webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h - Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

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Unified Diff: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Removed use of <functional> and moved MockAudioEncoderHelper things into the respective MockAudioEn… Created 4 years, 10 months ago

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« webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc ('K') | « webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc ('k') | webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc » ('j') | no next file with comments »
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Index: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

index 8bae160476149c2227f1422c90304c03459a0fd5..43c95b80625e5d7949c04998e116884f338ab316 100644

--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

@@ -77,7 +77,7 @@ class NetEqQualityTest : public ::testing::Test {

// 2. save the bit stream to |payload| of |max_bytes| bytes in size,

// 3. returns the length of the payload (in bytes),

virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,

- uint8_t* payload, size_t max_bytes) = 0;

+ rtc::Buffer* payload, size_t max_bytes) = 0;

// PacketLost(...) determines weather a packet sent at an indicated time gets

// lost or not.

@@ -128,7 +128,7 @@ class NetEqQualityTest : public ::testing::Test {

std::unique_ptr<LossModel> loss_model_;

std::unique_ptr<int16_t[]> in_data_;

- std::unique_ptr<uint8_t[]> payload_;

+ rtc::Buffer payload_;

std::unique_ptr<int16_t[]> out_data_;

WebRtcRTPHeader rtp_header_;