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Unified Diff: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h

Issue 2326003002: Moved codec-specific audio packet splitting into decoders. (Closed)
Patch Set: Fixed types in packet splitting (size_t vs. uint32_t) Created 4 years, 3 months ago
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Index: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
diff --git a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
new file mode 100644
index 0000000000000000000000000000000000000000..1c9c3f542a8049f27e864bf8730a0bfaa066f165
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2016 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_LEGACY_ENCODED_AUDIO_FRAME_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+namespace webrtc {
+
+class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+ LegacyEncodedAudioFrame(AudioDecoder* decoder,
+ rtc::Buffer&& payload,
+ bool is_primary_payload);
+ ~LegacyEncodedAudioFrame() override;
+
+ static std::vector<AudioDecoder::ParseResult> SplitBySamples(
+ AudioDecoder* decoder,
+ rtc::Buffer&& payload,
+ uint32_t timestamp,
+ bool is_primary,
+ size_t bytes_per_ms,
+ uint32_t timestamps_per_ms);
+
+ size_t Duration() const override;
+
+ rtc::Optional<DecodeResult> Decode(
+ rtc::ArrayView<int16_t> decoded) const override;
+
+ // For testing:
+ const rtc::Buffer& payload() const { return payload_; }
+
+ private:
+ AudioDecoder* const decoder_;
+ const rtc::Buffer payload_;
+ const bool is_primary_payload_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_

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