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 442ddc1e4b86b081653da168b363bf26e6bc9683..434ac287f1326f822e68c481ab21e8d4ebb3622a 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc |
@@ -65,6 +65,11 @@ int AudioDecoder::IncomingPacket(const uint8_t* payload, |
return 0; |
} |
+AudioDecoder::PacketSplits AudioDecoder::SplitPacket(const uint8_t* payload, |
+ size_t payload_len) const { |
+ return {PacketSplit{0, payload_len, 0}}; |
+} |
+ |
int AudioDecoder::ErrorCode() { return 0; } |
int AudioDecoder::PacketDuration(const uint8_t* encoded, |
@@ -95,4 +100,42 @@ AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) { |
} |
} |
+AudioDecoder::PacketSplits AudioDecoder::SplitBySamples( |
+ const uint8_t* payload, |
+ size_t payload_len, |
+ size_t bytes_per_ms, |
+ uint32_t timestamps_per_ms) { |
kwiberg-webrtc
2016/08/26 12:39:24
This is the sort of function that should have a un
ossu
2016/08/26 13:05:29
It does. Several of them even.
Though, granted, it
kwiberg-webrtc
2016/08/26 22:05:13
It's arguably not a unit test of this function in
|
+ PacketSplits splits; |
+ RTC_DCHECK(payload); |
+ |
+ size_t split_size_bytes = payload_len; |
+ |
+ // Find a "chunk size" >= 20 ms and < 40 ms. |
+ size_t min_chunk_size = bytes_per_ms * 20; |
kwiberg-webrtc
2016/08/26 12:39:24
const
ossu
2016/08/26 13:05:29
Hey, don't blame me, I just clean up around here!
kwiberg-webrtc
2016/08/26 22:05:13
Yeah, I think I wrote this comment before I realiz
|
+ // Reduce the split size by half as long as |split_size_bytes| is at least |
+ // twice the minimum chunk size (so that the resulting size is at least as |
+ // large as the minimum chunk size). |
+ while (split_size_bytes >= 2 * min_chunk_size) { |
+ split_size_bytes >>= 1; |
kwiberg-webrtc
2016/08/26 12:39:24
Less cleverly written as split_size_bytes /= 2;
I
ossu
2016/08/26 13:05:29
Yeah, I mean the code can be improved but I've aim
kwiberg-webrtc
2016/08/26 22:05:13
sgtm.
|
+ } |
+ uint32_t timestamps_per_chunk = static_cast<uint32_t>( |
kwiberg-webrtc
2016/08/26 12:39:24
const
|
+ split_size_bytes * timestamps_per_ms / bytes_per_ms); |
+ PacketSplit split = {0, split_size_bytes, 0}; |
+ size_t len = payload_len; |
+ for (len = payload_len; |
+ len >= (2 * split_size_bytes); |
+ len -= split_size_bytes) { |
+ splits.push_back(split); |
+ split.byte_offset += split_size_bytes; |
+ split.timestamp_offset += timestamps_per_chunk; |
+ } |
+ |
+ if (len > 0) { |
+ split.num_bytes = len; |
+ splits.push_back(split); |
+ } |
+ |
+ return splits; |
+} |
+ |
} // namespace webrtc |