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_); |