Moved codec-specific audio packet splitting into decoders.

There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}

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 +51,53 @@ 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::ParsePayload: Payload too large";
+    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::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload), is_primary));
+    results.emplace_back(timestamp, is_primary, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (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, std::move(new_payload), is_primary));
+      results.emplace_back(timestamp + timestamp_offset, is_primary,
+                           std::move(frame));
+    }
+  }
+
+  return results;
+}
+
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h

@@ -25,6 +25,9 @@ class AudioDecoderIlbc final : public AudioDecoder {
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp,
+                                        bool is_primary) override;
   int SampleRateHz() const override;
   size_t Channels() const override;
 

webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc

@@ -11,6 +11,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
@@ -54,4 +55,85 @@ TEST(IlbcTest, BadPacket) {
                            decoded_samples.data(), &speech_type));
 }
 
+class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
+ protected:
+  virtual void SetUp() {
+    const std::pair<int, int> parameters = GetParam();
+    num_frames_ = parameters.first;
+    frame_length_ms_ = parameters.second;
+    frame_length_bytes_ = (frame_length_ms_ == 20) ? 38 : 50;
+  }
+  size_t num_frames_;
+  int frame_length_ms_;
+  size_t frame_length_bytes_;
+};
+
+TEST_P(SplitIlbcTest, NumFrames) {
+  AudioDecoderIlbc decoder;
+  const size_t frame_length_samples = frame_length_ms_ * 8;
+  const auto generate_payload = [] (size_t payload_length_bytes) {
+    rtc::Buffer payload(payload_length_bytes);
+    // Fill payload with increasing integers {0, 1, 2, ...}.
+    for (size_t i = 0; i < payload.size(); ++i) {
+      payload[i] = static_cast<uint8_t>(i);
+    }
+    return payload;
+  };
+
+  const auto results = decoder.ParsePayload(
+      generate_payload(frame_length_bytes_ * num_frames_), 0, true);
+  EXPECT_EQ(num_frames_, results.size());
+
+  size_t frame_num = 0;
+  uint8_t payload_value = 0;
+  for (const auto& result : results) {
+    EXPECT_EQ(frame_length_samples * frame_num, result.timestamp);
+    const LegacyEncodedAudioFrame* frame =
+        static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+    const rtc::Buffer& payload = frame->payload();
+    EXPECT_EQ(frame_length_bytes_, payload.size());
+    for (size_t i = 0; i < payload.size(); ++i, ++payload_value) {
+      EXPECT_EQ(payload_value, payload[i]);
+    }
+    ++frame_num;
+  }
+}
+
+// Test 1 through 5 frames of 20 and 30 ms size.
+// Also test the maximum number of frames in one packet for 20 and 30 ms.
+// The maximum is defined by the largest payload length that can be uniquely
+// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
+INSTANTIATE_TEST_CASE_P(
+    IlbcTest, SplitIlbcTest,
+    ::testing::Values(std::pair<int, int>(1, 20),  // 1 frame, 20 ms.
+                      std::pair<int, int>(2, 20),  // 2 frames, 20 ms.
+                      std::pair<int, int>(3, 20),  // And so on.
+                      std::pair<int, int>(4, 20),
+                      std::pair<int, int>(5, 20),
+                      std::pair<int, int>(24, 20),
+                      std::pair<int, int>(1, 30),
+                      std::pair<int, int>(2, 30),
+                      std::pair<int, int>(3, 30),
+                      std::pair<int, int>(4, 30),
+                      std::pair<int, int>(5, 30),
+                      std::pair<int, int>(18, 30)));
+
+// Test too large payload size.
+TEST(IlbcTest, SplitTooLargePayload) {
+  AudioDecoderIlbc decoder;
+  constexpr size_t kPayloadLengthBytes = 950;
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0, true);
+  EXPECT_TRUE(results.empty());
+}
+
+// Payload not an integer number of frames.
+TEST(IlbcTest, SplitUnevenPayload) {
+  AudioDecoderIlbc decoder;
+  constexpr size_t kPayloadLengthBytes = 39;  // Not an even number of frames.
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0, true);
+  EXPECT_TRUE(results.empty());
+}
+
 }  // namespace webrtc