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Unified Diff: webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

Issue 1342933005: Move AudioDecoderOpus next to AudioEncoderOpus (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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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_);
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