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Скачать или смотреть Handling Variable Sample Rates in Audio Streams with FFmpeg

  • vlogize
  • 2025-09-05
  • 3
Handling Variable Sample Rates in Audio Streams with FFmpeg
Feeding multiple sample rates to the same buffer source : ffmpeg filtersffmpeg
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Описание к видео Handling Variable Sample Rates in Audio Streams with FFmpeg

Discover how to process PCM audio with changing sample rates using FFmpeg filters effectively, ensuring seamless audio playback.
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This video is based on the question https://stackoverflow.com/q/63130471/ asked by the user 'samofoz' ( https://stackoverflow.com/u/3837653/ ) and on the answer https://stackoverflow.com/a/63169844/ provided by the user 'samofoz' ( https://stackoverflow.com/u/3837653/ ) at 'Stack Overflow' website. Thanks to these great users and Stackexchange community for their contributions.

Visit these links for original content and any more details, such as alternate solutions, latest updates/developments on topic, comments, revision history etc. For example, the original title of the Question was: Feeding multiple sample rates to the same buffer source : ffmpeg filters

Also, Content (except music) licensed under CC BY-SA https://meta.stackexchange.com/help/l...
The original Question post is licensed under the 'CC BY-SA 4.0' ( https://creativecommons.org/licenses/... ) license, and the original Answer post is licensed under the 'CC BY-SA 4.0' ( https://creativecommons.org/licenses/... ) license.

If anything seems off to you, please feel free to write me at vlogize [AT] gmail [DOT] com.
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Handling Variable Sample Rates in Audio Streams with FFmpeg: A Practical Guide

In today’s digital landscape, audio processing can often present unique challenges, particularly when dealing with multiple sample rates from varying sources. A common scenario arises when an audio stream, like PCM samples, fluctuates between different sample rates depending on network conditions. This can complicate playback, especially when you aim to convert these samples for use in audio decoders. In this guide, we will explore how to tackle this problem using FFmpeg and its powerful filtering capabilities.

The Challenge: Multiple Sample Rates

Imagine you have a remote audio source that provides PCM samples. However, this stream varies in sample rates, sometimes delivering audio at 16 kHz and at other times at 48 kHz. When attempting to process these audio samples, you might encounter the frustrating error message:

[[See Video to Reveal this Text or Code Snippet]]

This indicates a need for a robust solution that can handle the variability without interrupting the audio processing workflow. So, how can we achieve that?

The Solution: Using swr_convert_frame()

To address this issue effectively, the recommended approach is to utilize FFmpeg’s SwrContext. Here’s a breakdown of how you can set this up:

Step 1: Initialize SwrContext for Different Sample Rates

Since your audio source can switch between two distinct sample rates, you will need to create an array of SwrContext pointers. Each context can be configured for a specific sample rate. Here’s what the setup might look like:

Create an array to hold SwrContext * for each sample rate you expect to handle:

[[See Video to Reveal this Text or Code Snippet]]

Step 2: Configure the Contexts

Each SwrContext should be initialized according to its specific sample rate. For example:

[[See Video to Reveal this Text or Code Snippet]]

Make sure to replace parameters according to your audio requirements.

Step 3: Convert Audio Frames

As audio samples are fed into your application, use the correct SwrContext depending on the sample rate of the incoming audio. You might implement a switch-case or if-else structure to select the appropriate context for conversion:

[[See Video to Reveal this Text or Code Snippet]]

Step 4: Handling Output

Once conversion is complete, you can feed the processed audio to your decoder without needing to alter the underlying buffer format dynamically.

Conclusion

By utilizing multiple SwrContext instances, you can effectively manage and convert audio samples with variable sample rates without running into the error related to dynamic audio frame properties. This technique not only simplifies the process but also enhances audio playback quality and consistency.

If you are working with dynamic audio sources and FFmpeg, this approach will prove invaluable in managing complex audio transformations smoothly and efficiently. Don't hesitate to reach out if you have further questions or need assistance with your audio processing needs!

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