has_data_to_send = false;
// Skip the following part for the first packet in a RED session.
@ -457,7 +452,7 @@ int32_t AudioCodingModuleImpl::Process() {
CriticalSectionScoped lock(callback_crit_sect_);
if (packetization_callback_ != NULL) {
if (red_active || my_fragmentation.fragmentationVectorSize > 0) {
if (my_fragmentation.fragmentationVectorSize > 0) {
// Callback with payload data, including redundant data (RED).
packetization_callback_->SendData(frame_type, current_payload_type,
rtp_timestamp, stream, length_bytes,

11
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -809,15 +809,14 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) {
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
"98d960600eb4ddb3fcbe11f5057ddfd7",
"2b3c387d06f00b7b7aad4c9be56fb83d",
"",
"2f6dfe142f735f1d96f6bd86d2526f42"),
"5683b58da0fbf2063c7adc2e6bfb3fb8"),
AcmReceiverBitExactness::PlatformChecksum(
"cc9d2d86a71d6f99f97680a5c27e2762",
"bcc2041e7744c7ebd9f701866856849c",
"",
"7b214fc3a5e33d68bf30e77969371f31"),
33,
test::AcmReceiveTest::kMonoOutput);
"ce86106a93419aefb063097108ec94ab"),
33, test::AcmReceiveTest::kMonoOutput);
}
TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) {

11
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -960,15 +960,14 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb60ms) {
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"98d960600eb4ddb3fcbe11f5057ddfd7",
"2b3c387d06f00b7b7aad4c9be56fb83d",
"",
"2f6dfe142f735f1d96f6bd86d2526f42"),
"5683b58da0fbf2063c7adc2e6bfb3fb8"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"cc9d2d86a71d6f99f97680a5c27e2762",
"bcc2041e7744c7ebd9f701866856849c",
"",
"7b214fc3a5e33d68bf30e77969371f31"),
33,
test::AcmReceiveTestOldApi::kMonoOutput);
"ce86106a93419aefb063097108ec94ab"),
33, test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {