Index: webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc |
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc |
index dab5805f98afcc41eaa714d49e4cd8baec5ebe59..9eacfc0ff0ac24fa8d221c2759eebda2bcf65c0b 100644 |
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc |
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc |
@@ -11,7 +11,9 @@ |
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/logging.h" |
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" |
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h" |
namespace webrtc { |
@@ -49,6 +51,51 @@ void AudioDecoderIlbc::Reset() { |
WebRtcIlbcfix_Decoderinit30Ms(dec_state_); |
} |
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload( |
+ rtc::Buffer* payload, |
+ uint32_t timestamp, |
+ bool is_primary) { |
+ std::vector<ParseResult> results; |
+ size_t bytes_per_frame; |
+ int timestamps_per_frame; |
+ if (payload->size() >= 950) { |
+ LOG(LS_WARNING) << "AudioDecoderIlbc::SplitPacket: Payload too large"; |
ossu
2016/09/13 14:25:55
This comment should say ParsePayload and not Split
|
+ return results; |
+ } |
+ if (payload->size() % 38 == 0) { |
+ // 20 ms frames. |
+ bytes_per_frame = 38; |
+ timestamps_per_frame = 160; |
+ } else if (payload->size() % 50 == 0) { |
+ // 30 ms frames. |
+ bytes_per_frame = 50; |
+ timestamps_per_frame = 240; |
+ } else { |
+ LOG(LS_WARNING) << "AudioDecoderIlbc::SplitPacket: Invalid payload"; |
ossu
2016/09/13 14:25:55
This comment should say ParsePayload and not Split
|
+ return results; |
+ } |
+ |
+ RTC_DCHECK(payload->size() % bytes_per_frame == 0); |
hlundin-webrtc
2016/09/15 08:49:14
RTC_DCHECK_EQ
ossu
2016/09/15 08:58:11
I did that initially, but the RTC_DHECK_EQ macro b
hlundin-webrtc
2016/09/15 09:43:38
I hate it when that happens, but I prefer the nice
kwiberg-webrtc
2016/09/15 13:01:25
+1. It's annoying, but since a "u" suffix will sol
|
+ if (payload->size() == bytes_per_frame) { |
+ std::unique_ptr<EncodedAudioFrame> frame( |
+ new LegacyEncodedAudioFrame(this, payload, is_primary)); |
+ results.emplace_back(timestamp, is_primary, std::move(frame)); |
+ } else { |
+ for (size_t byte_offset = 0, timestamp_offset = 0; |
+ byte_offset < payload->size(); |
+ byte_offset += bytes_per_frame, |
+ timestamp_offset += timestamps_per_frame) { |
+ rtc::Buffer new_payload(payload->data() + byte_offset, bytes_per_frame); |
+ std::unique_ptr<EncodedAudioFrame> frame( |
+ new LegacyEncodedAudioFrame(this, &new_payload, is_primary)); |
+ results.emplace_back(timestamp + timestamp_offset, is_primary, |
+ std::move(frame)); |
+ } |
+ } |
+ |
+ return results; |
+} |
+ |
int AudioDecoderIlbc::SampleRateHz() const { |
return 8000; |
} |