Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| index 592f17b097d5b343b0edddc72f142c18e45dd4e6..2e05fe1c5191334a72c0baf35040cbfb06a0b95b 100644 |
| --- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc |
| @@ -29,7 +29,7 @@ |
| #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h" |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| -#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" |
| +#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h" |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" |
| @@ -299,86 +299,6 @@ void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded, |
| } |
| #endif |
| -// Opus |
| -#ifdef WEBRTC_CODEC_OPUS |
|
hlundin-webrtc
2015/09/15 10:30:45
Did you just lose this conditional compilation of
kwiberg-webrtc
2015/09/15 10:45:07
The entire file I moved it to is conditioned on in
hlundin-webrtc
2015/09/15 10:49:13
Acknowledged.
|
| -AudioDecoderOpus::AudioDecoderOpus(size_t num_channels) |
| - : channels_(num_channels) { |
| - DCHECK(num_channels == 1 || num_channels == 2); |
| - WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_)); |
| - WebRtcOpus_DecoderInit(dec_state_); |
| -} |
| - |
| -AudioDecoderOpus::~AudioDecoderOpus() { |
| - WebRtcOpus_DecoderFree(dec_state_); |
| -} |
| - |
| -int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded, |
| - size_t encoded_len, |
| - int sample_rate_hz, |
| - int16_t* decoded, |
| - SpeechType* speech_type) { |
| - DCHECK_EQ(sample_rate_hz, 48000); |
| - int16_t temp_type = 1; // Default is speech. |
| - int ret = WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, |
| - &temp_type); |
| - if (ret > 0) |
| - ret *= static_cast<int>(channels_); // Return total number of samples. |
| - *speech_type = ConvertSpeechType(temp_type); |
| - return ret; |
| -} |
| - |
| -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, |
| - speech_type); |
| - } |
| - |
| - DCHECK_EQ(sample_rate_hz, 48000); |
| - int16_t temp_type = 1; // Default is speech. |
| - int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded, |
| - &temp_type); |
| - if (ret > 0) |
| - ret *= static_cast<int>(channels_); // Return total number of samples. |
| - *speech_type = ConvertSpeechType(temp_type); |
| - return ret; |
| -} |
| - |
| -void AudioDecoderOpus::Reset() { |
| - WebRtcOpus_DecoderInit(dec_state_); |
| -} |
| - |
| -int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, |
| - size_t encoded_len) const { |
| - return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len); |
| -} |
| - |
| -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); |
| - } |
| - |
| - return WebRtcOpus_FecDurationEst(encoded, encoded_len); |
| -} |
| - |
| -bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, |
| - size_t encoded_len) const { |
| - int fec; |
| - fec = WebRtcOpus_PacketHasFec(encoded, encoded_len); |
| - return (fec == 1); |
| -} |
| - |
| -size_t AudioDecoderOpus::Channels() const { |
| - return channels_; |
| -} |
| -#endif |
| - |
| AudioDecoderCng::AudioDecoderCng() { |
| CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_)); |
| WebRtcCng_InitDec(dec_state_); |