Move AudioDecoderOpus next to AudioEncoderOpus
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1342933005 .
Cr-Commit-Position: refs/heads/master@{#9944}
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000..e78fc04
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+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_;
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
new file mode 100644
index 0000000..9fa77b0
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+namespace webrtc {
+
+class AudioDecoderOpus : public AudioDecoder {
+ public:
+ explicit AudioDecoderOpus(size_t num_channels);
+ ~AudioDecoderOpus() override;
+
+ void Reset() override;
+ int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+ int PacketDurationRedundant(const uint8_t* encoded,
+ size_t encoded_len) const override;
+ bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+ size_t Channels() const override;
+
+ protected:
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+ int DecodeRedundantInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ OpusDecInst* dec_state_;
+ const size_t channels_;
+ DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_DECODER_OPUS_H
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index 4ae4340..5a420b4 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
@@ -43,7 +43,9 @@
'<(webrtc_root)',
],
'sources': [
+ 'audio_decoder_opus.cc',
'audio_encoder_opus.cc',
+ 'interface/audio_decoder_opus.h',
'interface/audio_encoder_opus.h',
'interface/opus_interface.h',
'opus_inst.h',