Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
index 08d101c5ae4355fd41beae0dbee6d628ea456b04..d2984b97b0918d4081c5cf7dc5f403d532d799cc 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
@@ -13,12 +13,14 @@ |
#include <assert.h> |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/trace_event.h" |
namespace webrtc { |
int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, |
int sample_rate_hz, size_t max_decoded_bytes, |
int16_t* decoded, SpeechType* speech_type) { |
+ TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); |
int duration = PacketDuration(encoded, encoded_len); |
if (duration >= 0 && |
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
@@ -31,6 +33,7 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len, |
int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, |
int sample_rate_hz, size_t max_decoded_bytes, |
int16_t* decoded, SpeechType* speech_type) { |
+ TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); |
int duration = PacketDurationRedundant(encoded, encoded_len); |
if (duration >= 0 && |
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { |
@@ -40,12 +43,6 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len, |
speech_type); |
} |
-int AudioDecoder::DecodeInternal(const uint8_t* encoded, size_t encoded_len, |
- int sample_rate_hz, int16_t* decoded, |
- SpeechType* speech_type) { |
- return kNotImplemented; |
-} |
- |
int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, |
size_t encoded_len, |
int sample_rate_hz, int16_t* decoded, |