From 086c044dc34dfc0f74fbe41f4ecb402b2cd34884 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 03:13:33 +0200
Subject: Merging upstream version 125.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp')

diff --git a/dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp b/dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp
index 43041f81ea..1e8e488e25 100644
--- a/dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp
+++ b/dom/media/platforms/ffmpeg/FFmpegAudioDecoder.cpp
@@ -164,7 +164,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     int16_t* data = reinterpret_cast<int16_t**>(aFrame->data)[0];
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = AudioSampleToFloat(*data++);
+        *tmp++ = ConvertAudioSample<float>(*data++);
       }
     }
   } else if (aFrame->format == AV_SAMPLE_FMT_S16P) {
@@ -174,7 +174,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     int16_t** data = reinterpret_cast<int16_t**>(aFrame->data);
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = AudioSampleToFloat(data[channel][frame]);
+        *tmp++ = ConvertAudioSample<float>(data[channel][frame]);
       }
     }
   } else if (aFrame->format == AV_SAMPLE_FMT_S32) {
@@ -183,7 +183,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     int32_t* data = reinterpret_cast<int32_t**>(aFrame->data)[0];
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = AudioSampleToFloat(*data++);
+        *tmp++ = ConvertAudioSample<float>(*data++);
       }
     }
   } else if (aFrame->format == AV_SAMPLE_FMT_S32P) {
@@ -193,7 +193,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     int32_t** data = reinterpret_cast<int32_t**>(aFrame->data);
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = AudioSampleToFloat(data[channel][frame]);
+        *tmp++ = ConvertAudioSample<float>(data[channel][frame]);
       }
     }
   } else if (aFrame->format == AV_SAMPLE_FMT_U8) {
@@ -202,7 +202,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     uint8_t* data = reinterpret_cast<uint8_t**>(aFrame->data)[0];
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = UInt8bitToAudioSample<AudioDataValue>(*data++);
+        *tmp++ = ConvertAudioSample<float>(*data++);
       }
     }
   } else if (aFrame->format == AV_SAMPLE_FMT_U8P) {
@@ -212,7 +212,7 @@ static AlignedAudioBuffer CopyAndPackAudio(AVFrame* aFrame,
     uint8_t** data = reinterpret_cast<uint8_t**>(aFrame->data);
     for (uint32_t frame = 0; frame < aNumAFrames; frame++) {
       for (uint32_t channel = 0; channel < aNumChannels; channel++) {
-        *tmp++ = UInt8bitToAudioSample<AudioDataValue>(data[channel][frame]);
+        *tmp++ = ConvertAudioSample<float>(data[channel][frame]);
       }
     }
   }
-- 
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