webrtc/modules/audio_coding/codecs/audio_encoder.h - Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView

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

Issue 1418423010: Pass audio to AudioEncoder::Encode() in an ArrayView (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 5 years, 1 month ago

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Index: webrtc/modules/audio_coding/codecs/audio_encoder.h

diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h

index cda9d86f2e6ce40589f9f04b3116c89c30fa88a7..553c35e2690f76b29431f86063bd2dc39e1ca07f 100644

--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h

+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h

@@ -14,6 +14,7 @@

#include <algorithm>

#include <vector>

+#include "webrtc/base/array_view.h"

#include "webrtc/typedefs.h"

namespace webrtc {

@@ -91,13 +92,12 @@ class AudioEncoder {

// Encode() checks some preconditions, calls EncodeInternal() which does the

// actual work, and then checks some postconditions.

EncodedInfo Encode(uint32_t rtp_timestamp,

- const int16_t* audio,

- size_t num_samples_per_channel,

+ rtc::ArrayView<const int16_t> audio,

size_t max_encoded_bytes,

uint8_t* encoded);

virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,

- const int16_t* audio,

+ rtc::ArrayView<const int16_t> audio,

size_t max_encoded_bytes,

uint8_t* encoded) = 0;