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Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ )

Browse Source 
Reason for revert:
breaking downstream projects

Original issue's description:
> Opus implementation of the AudioDecoderFactoryTemplate API
>
> BUG=webrtc:7837
>
> Review-Url: https://codereview.webrtc.org/2942733003
> Cr-Commit-Position: refs/heads/master@{#18646}
> Committed: d053fe4ab3

TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7837

Review-Url: https://codereview.webrtc.org/2944763002
Cr-Commit-Position: refs/heads/master@{#18648}
This commit is contained in:
charujain
2017-06-18 02:37:17 -07:00
committed by Commit Bot
parent af6293517f
commit eb2d2d31d1
11 changed files with 33 additions and 165 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
View File

@ -162,7 +162,7 @@ NamedDecoderConstructor decoder_constructors[] = {
if (format.clockrate_hz == 48000 && format.num_channels == 2 &&
num_channels) {
if (out) {
out->reset(new AudioDecoderOpusImpl(*num_channels));
out->reset(new AudioDecoderOpus(*num_channels));
}
return true;
} else {

48
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
View File

@ -19,7 +19,7 @@ namespace webrtc {
namespace {
class OpusFrame : public AudioDecoder::EncodedAudioFrame {
public:
OpusFrame(AudioDecoderOpusImpl* decoder,
OpusFrame(AudioDecoderOpus* decoder,
rtc::Buffer&& payload,
bool is_primary_payload)
: decoder_(decoder),
@ -57,25 +57,25 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
}
private:
AudioDecoderOpusImpl* const decoder_;
AudioDecoderOpus* const decoder_;
const rtc::Buffer payload_;
const bool is_primary_payload_;
};
} // namespace
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
: channels_(num_channels) {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
WebRtcOpus_DecoderCreate(&dec_state_, channels_);
WebRtcOpus_DecoderInit(dec_state_);
}
AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
AudioDecoderOpus::~AudioDecoderOpus() {
WebRtcOpus_DecoderFree(dec_state_);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
std::vector<AudioDecoder::ParseResult> AudioDecoderOpus::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
@ -95,11 +95,11 @@ std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
return results;
}
int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int ret =
@ -110,11 +110,11 @@ int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
return ret;
}
int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
@ -131,17 +131,17 @@ int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
return ret;
}
void AudioDecoderOpusImpl::Reset() {
void AudioDecoderOpus::Reset() {
WebRtcOpus_DecoderInit(dec_state_);
}
int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
}
int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return PacketDuration(encoded, encoded_len);
@ -150,18 +150,18 @@ int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
return WebRtcOpus_FecDurationEst(encoded, encoded_len);
}
bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
int fec;
fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
return (fec == 1);
}
int AudioDecoderOpusImpl::SampleRateHz() const {
int AudioDecoderOpus::SampleRateHz() const {
return 48000;
}
size_t AudioDecoderOpusImpl::Channels() const {
size_t AudioDecoderOpus::Channels() const {
return channels_;
}

8
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
View File

@ -17,10 +17,10 @@
namespace webrtc {
class AudioDecoderOpusImpl final : public AudioDecoder {
class AudioDecoderOpus final : public AudioDecoder {
public:
explicit AudioDecoderOpusImpl(size_t num_channels);
~AudioDecoderOpusImpl() override;
explicit AudioDecoderOpus(size_t num_channels);
~AudioDecoderOpus() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp) override;
@ -47,7 +47,7 @@ class AudioDecoderOpusImpl final : public AudioDecoder {
private:
OpusDecInst* dec_state_;
const size_t channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpusImpl);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
};
} // namespace webrtc

4
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -433,7 +433,7 @@ class AudioDecoderOpusTest : public AudioDecoderTest {
codec_input_rate_hz_ = 48000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpusImpl(1);
decoder_ = new AudioDecoderOpus(1);
AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
@ -446,7 +446,7 @@ class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
delete decoder_;
decoder_ = new AudioDecoderOpusImpl(2);
decoder_ = new AudioDecoderOpus(2);
AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;